00:00.10 | japerry | both of those error messages come with varying gibberish in debug every second or so |
00:00.21 | directory | they aren't errors |
00:00.26 | directory | they are debug messages, vastly different |
00:00.38 | japerry | directory: err debug message. sorry =) |
00:01.09 | mrdigital | hey CunningPike: can i pm u |
00:01.13 | CunningPike | japerry: Hmm - I'm beginning to wonder about network latency or some other issue |
00:01.34 | CunningPike | mrdigital: Spit it out right here - we're all friends here ;) |
00:02.00 | mrdigital | whats the difference between SPA 3000 and 3102 |
00:02.08 | rob0 | 102! |
00:02.08 | Druken | 102 |
00:02.10 | Druken | :) |
00:02.17 | japerry | CunningPike: hmm that'd be odd though eh? because its on a dedicated 4 channel T1, and the phones+asterisk system are on their own 10/100 switch segmented physically away from the rest of the network |
00:02.19 | CunningPike | Spot the comedians |
00:02.50 | Druken | a 4 channel t1? |
00:02.50 | CunningPike | mrdigital: It is supposed to be a newer replacement for the 3000 - not sure what the difference is |
00:03.03 | mrdigital | will the 3000 work just fine? |
00:03.10 | Druken | mine works... |
00:03.29 | CunningPike | japerry: What switches are you using? |
00:03.31 | japerry | the T1 is called 'flexgrow' |
00:03.44 | CunningPike | japerry: Just tossing out ideas here, really |
00:03.48 | Druken | mines called fantasy |
00:03.52 | japerry | Cunningpike its a D-Link soho 24port 10/100 switch |
00:03.53 | mrdigital | Druken: how does it work with astrisk?.. you plug the pstn line into the box.. then connect to asterisk so when a call comes in it goes to asterrisk then asterisk spits it back to the box to the phoen connected to it |
00:04.16 | Druken | it can work that way... |
00:04.19 | japerry | the flexgrow however doesn't act like a PRI |
00:04.34 | japerry | it acts more like 4 PSTN lines bundled onto a T1 line |
00:04.47 | mrdigital | i have 1 pstn. and 1 analog phone i want to use with asterisk |
00:04.56 | japerry | so channels 9-12 are being used and configured as fxo |
00:05.14 | Druken | mrdigital: then you want the 3000 or 3102 |
00:05.39 | CunningPike | japerry: So, what do your zaptel and zapata files look like? |
00:05.42 | CunningPike | ~pb |
00:05.43 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
00:05.43 | mrdigital | and thats it besides the comp? |
00:06.15 | Druken | mrdigital: yep |
00:06.17 | *** part/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
00:06.43 | mrdigital | can i mod the box to connect to multiple analogs? |
00:07.00 | Druken | meaning? |
00:07.08 | CunningPike | :S |
00:07.48 | mrdigital | <pstn line> ------ <spa3000> ----- <rj11/cat3cable> ------ <box 1> ------- <box 2> ------ <box 3> |
00:07.57 | mrdigital | box 1 - 3 = normal phone outlet |
00:08.19 | Druken | uhmm.. yeah... just plug in as many phones as you want... |
00:08.26 | CunningPike | mrdigital: If you want them all to be on the one line, I guess - not sure what REN the SPA is able to provide |
00:08.46 | japerry | heh |
00:08.46 | mrdigital | ren? |
00:08.49 | japerry | pastebin not working |
00:08.51 | Druken | i know my rt31p2 rings like 6 phones in my house |
00:09.03 | CunningPike | japerry: try pastebin.ca |
00:09.11 | CunningPike | ~ren |
00:09.19 | h3x | thats coz current day phones have a ren of like 0.000001 |
00:09.43 | Druken | exactly |
00:09.44 | Druken | :) |
00:09.58 | Druken | unless you got an old rotary phone... it's no big deal |
00:10.06 | h3x | a crystal radio prob uses more power |
00:10.06 | h3x | heh |
00:10.10 | mrdigital | so i should be ok? |
00:10.18 | Druken | yep |
00:10.25 | mrdigital | theres only 2 phones getting hooked up |
00:11.02 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
00:11.17 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
00:12.25 | phigwork | giesen: Sorry, are you still there? |
00:12.28 | Druken | this is such a fucked up tv show.... |
00:12.29 | japerry | Cunningpike: its e&m |
00:12.42 | CunningPike | Hmmm |
00:12.56 | Druken | anyone ever seen "big love" ? |
00:13.33 | phigwork | giesen: If I do exten => _9.,1,Dial(Zap/1,${EXTEN:1}), it just gives me a dialtone when I dial 9+anything (which I can then dial out from, but that's silly) |
00:14.01 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
00:14.04 | Druken | uhmm... change that , to a / |
00:14.07 | CunningPike | japerry: I gotta run for home - I'll be back on later. I'd focus on the 'pri debug' side of things and see what comes up, seeing as it's external outgoing calls only |
00:14.20 | CunningPike | jbot ren is Ringer Equivalence Number - a telephone line can normally supply upto 4 REN, where a standard telephone/answering machine etc would equal 1 REN |
00:14.22 | jbot | CunningPike: okay |
00:14.25 | japerry | Cunningpike: okay, me too http://pastebin.ca/73561 |
00:14.51 | h3x | i would say an old phone with a bell ringer in it |
00:14.54 | japerry | cunningpike: thats the pastbin for zapata.. you'll notice I statically assigned the callerid because the other method was reulting in no numbers |
00:15.13 | CunningPike | japerry: OK |
00:15.29 | CunningPike | I'll do some thinking on the ride home :) |
00:16.33 | Druken | phigwork: you make that change? |
00:17.16 | phigwork | Druken: which one? |
00:17.16 | japerry | cool, thanks =) |
00:17.30 | Druken | [20:14] <Druken> uhmm... change that , to a / |
00:18.03 | phigwork | sorry missed that :) |
00:18.44 | Druken | :) |
00:19.14 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
00:19.34 | phigwork | Druken: i don't think it's dialing all the numbers |
00:19.47 | phigwork | sitting at silence atm.. then three tones, we're sorry, your call did not go through |
00:19.53 | Druken | well, not with your old way... no :) |
00:20.09 | phigwork | exten => _9.,1,Dial(Zap/1/${EXTEN:1}) |
00:20.47 | Druken | show me the cli output |
00:20.56 | Druken | cause that would be working |
00:20.57 | phigwork | <PROTECTED> |
00:21.14 | phigwork | i got the "remember, you must now dial 1 + the area code, or 0 + the area code" |
00:21.20 | phigwork | <PROTECTED> |
00:21.36 | Druken | it did dial the 1.. hehe |
00:21.44 | phigwork | maybe the card is dialing too fast |
00:21.54 | Druken | could be... |
00:21.56 | phigwork | not enough time between pick up of line and dialing |
00:22.16 | phigwork | is there a pause? |
00:22.23 | phigwork | pause character, that is |
00:22.30 | Druken | nope |
00:22.38 | phigwork | :/ |
00:22.46 | phigwork | atdt, |
00:22.46 | Druken | browse in zapata.conf |
00:22.47 | phigwork | :) |
00:23.20 | Druken | not dialing a modem.... |
00:23.21 | Druken | hehe |
00:23.45 | phigwork | yeh, not much in there |
00:24.08 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
00:24.13 | *** join/#asterisk chumper2342 (n=cj@cpe-70-112-211-200.austin.res.rr.com) |
00:24.51 | chumper2342 | How do I create custom intro voice menus? Like "Thanks for calling Company name" |
00:24.54 | phigwork | ah well. I guess I'll just let it dial the 9 |
00:25.38 | phigwork | chumper: i bet that's in a faq somewhere, cause i know i'd like to know the same myself. i'm sure a lot of people do. |
00:26.18 | chumper2342 | i know its possible, cause the company I work for has a custom intro |
00:26.51 | chumper2342 | i think it might cost money |
00:26.59 | phigwork | Drukon: how do you add strings together? commas? |
00:27.18 | Druken | add string together? |
00:27.25 | phigwork | concatinate |
00:27.31 | InfraRed | chumper2342: save your own .wav file |
00:27.34 | InfraRed | then read the wiki |
00:27.36 | Druken | like math? or just stringstring ? |
00:27.40 | phigwork | stringstring |
00:27.47 | Druken | exactly like that |
00:27.51 | phigwork | cool |
00:27.56 | Druken | exactly like that ${exten} |
00:27.59 | Druken | bah! |
00:28.07 | Druken | ${exten}${exten} |
00:28.10 | phigwork | thanks |
00:28.42 | benjamin7062 | Chumper... you are looking for something like this: |
00:28.44 | benjamin7062 | exten => s,n,Playback(tt-somethingwrong&tt-weasels) |
00:29.12 | chumper2342 | yep |
00:29.43 | phigwork | oh ya, i've been wondering looking through examples and stuff, what is N? |
00:29.44 | benjamin7062 | Open up sound recorded... make a wav file... copy it to the same dir as the file tt-weasels... then use that exten line and call your file |
00:29.45 | benjamin7062 | poof |
00:29.49 | phigwork | oh number, duh |
00:29.55 | phigwork | and X is any character? |
00:29.56 | benjamin7062 | n is the 'next' priority |
00:30.00 | phigwork | oh |
00:30.01 | phigwork | word |
00:30.01 | benjamin7062 | instead of numbering the priorities |
00:30.05 | Druken | NEXT! :) |
00:30.49 | chumper2342 | how can I get it to sound like the lady? |
00:30.58 | phigwork | i'm just setting up outgoing rules for each scenario.. local with no area code, local area codes, 18XXetc, 17XXetc, and 0 |
00:30.59 | *** join/#asterisk moonwick (n=moonwick@core.dump.net) |
00:31.00 | benjamin7062 | You can also do n(name) and call a priority by 'name' with a Goto Application |
00:31.06 | phigwork | is there a better way of setting this up? |
00:31.16 | *** join/#asterisk coppice (n=chatzill@223.193.17.210.dyn.pacific.net.hk) |
00:31.19 | benjamin7062 | phigwork, not really |
00:31.25 | phigwork | cool |
00:31.28 | Druken | chumper2342: get allison to record it for you |
00:31.41 | benjamin7062 | phigwork, don't forget international.. =) |
00:31.48 | benjamin7062 | _X011XXXX etc |
00:31.54 | benjamin7062 | err.. that was wrong |
00:32.10 | phigwork | i'm not payin no long distance :) |
00:32.10 | benjamin7062 | _9011XXX |
00:32.14 | benjamin7062 | whatever.. you get the point |
00:32.48 | benjamin7062 | =) |
00:33.07 | *** join/#asterisk droops (n=root@adsl-065-005-212-128.sip.jan.bellsouth.net) |
00:34.51 | coppice | wow. intel has found a sucker to buy its xscale and phone chip business. weird :-\ |
00:35.19 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
00:35.19 | phigwork | i wonder why you can't call 800 numbers through fwd anymore |
00:37.29 | benjamin7062 | coppice, That's because Intel wants to see a P4 in every phone... instead of cancer... you then have to worry about skin burns. |
00:38.07 | benjamin7062 | coppice, They are trying to get people to stop making calls while driving |
00:38.30 | Druken | ain't gonna happen |
00:38.49 | benjamin7062 | Beh... you can wear batteries around your waist... it'll happen |
00:38.54 | benjamin7062 | =) |
00:39.26 | Druken | ain't gonna happen people stop using the phone while driving :) |
00:39.44 | coppice | intel's phone chips suck (jokes about heat sink fans in a GSM phone aside), and nobody has bought them. dumping the business at a massive loss make sense. why does anyone want to buy it though. |
00:40.04 | coppice | will Dialogic be next? |
00:41.18 | coppice | they can't get 802.11 working well, either, but they still aim to take the world by storm with WiMax :-) |
00:42.29 | anonymouz666 | my friend won here... "intel hero" with wimax projects |
00:42.52 | benjamin7062 | Wireless net everywhere... Wi-whatever... I don't care... but I want Wi(something) everywhere in a major city... That will simply be HAWT |
00:42.59 | benjamin7062 | something needs to do this.. NOW.. and for free |
00:43.17 | benjamin7062 | coppice, that's a good business model... lets start a company |
00:43.32 | benjamin7062 | If we could go back to 1997 I bet we could get a BUTT load of VC |
00:44.18 | coppice | if i were really rich i'd like to buy dialogic and clsoe them down as a public service. far more benevolent than anything bill gates will ever do |
00:44.47 | Druken | benjamin7062: uhmm... yeah... toronto hydro has ya beat :) |
00:44.48 | anonymouz666 | hah |
00:46.58 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
00:47.47 | rob0 | LOL @ http://www.voip-info.org/wiki/view/Cheapest+ATAs+and+Service ... "VoIPSUPPLY PLEASE QUIT DELETING OUR LINKS" |
00:47.58 | anonymouz666 | the ISP's here wanna drop all the VoIP traffic. No more VoIP with domestic ADSL modems |
00:50.55 | benjamin7062 | anonymouz666, I would hate to be with an ISP that does that. |
00:51.27 | benjamin7062 | rob0, that's dirty... |
00:52.17 | anonymouz666 | benjamin7062: No choice. The ISP's are the Telcos. |
00:52.35 | rob0 | Yeah, if that's true, VoIPSupply ought to be kicked off the wiki. |
00:53.10 | rob0 | If not, Digiumcards.com ought to be kicked off. :) |
00:54.06 | rob0 | You give a greedy and stupid merchant free advertising ... they do greedy and stupid things. |
00:55.03 | Druken | i guess i was never a greedy and stupid merchant.... |
00:56.21 | chumper2342 | anybody have any ideas on why I can't hear audio when I make external calls, but I can call the soft phone next to me and hear audio |
00:56.46 | chumper2342 | i set shorewall rules ACCEPT all all udp 10000:33000 |
00:56.47 | X-Rob_ | chumper2342, give your asterisk box a non-natted IP address. |
00:56.57 | chumper2342 | it has a public ip |
00:57.05 | *** join/#asterisk Freman (n=twitsrus@jaguar.wbs.net.au) |
00:57.07 | chumper2342 | it was working yesterday i believe |
00:57.20 | [TK]D-Fender | chumper2342 : You need 5060, 10000-20000, and a bunch of settings in [general] in sip.conf |
00:57.20 | X-Rob_ | well, reboot stuff until it starts working again then. |
00:57.33 | [TK]D-Fender | chumper2342 : externip, localnet, and nat=yes |
00:57.34 | chumper2342 | sure |
00:57.48 | X-Rob_ | anyone had any experience with the netcomm v85 phones? |
00:58.19 | Freman | any tips on making a multi-user multi-vsp system? for example I'm already running provider A for myself, garry has some along and I'm going to proxy him through my asterisk so he gets extra features, but he's also signed up with provider A. I want to make minimal changes to my existing extensions so that when garry calls it uses his account at provider A, but when I call it uses mine |
01:00.43 | [TK]D-Fender | Freman : Contexts are your friends. I have done a setup for 4 companies running on the same server with seperate dialplans, ivr's, queues and the works. |
01:01.22 | Freman | I'd rather not duplicate everything for garry tho |
01:01.24 | chumper2342 | actually, i can hear sound for half a sec when i pickup my cell (using my softphone to call cell) |
01:01.41 | Freman | like... setting a variable and using the variable all through to tell the difference |
01:09.04 | *** join/#asterisk Carp1 (i=Carp1@ip-204-97-151-161.modem.logical.net) |
01:09.05 | phigwork | man, I'm still wishing for a skype gateway, or at least some way to log in to skype |
01:10.00 | Carp1 | Which ports do I need open? I am using NuFone connecting with IAX and Using SIP for IP Phones |
01:13.09 | rob0 | Hey! I was just going to ask about ports too. :) |
01:14.37 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
01:14.57 | *** join/#asterisk svenbart (n=svenbarg@pool-70-20-30-53.bstnma.fios.verizon.net) |
01:15.02 | rob0 | I have 2000/tcp, and {2727,4520,5060,4569}/udp bound. I know what some of those are, but not all. |
01:15.38 | rob0 | http://www.voip-info.org/wiki/index.php?page=Asterisk+firewall+rules |
01:15.48 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
01:16.26 | rob0 | What are 2000/tcp and 4520/udp? |
01:22.38 | *** join/#asterisk Greek-Boy (n=Greek-Bo@193.220.93.162) |
01:22.41 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.4) |
01:22.41 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
01:22.50 | Brijn | Good evening |
01:23.08 | Greek-Boy | oops |
01:23.09 | Greek-Boy | sorry |
01:23.11 | Greek-Boy | mistake |
01:23.55 | tengulre11 | Good morning!! |
01:24.33 | Greek-Boy | I can't record as described in http://www.voip-info.org/wiki/view/Monitor+stereo-example |
01:24.34 | Greek-Boy | :( |
01:24.38 | Greek-Boy | can someone help me? |
01:25.33 | [TK]D-Fender | Greek-Boy : Pastebin all the related bits of your config. |
01:25.35 | [TK]D-Fender | ~pb |
01:25.37 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
01:27.32 | [TK]D-Fender | Greek-Boy : and CLI output... |
01:29.18 | rob0 | 2000/tcp appears to be SCCP |
01:31.19 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
01:32.20 | Greek-Boy | http://pastebin.com/734096 |
01:32.24 | Greek-Boy | no CLI output |
01:34.18 | chumper2342 | can anyone take a look at this one way audio problem? http://bzflag.pastebin.ca/73605 |
01:35.47 | *** join/#asterisk userdefined (n=jross@cpe-24-169-142-23.rochester.res.rr.com) |
01:36.13 | rob0 | Looks like I might close SCCP with a "noload => chan_skinny.so" in modules.conf ? |
01:36.18 | chumper2342 | when I use my cell and call in, i can hear audio both ways |
01:37.23 | rob0 | 4520/udp appears to be DUNDi, which I probably want to keep, IIUC what it does. |
01:42.45 | [TK]D-Fender | Greek-Boy : You goofed pretty big... you have to do monitor BEFORE you dial... |
01:43.38 | Greek-Boy | lol |
01:44.05 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
01:44.08 | rob0 | Looks like I might close 2727/udp with a "noload => chan_mgcp.so" in modules.conf ? Do I need that if I'm just using Zap hardware and some remote IAX2 connections? |
01:44.20 | CrashHD | how can I pass a var from one a sub macro to the parent macro? |
01:44.24 | Carp1 | I just intalled asterisk and it used to config it for me |
01:44.31 | CrashHD | s/one/ |
01:44.38 | Carp1 | Just type 'asterisk ' and it starts |
01:44.42 | [TK]D-Fender | rob0 : nope... I killed Dundi, MGCP, and SCCP myself.. |
01:44.45 | Carp1 | whats its path/ |
01:44.52 | rob0 | [TK]D-Fender: thanks |
01:44.54 | *** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net) |
01:45.04 | chumper2342 | anybody seen this in cli when making call? Forcing Marker bit, because SSRC has changed |
01:47.52 | CrashHD | how can I modify the vars from another call var space? |
01:48.36 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
01:48.54 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
01:49.45 | chumper2342 | got it, forgot externip and localnet in sip.conf |
01:51.22 | CrashHD | I want to use the M option on the Dial() app but in the macro that is call I want to be able to set a var or return some value so that my dialplan can take that into account....this does not seem to happen. |
01:52.00 | [TK]D-Fender | CrashHD : Use AstDB. |
01:52.16 | CrashHD | but those vars are persistent correct? |
01:52.26 | [TK]D-Fender | CrashHD : and pass the UNIQUEID as part of the keyvalue |
01:52.26 | CrashHD | would that not conflict with multiple calls |
01:52.28 | CrashHD | ? |
01:52.36 | [TK]D-Fender | CrashHD : AstDB = consistant. |
01:53.07 | CrashHD | I will need to clear out the entries in the DB correct, to ensure the db size does not get to great? |
01:53.27 | [TK]D-Fender | CrashHD : Yup. |
01:53.52 | CrashHD | no easy way to just pass a return value eh? |
01:53.55 | CrashHD | lol |
01:54.21 | [TK]D-Fender | First thing you should do is set a local var upon hangup based on the DB value, then wipe it. Cron up a Family wipe call. |
01:54.33 | [TK]D-Fender | CrashHD : Imperfect solutions for an imperfect world. |
01:55.06 | CrashHD | *nods* |
01:55.18 | CrashHD | I might as well just use an AGI at that point |
01:55.44 | CrashHD | thanks for the heads up |
01:58.53 | [TK]D-Fender | np |
01:59.20 | [TK]D-Fender | There are just so many places that this is the only really viable mechanism for passing values... |
01:59.21 | Greek-Boy | [TK]D-Fender is it correct to use $SOXMIX -v 1 $LEFT-tmp.wav -v 1 $RIGHT-tmp.wav -v 1 $OUT.mp3? |
01:59.33 | chumper2342 | actually setting the externip and localnet only fixed 1 call, it worked once but still doesn't |
01:59.38 | [TK]D-Fender | Greek-Boy : Beats me..... I suck at Linux :) |
01:59.38 | Greek-Boy | doesn't seem to work for me. the output mp3 comes out empty :( |
02:01.27 | Carp1 | How to I start asterisk? |
02:01.27 | Carp1 | ? |
02:02.33 | Carp1 | ./path/to/asterisk -vvvc |
02:02.34 | Carp1 | ? |
02:02.34 | Greek-Boy | Carp1 /usr/sbin/asterisk -g |
02:02.35 | ManxPower | Carp1, "service asterisk start" or "asterisk" or "safe_asterisk" |
02:02.40 | Greek-Boy | read the wiki |
02:03.10 | *** join/#asterisk {DDP}Natas (n=natas212@69-165-73-122.sbtnvt.adelphia.net) |
02:03.22 | Carp1 | no such file or directory |
02:03.23 | Carp1 | hmm |
02:03.27 | {DDP}Natas | hi everyone |
02:03.32 | Carp1 | I installed using SVN |
02:04.14 | ManxPower | And? |
02:04.47 | Carp1 | [root@pbx asterisk]# service asterisk start |
02:04.47 | Carp1 | bash: service: command not found |
02:04.48 | Carp1 | [root@pbx asterisk]# /usr/sbin/asterisk -g |
02:04.48 | Carp1 | bash: /usr/sbin/asterisk: No such file or directory |
02:04.52 | {DDP}Natas | does anyone have any experience with a2billing? |
02:04.59 | Qwell | did you install it? |
02:05.12 | anonymouz666 | what means bridge two channels together? ZAP 1 and SIP-channel for example? |
02:05.14 | Carp1 | I followed hte download instructions |
02:05.19 | anonymouz666 | I did'n't understand |
02:05.24 | Qwell | Carp1: which are? |
02:05.24 | Carp1 | and at the end of the install it showed a big Asterisk in ASCII |
02:05.43 | Qwell | You didn't follow the instructions |
02:05.53 | Carp1 | http://www.asterisk.org/download |
02:06.00 | Carp1 | I followed the SVN instructions |
02:06.21 | Qwell | and what about the crap that it said on your screen during the install? |
02:06.32 | Carp1 | It was going to fast |
02:06.38 | Qwell | specifically the part where it says "make must be restarted" at the VERY end |
02:06.49 | Carp1 | Yeah, I seen that |
02:06.52 | Qwell | and? |
02:07.17 | *** join/#asterisk skraelings001 (n=skraelin@190.40.104.78) |
02:07.24 | Carp1 | I ignored it because when I installed asterisk like 5 years ago I remembered a whole bunch of errors and everything worked fine |
02:07.32 | Carp1 | Can you tell me whatr that means? |
02:07.36 | Qwell | it means... |
02:07.40 | Qwell | MAKE MUST BE RESTARTED |
02:08.05 | Carp1 | I don't know how to do that, is what I meant. Sorry |
02:08.14 | Qwell | you type make install... |
02:08.18 | Carp1 | right |
02:08.47 | Carp1 | I did that |
02:08.58 | Carp1 | I did "make clean; make instal" |
02:09.03 | Carp1 | install* |
02:09.15 | anonymouz666 | Qwell did you already use an app called bridge? |
02:09.25 | Carp1 | Does that mean to make install again? |
02:09.25 | Qwell | If you can't follow instructions...don't use trunk |
02:09.30 | Qwell | Carp1: YES |
02:10.00 | Carp1 | How am I supposed to know that? YOu don't need to get mad. I dont know if it mean type "make restart" I don't know linux...However, thanks. |
02:10.57 | Qwell | Like I said.. If you can't follow instructions...don't use trunk |
02:11.28 | directory | Qwell: but I want to! |
02:11.37 | Qwell | directory: no trunk for you either |
02:11.46 | Qwell | directory: You must maintain 1.0.x now |
02:11.47 | Carp1 | That still doesnt help me get asterisk started |
02:11.47 | directory | darn |
02:11.58 | {DDP}Natas | if anyone has any experience with a2billing plz msg me, i have a question! thnkx |
02:13.06 | *** join/#asterisk Synthe (i=Synthe@odo.synthe.net) |
02:13.51 | userdefined | anyone running * behind an ipcop firewall? |
02:14.57 | userdefined | i'm having a heck of a time getting internal->dmz to asterisk working, just curious if/how others have done it |
02:15.23 | userdefined | (tried asking on #ipcop, they're apparently all sleeping over there ;-) |
02:16.04 | rob0 | I don't use ipcop, but am pretty handy with iptables. |
02:17.02 | skraelings001 | good evening |
02:18.40 | userdefined | rob0: cool, thanks. unfortunately ipcop's got this nice little maze of twisty tables (all the same) and i'm unsure which one to add a '-j ACCEPT' to =/ |
02:19.02 | X-Rob_ | heh |
02:19.13 | X-Rob_ | iptables -P FORWARD ACCPT |
02:19.17 | X-Rob_ | iptables -F FORWARD |
02:19.22 | X-Rob_ | repeat for input and output |
02:19.24 | X-Rob_ | then start again |
02:19.33 | X-Rob_ | and, spell ACCEPT properly, too 8) |
02:19.36 | rob0 | xyzzy! |
02:20.09 | userdefined | heh |
02:20.37 | *** join/#asterisk onweald_tim (n=onweald_@c-67-173-213-205.hsd1.tx.comcast.net) |
02:21.12 | rob0 | userdefined: what specifically is your * trying to do that the firewall is blocking? |
02:21.30 | rob0 | userdefined: see also http://www.voip-info.org/wiki/index.php?page=Asterisk+firewall+rules |
02:21.52 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
02:21.58 | userdefined | asterisk to/from the net is working |
02:22.10 | userdefined | problem is from the 'green' interface to the 'orange' is failing |
02:22.21 | userdefined | (green == trusted, orange == dmz in ipcop parlance fwiw) |
02:22.40 | rob0 | SIP phones? *What* is it doing? |
02:23.04 | userdefined | rob0: timeout on register |
02:23.14 | rob0 | SIP? |
02:23.22 | userdefined | aye, sorry |
02:23.55 | rob0 | Usually the RELATED,ESTABLISHED rule should cover this. I presume ipcop uses that. |
02:24.38 | userdefined | it does |
02:24.52 | userdefined | and as i understand it, by default green->orange should be permitted |
02:24.53 | rob0 | go ahead and pastebin your iptables-save |
02:25.01 | userdefined | k. one sec |
02:25.14 | rob0 | make it "iptables-save -c" |
02:25.32 | rob0 | (doesn't matter much) |
02:27.10 | userdefined | iptables -n -L work ? |
02:27.17 | userdefined | not sure where ipcop keeps the save |
02:27.22 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
02:27.38 | rob0 | well it's not as good, but I'll look |
02:27.48 | userdefined | that's here-> http://pastebin.ca/73631 |
02:28.51 | rob0 | oops, no -v on that |
02:28.56 | userdefined | (fwiw, iptables-save isn't present on ipcop, not just being ornery =0) |
02:28.57 | rob0 | so it's useless |
02:29.01 | userdefined | ah. one sec, i'll fix |
02:29.38 | onweald_tim | Hi all. I am trying to get SIP working through a vonage router (motorola VT2442) and I think that vonage uses sip too. |
02:29.41 | onweald_tim | So is this an impossible senario? |
02:30.18 | onweald_tim | Essentially, two SIP connections can't NAT if you want connections in both directions. |
02:30.23 | onweald_tim | That is my theory... |
02:30.24 | rob0 | userdefined: I take it that the * server is in the DMZ and the SIP clients are in the LAN? |
02:30.56 | rob0 | SIP in green, * in orange? |
02:31.15 | onweald_tim | Looks like this: Grandstream gxp2000 -> Vonage router -> Cable modem -> internet -> cablemodem -> general router -> asterisk server |
02:31.51 | userdefined | rob0: correct |
02:31.53 | onweald_tim | My config, not userdefined in case there is any confusion... :-) |
02:32.55 | userdefined | rob0: http://pastebin.ca/73633 <-- iptables -Z;iptables -v -n -L |
02:34.58 | rob0 | so which ethX is which? |
02:35.08 | userdefined | eth0 == green, eth2 == orange |
02:35.46 | userdefined | it amuses me to no end that i can connect to this from work, but not from my home network =) |
02:36.28 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
02:36.43 | rob0 | is there any NAT being done? |
02:36.47 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
02:37.51 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
02:38.57 | rob0 | 192.168.3.3 is the * IP ... maybe it's a NAT problem |
02:39.37 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
02:39.48 | rob0 | Add rules to Chain PORTFWACCESS to ACCEPT for -i eth0 |
02:40.14 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
02:41.21 | rob0 | hmmm, looks like line 33 is a blanket ACCEPT for eth0 |
02:44.19 | rob0 | You could simply insert a rule higher up to ACCEPT for eth0. But I suspect it's NAT. If you're using the external IP for the SIP clients, perhaps that's not being DNATed from inside. Tell them to use 192.168.3.3, does that work? (Does the * machine have a route to get back to the LAN?) |
02:46.00 | userdefined | i've used both the outside (hostname) and the 3.3 ip, neither are working. wrt route back to the lan, it doesn't right this second, i added one to DMZHOLES earlier to see if that was the issue to no avail |
02:46.18 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
02:47.03 | rob0 | a route, as in "ip route add ..." or "route add ...": ip(8) or route(8). |
02:47.43 | rob0 | anyway, I'm done for ... afk. |
02:47.47 | userdefined | ah. as opposed to a route as in "a firewall rule" ... which isn't a route at all =P |
02:47.51 | rob0 | right |
02:48.27 | rob0 | if you're still having the trouble tomorrow I'll be around. |
02:48.46 | userdefined | cool. i'll keep poking at it, thanks for taking a look |
03:08.28 | benjamin7062 | If I can make internal calls fine; but when connecting via PRI I show the call connect but no audio? What could be wrong? Last dial attempt slammed the screen with: |
03:08.31 | benjamin7062 | app_dial.c:713 wait_for_answer: Unable to forward voice |
03:11.09 | Brijn | benjamin7062: where you the one setting sup the Prove of Concept before monday? |
03:11.34 | Brijn | Proof :) |
03:12.03 | benjamin7062 | Yes, and it worked. |
03:12.14 | benjamin7062 | Now I jsut compiled on a new machine using 2.16.x kernel and udev |
03:12.28 | benjamin7062 | and, well, dif. installation issues |
03:13.09 | benjamin7062 | The PoC went very well... we invested in the phones today so now I have to get this sucker live. |
03:13.14 | benjamin7062 | I have... well, 3 days |
03:13.15 | benjamin7062 | yippee |
03:14.54 | Brijn | Hahaha |
03:15.09 | Brijn | did you budget include Digium support :) |
03:15.40 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
03:16.25 | *** join/#asterisk Samoied (n=Samoied@201.22.215.135.adsl.gvt.net.br) |
03:16.55 | Brijn | benjamin7062: http://lists.digium.com/pipermail/asterisk-users/2003-August/011207.html |
03:17.37 | Brijn | Never used a PRI, but can you enable debug on PRI messages |
03:19.56 | benjamin7062 | Brijn, Yeah... tried all the 'normal' stuff |
03:19.59 | benjamin7062 | D channels are right |
03:20.07 | benjamin7062 | heck, I'm using the same config from the other machine |
03:20.25 | Brijn | Adn that box is OK? same driver version, same *? |
03:21.01 | benjamin7062 | yup |
03:21.04 | benjamin7062 | =( |
03:21.19 | benjamin7062 | I had seen that email thread |
03:21.25 | benjamin7062 | checked google and wiki first. |
03:21.26 | benjamin7062 | no love |
03:21.39 | benjamin7062 | thought I'd come here and see if other have had the same issue with PRI cards |
03:22.14 | *** join/#asterisk pdavid (n=chatzill@adsl-068-209-191-127.sip.mob.bellsouth.net) |
03:22.19 | pdavid | evening all |
03:22.41 | pdavid | anyone point me to some docs on setting up an spa3000 (NON-trixbox/A@H!) |
03:23.59 | *** join/#asterisk jsaunders (n=root@70.71.224.65) |
03:25.29 | jsaunders | Anyone know whats up w/ playing .mp3's through moh (* 1.2.6) and it sounding like it's cutting in and out. It's like it's auto dropping the volume at quiet parts of the song or something. When the song gets more intense you can hear it clear. Sounds weird. |
03:25.55 | Brijn | benjamin7062: does the other box have the same PRI card, identical? |
03:26.44 | benjamin7062 | Brijn, yeah -- pulled the card from the other system |
03:27.26 | Brijn | Box-A, Pri-A = problem, Box-B, Pri-B = OK, Box-A Pri-B = ? |
03:27.30 | Brijn | Where PRI is the card |
03:29.09 | Brijn | ~fxs |
03:29.14 | jbot | it has been said that fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
03:30.11 | jsaunders | Hmm, weird... you know what it was, having it on speakerphone and at full volume. If I back it off a little volume wise, it clears it up. Guess I'm taxing the transmitter or somethin'. |
03:30.33 | jsaunders | This is not the 1st phone I've experienced this with. |
03:30.33 | *** join/#asterisk seb- (n=seb@cpe-72-132-242-171.san.res.rr.com) |
03:30.37 | *** part/#asterisk phigwork (n=phigan@71-209-152-225.phnx.qwest.net) |
03:31.03 | seb- | my voip phone can accept calls even though it is BEHIND a NAT'ing firewall. How is it doing this? |
03:31.32 | *** join/#asterisk phigwork (n=phigan@71-209-152-225.phnx.qwest.net) |
03:31.58 | jsaunders | seb: What voip phone you talkin' about? |
03:32.54 | pdavid | no place i could get a hand with setting up the spa3000? |
03:33.40 | *** join/#asterisk spackle (n=spackle@ip207-199-243-35.static.ishsi.com) |
03:33.50 | directory | spackle: it's a spackle! |
03:35.42 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
03:39.18 | benjamin7062 | sigh |
03:39.22 | benjamin7062 | well, I got it working |
03:39.40 | benjamin7062 | But it works only if I disable the EC on the Sangoma |
03:39.43 | pdavid | maybe someone could walk through a quick setup of the spa3000 with me? |
03:39.47 | benjamin7062 | which is the entire reason we bought the Sangoma |
03:39.49 | benjamin7062 | bleh |
03:40.40 | *** join/#asterisk morcegao (n=jkj@c92537dc.rjo.virtua.com.br) |
03:40.45 | *** join/#asterisk JunK-Y (n=junky@70.81.175.205) |
03:41.32 | morcegao | Can anyone give me documentation about how to mount an register string to an asterisk using outbound proxy ? |
03:41.56 | *** join/#asterisk jayb (n=jaybinks@59.167.212.49) |
03:42.23 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
03:42.30 | jayb | hey guys... im having problems with the blindtx bug.. |
03:42.41 | jayb | can somone suggest how to turn off blind transfer all together ? |
03:44.22 | jayb | this is the bug im refering to : v |
03:44.23 | jayb | http://bugs.digium.com/view.php?id=7289 |
03:45.51 | [TK]D-Fender | benjamin7062 : whats the problem with it? |
03:45.52 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
03:46.44 | morcegao | any knows register string using outbound proxy ? |
03:47.32 | benjamin7062 | [TK]D-Fender, well, not sure actually. All I know is if I disable HW EC and reload wan iface's... ztcfg ... restart asterisk... I hear audio on the PRI's... if I enable following the same patterns... no audio... debug on the pri's .. doesn't show much and I get something about not able to pass audio from the zap_dial module |
03:48.12 | *** join/#asterisk japerry (n=japerry@c-71-197-215-234.hsd1.or.comcast.net) |
03:48.14 | benjamin7062 | debug on the span's shows a lot I should say... just nothing out of the ordinary |
03:49.03 | benjamin7062 | Followed several versions of directions from sangoma and the wiki... thus far no luck |
03:52.10 | spackle | benjamin7062 - did you disable echocan other places? IIRC you have to disable other forms of echo can. |
03:52.33 | *** join/#asterisk Joe__11 (n=develope@host217-114-154-220.pppoe.mark-itt.net) |
03:52.49 | benjamin7062 | spackle, I disabled per instructions in the wanpipe* configs... and left on in * as sangoma states this is required. |
03:53.11 | spackle | new card or current firmware? |
03:53.19 | benjamin7062 | brand new |
03:53.34 | benjamin7062 | Yeah... me too |
03:53.38 | *** part/#asterisk skraelings001 (n=skraelin@190.40.104.78) |
03:54.23 | *** join/#asterisk cypromis (n=michal@voiceworks.pl) |
03:54.58 | morcegao | any knows how to crete register string using outbound proxy ? |
03:55.05 | benjamin7062 | I did just notice the time stamps on the zap drivers vs the winpipe drivers were way off. Perhaps the sangoma install forgot to run ../zap_dir/make install.. =) |
03:55.10 | benjamin7062 | I just did and rebooted |
03:55.11 | benjamin7062 | who knows |
03:55.59 | benjamin7062 | nope |
03:56.00 | benjamin7062 | no love |
03:56.18 | spackle | are you connected to a PRI or a channel bank? |
03:56.24 | benjamin7062 | PRI |
03:56.52 | benjamin7062 | Wonder if the latest sangoma beta is compatible with * 1.2? Maybe it only works with 1.0? |
03:57.40 | benjamin7062 | I had to fix all sorts of broken links in the sangoma install script... for udev.... but unfortunately, it all worked and it isn't that easy for me |
03:57.44 | spackle | it only has to work with zaptel |
03:58.02 | benjamin7062 | spackle, err, good point |
03:58.28 | benjamin7062 | Perhaps this is a plot by digium to push their cards o.O |
03:58.37 | directory | ha |
03:58.42 | spackle | nah. |
03:58.52 | spackle | go back over your settings |
03:59.24 | benjamin7062 | Perhaps I'll find a line # if (sangoma) { loop forever and make noises }; |
03:59.42 | directory | wouldn't that be funny |
03:59.54 | benjamin7062 | I'd laugh... even if it was a comment |
04:00.30 | benjamin7062 | thing is... the card works fine as long as I turn off EC |
04:00.33 | benjamin7062 | ;-) |
04:00.51 | *** join/#asterisk netoguy (n=skelley@ppp-70-129-186-62.dsl.spfdmo.swbell.net) |
04:00.51 | benjamin7062 | so 'something' in the zap driver needs to wake up and listen to the EC channel or something |
04:01.49 | benjamin7062 | I would guess it would be something in the zaptel.conf but .. I see no relavence... |
04:01.51 | benjamin7062 | bum sticks |
04:01.55 | jayb | any ideas guys on how to fix my blind transfer problem ?? |
04:02.47 | Greek-Boy | why can't I have macros in my dialplan for non patterns? |
04:04.54 | onweald_tim | Any suggestion on how to trace sip through two routers + two cable modems to determine which is causing the problem? The firewalls look right. |
04:05.17 | benjamin7062 | onweald - tcpdump? |
04:05.56 | onweald_tim | benjamin7062: That would trace it on one side or the other but I don't know which router/modem is rejecting the packets. |
04:06.12 | onweald_tim | What I really need is a tracert for SIP ports |
04:06.39 | onweald_tim | I know there is some utility that will do it but I'll be damned if I can't find it. |
04:06.55 | onweald_tim | Using debian and I have searched apt up and down. |
04:06.59 | benjamin7062 | do you see the traffic on both ends.. or neither? |
04:07.25 | onweald_tim | It looks like the control ports are working but I get audio in one direction only. |
04:07.39 | benjamin7062 | is one behind a nat? |
04:08.06 | onweald_tim | Sporadic audio. I think one of the routers is flaky. |
04:08.09 | onweald_tim | Both are behind nat |
04:08.28 | onweald_tim | I think the other router is flaky but can't prove it to the person on the other end. |
04:08.35 | benjamin7062 | hmm... i assume you did a * forward on both firewalls to the asterisk box just to test? |
04:08.58 | onweald_tim | We went so far as to set both up on the DMZ. No good. |
04:08.59 | benjamin7062 | * = wildcard... forward all packets incoming to * box |
04:09.08 | benjamin7062 | hrrm |
04:09.15 | onweald_tim | Yeah. Very frustrating. |
04:09.20 | benjamin7062 | dmz could still = firewall |
04:09.26 | benjamin7062 | did you check the firewall on the * boxes? |
04:09.37 | onweald_tim | Important info: My side is a vonage router. |
04:09.42 | benjamin7062 | (just running through things in my head) |
04:10.03 | onweald_tim | Checked the firewall on the asterisk box. iptables -L showed everything clear. |
04:10.13 | onweald_tim | just during testing :-) |
04:10.38 | benjamin7062 | jayb, have you tried changing blind transfers to a completely dif feature... just to see if that removes it? |
04:10.54 | benjamin7062 | what kind of routers/firewalls? |
04:11.21 | jayb | I put blind transfer as ** |
04:11.35 | benjamin7062 | jayb, did you do a reload? (just asking) |
04:11.36 | jayb | but that then means that the DTMF tone for * is not transmitted to any IVR's |
04:11.43 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.39.Dial1.SanJose1.Level3.net) |
04:11.46 | benjamin7062 | oh |
04:11.47 | benjamin7062 | right |
04:11.47 | jayb | did reload res_features |
04:11.48 | onweald_tim | Grandstream gxp2000 -> Vonage Motorola VT2442 -> Cable modem -> internet -> cablemodem -> general router w/SIP support -> asterisk server |
04:11.49 | benjamin7062 | umm |
04:11.56 | onweald_tim | Not sure exactly what his router is. |
04:12.21 | benjamin7062 | jayb, can you change blind to ## or ###? Dunno if that works.. never tried. ;-) |
04:12.29 | jayb | benjamin - do you know anyway to turn blind transfer off totaly ?? |
04:12.31 | spackle | benjamin7062 - did you see the stuff in sangoma wiki about hardware echo can? http://sangoma.editme.com/wanpipe-asterisk-configure |
04:12.45 | benjamin7062 | jayb, not without disabling in the code... i'm sure there is a better way |
04:12.45 | jayb | benjamin = I DID put blind transfer to ## thinking that would fix it |
04:12.56 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.39.Dial1.SanJose1.Level3.net) |
04:12.59 | jayb | but it still means DTMF # is not sent out to any endpoints.. |
04:13.05 | benjamin7062 | spackle, yeah. Unfortunately, I did.. but I'll go read for giggles to make sure |
04:13.07 | jayb | the blind transfer functionality STOPS it |
04:13.33 | benjamin7062 | jayb, good point. Wonder what happens if you leave it blank? |
04:13.40 | jayb | and it does the same thing no matter if its "#", "*", "##", "**" |
04:13.47 | [TK]D-Fender | benjamin7062 : Sorry for the delay. Check your wan_ec folder for a defunct PID file I believe that is locking up your EC and the cause of loss of audio |
04:13.57 | [TK]D-Fender | benjamin7062 : I've dealt with this twice before |
04:13.58 | jayb | yea I might try that later tonight.. |
04:14.08 | jayb | I have lots of calls in the system now, so dont want to go TO experimental.. |
04:14.09 | benjamin7062 | [TK]D-Fender, hawt... let me check |
04:14.28 | jayb | im actualy looking at jumping in there and recompiling res_features without any blind transfer support |
04:14.32 | benjamin7062 | jayb, if you type asterisk -vx'stop now' that will fix your call problem... |
04:14.33 | benjamin7062 | heheh |
04:14.37 | jayb | if worst comes to worst.. but I Dont want to do that if I can avoid it |
04:14.55 | benjamin7062 | s/v/r/ |
04:14.59 | [TK]D-Fender | jayb : Running Zap analog phones? |
04:15.07 | jayb | nope.. |
04:15.13 | jayb | only IAX and SIP endpoints. |
04:15.18 | *** join/#asterisk n3glv (n=Omega__@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
04:15.22 | [TK]D-Fender | jayb : then why are you using features.conf for blind transfers at all? |
04:16.05 | benjamin7062 | [TK]D-Fender, so, umm, where is wan_ec? /dev /proc? locate/find no match |
04:16.18 | [TK]D-Fender | benjamin7062 : /etc/wanpipe |
04:16.34 | onweald_tim | Adios. Gotta go get some sleep. |
04:16.40 | benjamin7062 | maybe that's my problem... dir doesn't exist. |
04:16.47 | benjamin7062 | onweald_tim, good luck! |
04:16.48 | [TK]D-Fender | benjamin7062 : Tha may be OK |
04:17.03 | spackle | benjamin7062 - do you have the udev devices set up with correct permissions? |
04:17.07 | [TK]D-Fender | benjamin7062 : it moved after a certain revision of wanpipe. |
04:17.14 | [TK]D-Fender | benjamin7062 : cruise for it. |
04:17.19 | jayb | D-Fender -- well how do I turn it off though !? |
04:17.20 | [TK]D-Fender | under /proc I think |
04:17.27 | onweald_tim | Thanks benjamin7062. Appreciate your brain cycles. Post here if you think of anything else. :-) |
04:17.35 | jayb | it was commented out in features.conf... but that defaults to # for blind transfer |
04:17.38 | benjamin7062 | onweald_tim, will do! |
04:17.39 | [TK]D-Fender | jayb : don't use "tT" when you Dial |
04:17.51 | benjamin7062 | spackle, hrrmm??? Maybe that's my problem |
04:18.05 | benjamin7062 | OMG... Why didn't I think of that |
04:18.09 | jayb | ive also removed Tt from the dial statement. |
04:18.13 | benjamin7062 | @ jayb |
04:18.27 | n3glv | he guys, why can't * handle the fourth column of DTMF? |
04:19.11 | n3glv | is there a conf somewhere to jiggle to allow A B C D touchtones? |
04:19.24 | benjamin7062 | spackle, if * is running as root and /dev/wp* is owned by root... I assume permissions are okay? |
04:19.44 | benjamin7062 | same with /dev/zap/* |
04:19.46 | spackle | yep. |
04:19.56 | spackle | prolly |
04:20.52 | n3glv | is x86 around out there? |
04:23.01 | benjamin7062 | I might have my problem |
04:23.11 | spackle | do tell |
04:23.20 | benjamin7062 | Setup points to cp util/wan_ec stuff .. but it doesn't exist in the source |
04:23.29 | *** part/#asterisk Joe__11 (n=develope@host217-114-154-220.pppoe.mark-itt.net) |
04:23.32 | [TK]D-Fender | What version are you running? |
04:23.37 | benjamin7062 | if wan_ec is the root.. it doesn't exist in source anymore I'd have to read more if this is relavant |
04:23.48 | benjamin7062 | wanpipe-beta4-2.3.4.tgz |
04:26.15 | benjamin7062 | The dir it's looking for doesn't exist in the source so this entire section doesn't run in Setup... which basically creates the /dev if kern = 2.4 and copies the wan_ec utils (which aren't there) |
04:26.45 | n3glv | I would like to do some specialized controll stuff with the 4th col of DTMF |
04:27.39 | [TK]D-Fender | benjamin7062: Sounds like a botched install all right |
04:28.42 | benjamin7062 | Maybe if I can find an older version of their tarbal it will have all the crud |
04:29.29 | *** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
04:29.40 | P-NuT | Hi all. |
04:30.11 | spackle | benjamin7062 - prolly won't work with the echo can then. |
04:31.01 | benjamin7062 | Well, those poop faces should include the EC stuff in their source! o.O |
04:31.13 | P-NuT | Say, if I was using an SPA3000 as a PSTN gateway, I'd only need to compile asterisk, asterisk addons and asterisk sounds wouldnt i? No zaptel and libpri. Is that right? |
04:31.22 | directory | [TK]D-Fender: !!! |
04:31.31 | jayb | guys one other question... |
04:31.39 | CunningPike | P-NuT: You don't even really need asterisk-addons or asterisk-sounds |
04:31.39 | jayb | asterisk & Vmware... |
04:31.49 | P-NuT | I don't? |
04:31.55 | jayb | is anyone doing this on any system that has a moderate amount of calls... ?? |
04:31.58 | CunningPike | P-NuT: They're optional extras |
04:32.04 | P-NuT | what are they? |
04:32.18 | CunningPike | P-NuT: Are you planning to use MeetMe or MOH? |
04:32.27 | P-NuT | MOH yes.. |
04:32.42 | benjamin7062 | jayb, -- I've talked to 'many' who do when evaluating the system |
04:32.47 | P-NuT | Damn, I can't connect to svn.digium.com |
04:33.07 | CunningPike | P-NuT: You'll need ztdummy then for timing, so you'd better install zaptek |
04:33.15 | P-NuT | zaptek? |
04:33.18 | CunningPike | s/zaptek/zaptel/ |
04:33.32 | P-NuT | oh ok then. |
04:33.38 | jayb | hehe . |
04:33.52 | jayb | yea I realise Ill need ZTDummy, but even with that.. Ive heard of problems with timing in vmware. |
04:33.58 | n3glv | on some SMP kernels there is an issue with zaptel / ztdummy |
04:34.14 | jayb | yea ok... anything to identify these kernels ?? |
04:34.15 | n3glv | trixbox for one ships with a bad smp setu |
04:34.16 | n3glv | setup |
04:34.32 | jayb | coz Id like to run a BIG (Quad or something) server |
04:34.36 | benjamin7062 | sigh... all their damn code points to wan_ec -- they 'forgot' to include it with the source |
04:34.37 | n3glv | only matters for dual cpu or dual core |
04:34.38 | benjamin7062 | argg |
04:34.44 | jayb | with VMWare ESX, and a few ASterisk instances.. and Id like it rock solid.. |
04:35.13 | n3glv | vmware running under unix? |
04:35.17 | jayb | if its not possible, then so be it.. just wanted to hear if anyone has actualy done it |
04:35.26 | n3glv | it can be done |
04:35.31 | n3glv | am sure someone is doing it |
04:35.32 | jayb | yea vmware running in unix.. (ESX is built on top of redhat enterprise) |
04:35.44 | [TK]D-Fender | directory : I DON'T WANT TO MEET YOU MOM! |
04:36.03 | jayb | id just like to hear from that "Somebody" who is doing it. |
04:36.08 | directory | [TK]D-Fender: hahahahaha |
04:36.11 | jayb | to talk about potential pitfalls, and what to look out for :) |
04:36.12 | directory | [TK]D-Fender: so THAT'S what you're doing |
04:36.16 | n3glv | there are some huge systems running * |
04:36.23 | *** part/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
04:36.24 | n3glv | lot of university systems etc |
04:36.43 | benjamin7062 | Sigh |
04:36.46 | benjamin7062 | I hate moments like this |
04:36.52 | benjamin7062 | Where I realize that I'm an idiot |
04:36.59 | jayb | :P |
04:37.14 | n3glv | ok jay, the one I know of is from trixbox, I have that issue personally |
04:37.22 | n3glv | but if you install from source etc |
04:37.25 | P-NuT | Does anyone work for digium here? |
04:37.29 | n3glv | then there seems to be no problem there |
04:37.31 | jayb | yea I do build asterisk from source.. |
04:37.31 | benjamin7062 | The HWEC addon must be installed for AFT A104D and A200D Cards when using Hardware Echo Cancellatoin. Without this addon the HW Echo Cancellation will NOT be enabled. |
04:37.35 | directory | P-NuT: what'cha need? |
04:37.48 | benjamin7062 | <-- steps on his own toe |
04:37.53 | benjamin7062 | <-- HARD |
04:37.55 | jayb | n3glv - but do you know what kernel problem is with trixbox ?? |
04:38.13 | P-NuT | I was going to say that the SVN repositories were offline, but they've come good now. |
04:38.20 | jayb | n3glv -- coz typicly Ive used binary kernels... but if you can point out to me what I need to do to recompile an asterisk happy kernel that would be great :) |
04:38.23 | n3glv | no, have not gotten that far into it, just know that the zaptel stuff is broken |
04:38.30 | jayb | oh ok. |
04:38.32 | n3glv | I can get u a kernel id |
04:38.38 | Juggie | jayb, distro? |
04:38.47 | jayb | Im still open to suggestions. |
04:38.52 | Juggie | what distro are you using? |
04:38.54 | jayb | was thinking CentOS.. |
04:38.59 | Juggie | good idea |
04:39.00 | Juggie | except |
04:39.06 | Juggie | ??centos |
04:39.07 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
04:39.11 | Juggie | ?? centosbug |
04:39.12 | benjamin7062 | n3glv, I've just compiled 3 times using the 2.4 and 2.6 kernels just fine... with the debian base installs.. FYI |
04:39.16 | benjamin7062 | Did one today |
04:39.20 | Juggie | grrr.... |
04:39.21 | benjamin7062 | In case that helps |
04:39.24 | Juggie | jbot, centosbug |
04:39.26 | jbot | i heard centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
04:39.31 | n3glv | 2.6.9-34.ELsmp |
04:39.34 | Juggie | dont forget that :) |
04:39.46 | jayb | id love to use debian... however te IBM hardware Im getting, provides support and drivers for RHEL (and I Assumed centos might be a good idea) |
04:39.47 | *** join/#asterisk _alex_mx_ (n=alex@dsl-200-67-125-45.prod-empresarial.com.mx) |
04:39.57 | Juggie | you assumed correctally. |
04:40.02 | Juggie | you can use the centos vanilla kernel |
04:40.06 | Juggie | no need to recompile |
04:40.14 | Juggie | just be sure to install kernel-devel |
04:40.14 | Juggie | and your good to go |
04:40.20 | jayb | ok.. |
04:40.20 | Juggie | and of course |
04:40.23 | Juggie | jbot, centosbug |
04:40.25 | jbot | centosbug is, like, a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
04:40.29 | Juggie | as i said. |
04:40.31 | Juggie | dont forget that |
04:40.33 | Juggie | or zaptel will bitch |
04:40.35 | jayb | haha ok |
04:40.59 | jayb | ok great... |
04:41.01 | benjamin7062 | So, jayb -- What are you going to run before you compile? |
04:41.03 | benjamin7062 | hint |
04:41.05 | benjamin7062 | jbot, centosbug |
04:41.06 | jbot | centosbug is probably a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
04:41.13 | jayb | jbot, centosbug :) |
04:41.16 | benjamin7062 | ahole! Don't ignore me! |
04:41.23 | benjamin7062 | heh |
04:41.25 | jayb | hehe . |
04:41.27 | benjamin7062 | I got ignored by a computer |
04:41.32 | benjamin7062 | ~jbot |
04:41.34 | jbot | somebody said jbot was only marginally useful at best, He got a C- on his Turing Test, or a complete idiot, or a dolt |
04:41.36 | benjamin7062 | ~fart |
04:41.37 | jbot | ACTION farts, releasing large quantities of methane and sulfur dioxide. "Evacuate the channel! GO! *gag* SAVE YOURSELVES *cough* MOVE *choke* MOVE!" |
04:41.41 | jayb | ok so I can do a clean centos 4.2 install... run that command |
04:41.48 | jayb | then build / install asterisk and it should be good |
04:41.52 | Juggie | uhhuh |
04:41.54 | jayb | to run in a VMWare guest OS ?? |
04:42.00 | Juggie | hah |
04:42.00 | Juggie | no |
04:42.05 | P-NuT | jbot: So, do I need to setup ztdummy, or is it just there automatically for the timing of MOH? |
04:42.07 | jbot | P-NuT: what are you talking about? |
04:42.16 | Juggie | why would you run * in vmware |
04:42.21 | Juggie | i guess it would be ok without any hardware |
04:42.24 | Juggie | but it wont work with hardware |
04:42.25 | P-NuT | jbot wrote: jbotCunningPike meant: P-NuT: You'll need ztdummy then for timing, so you'd better install zaptel |
04:42.33 | jayb | thats what my question was relating to (above :P) |
04:42.45 | directory | jbot: botsnack |
04:42.45 | jbot | :), directory |
04:42.47 | jayb | yea no hardware... just sip & IAX2 |
04:43.03 | P-NuT | jbot: that's what I'm talkking about |
04:43.29 | benjamin7062 | jbot -- you smell funny |
04:43.39 | n3glv | anyone got a url on that kernel fix for centos? is it in the wiki? |
04:44.05 | jayb | jbot, centosbug :) |
04:44.10 | jayb | is that what your after ? :) |
04:44.14 | jayb | jbot, centosbug |
04:44.15 | jbot | i heard centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
04:44.50 | benjamin7062 | jbot should be booted for flooding |
04:46.46 | _alex_mx_ | hello everyone, anyone know if something changed recently in zaptel trunk. Compiles fine, modprobe, then ztcfg shows 31 channels configured, but dmesg says TE4XXP: Span 1 configured for ESF/B8ZS which is for a T1. zaptel.conf and zapata.conf are setup for an E1 so asterisk will not start. |
04:47.13 | _alex_mx_ | reverting to an earlier trunk version works fine |
04:47.29 | Juggie | sure sounds like a bug |
04:47.35 | Juggie | see if you can isolate the exact revision |
04:48.03 | _alex_mx_ | been trying for a week, everything i have downloaded has the same problem :( |
04:48.24 | Juggie | what trunk version works? |
04:49.14 | _alex_mx_ | ouch, how to tell :) |
04:49.18 | P-NuT | CunningPike: What does asterisk addons actually give you? |
04:49.35 | P-NuT | CunningPike: and does it require mysql to run |
04:49.35 | Juggie | alex, show version in console |
04:49.56 | _alex_mx_ | hehe ok, brb need to recompile |
04:50.09 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.65) |
04:50.53 | *** part/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
04:55.01 | spackle | benjamin7062 - get it yet? |
04:55.07 | benjamin7062 | Sadly... yes |
04:55.13 | spackle | eh? |
04:55.38 | benjamin7062 | There is a seperate packages that adds to the source of the wanpipe driver |
04:55.47 | benjamin7062 | which contained wan_ec |
04:56.03 | benjamin7062 | The worst part is... it's in BIG BOLD LETTERS on their site |
04:56.04 | spackle | yeah, I was asking if you got that and installed it yet? |
04:56.20 | spackle | have it working? that sort of thing |
04:56.24 | benjamin7062 | One might think I could read... but perhaps elementary school is where I should start before installing a * box |
04:57.01 | benjamin7062 | Yeah, I recompiled, unloaded drivers... downed the iface, did the oposite.. ztcfg |
04:57.08 | benjamin7062 | and I have crystal clear audio to my cell |
04:57.10 | benjamin7062 | using EC |
04:57.16 | spackle | cool |
04:57.34 | directory | benjamin7062: we seem to attract people who don't... read |
04:57.42 | _alex_mx_ | Juggie: SVN-trunk-r29467M works anything from last week and today does not |
04:57.44 | benjamin7062 | directory, not true |
04:57.50 | benjamin7062 | directory, I read |
04:57.56 | benjamin7062 | directory, if there are big pictures |
04:57.56 | *** part/#asterisk netoguy (n=skelley@ppp-70-129-186-62.dsl.spfdmo.swbell.net) |
04:58.04 | directory | benjamin7062: with pink arrows? |
04:58.12 | benjamin7062 | directory, and voice! |
04:58.20 | directory | good idea |
04:58.22 | benjamin7062 | directory, and water colors involved |
04:58.55 | benjamin7062 | honestly, i skimmed over that several times but it 'looks' at a skim like it's pointing to the same driver |
04:59.07 | benjamin7062 | I thought it was saying something needed to be 'enabled'... which I thought I was doing |
04:59.09 | benjamin7062 | beh |
04:59.32 | directory | it's all good now |
05:00.07 | benjamin7062 | I'm talkin' on it now and it's workin' good |
05:04.00 | CrashHD | what is the 1.2 way of pulling a db variable? |
05:04.16 | CrashHD | ${DB(family/whatever)} ? |
05:08.40 | *** join/#asterisk DjTremors (n=newjacks@stargate.citadelcomputer.com.au) |
05:09.19 | DjTremors | hey all. can anyone help me with a voicemail problem? I must admit, I copied most of * from my server at home to a server here at work. |
05:09.35 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
05:12.51 | [TK]D-Fender | benjamin7062 : sO ALL IS GOOD NOW? |
05:13.02 | [TK]D-Fender | CrashHD : yup |
05:13.03 | *** join/#asterisk cryptnix (n=andrew@64.25.198.123) |
05:13.25 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
05:13.32 | cryptnix | Anyone here familiar with the broadsoft sip service setup? |
05:13.53 | *** join/#asterisk masked (n=masked@ppp66-113.lns1.mel4.internode.on.net) |
05:13.57 | [TK]D-Fender | cryptnix : They have a very comprehensive sample on their site... |
05:14.01 | masked | yo |
05:14.20 | masked | can the spa9000 have multiple connections to a ITSP? |
05:15.17 | cryptnix | ah, well basically i'm trying to connect to a broadsoft enabled server with no luck |
05:15.58 | cryptnix | asterisk -> sip account for my lines |
05:16.09 | cryptnix | pretty much mocking sipura settings ... with no luck |
05:16.21 | cryptnix | yet the sipura's aren't having any issues connecting to the system |
05:19.06 | *** join/#asterisk xbmodder_newlapp (i=nobody@atarack/staff/xbmodder) |
05:20.20 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
05:21.00 | benjamin7062 | [TK]D-Fender, yes, now I have to fix my music on hold |
05:21.03 | benjamin7062 | sounds like a robot |
05:21.10 | benjamin7062 | =) |
05:21.17 | benjamin7062 | but I'm talking on some 'clear' voice |
05:21.27 | *** part/#asterisk _alex_mx_ (n=alex@dsl-200-67-125-45.prod-empresarial.com.mx) |
05:21.42 | FuriousGeorge | if I had 200 sip clients i would probably need SER as a gateway, righ? |
05:21.48 | FuriousGeorge | *right |
05:22.10 | Juggie | not necessairly. |
05:22.31 | Juggie | i would ping jerjer and ask, he doesnt use ser, only static peers |
05:22.32 | Juggie | no realtime |
05:22.48 | DjTremors | can anyone help with voicemail? I'm getting 'WARNING[27435] app.c: No audio available on SIP/225-9032??' |
05:23.04 | DjTremors | i end up with a blank voicemail file. |
05:23.04 | FuriousGeorge | ehh, he's usually cranky at this time |
05:23.21 | FuriousGeorge | on an unrelated note, where does one go to buy 200 PCs |
05:23.51 | drray | FuriousGeorge - dell? |
05:23.58 | FuriousGeorge | drray: makes sense |
05:24.08 | FuriousGeorge | drray: unfortunately |
05:24.10 | spackle | HP, Lenovo? |
05:24.13 | drray | with 200 you could probably send them an image |
05:24.25 | benjamin7062 | FuriousGeorge, NewEgg =) |
05:26.06 | drray | I built, imaged, and tested 75 beige boxes one weekend |
05:26.14 | drray | having done it, I'd say make dell do it |
05:26.37 | tclark | curious if anyone has any wifi handsets in production with symbol ap 100 access point in a large warehouse environment ? |
05:28.04 | tclark | or any other wifi config in a large 250K sq ft home depot type big box envronment ? |
05:28.07 | FuriousGeorge | benjamin7062: i dont think even they carry that kind of stock that i wouldnt limit myself |
05:28.55 | benjamin7062 | FuriousGeorge, I think you're right... =) |
05:29.03 | FuriousGeorge | dell it is i guess |
05:29.18 | spackle | ugh |
05:29.19 | benjamin7062 | Besides, if someone told me to 'build' 200 machines... I'd quit |
05:29.34 | FuriousGeorge | benjamin7062: yeah, i didnt even think about that aspect |
05:29.45 | benjamin7062 | lol |
05:29.48 | drray | you hire a $9 hr screw driver monkey |
05:29.51 | spackle | what is the DOA ratio on Dell? |
05:30.08 | drray | you hire him for a week |
05:30.11 | benjamin7062 | spackle, they are actually pretty good.. but in that volume I've had some bad ones |
05:30.25 | benjamin7062 | Cool thing about dell... you call, they overnight... done |
05:30.28 | drray | the pita is not building the boxes, it's deploying them |
05:30.42 | FuriousGeorge | so whos in NJ |
05:30.53 | benjamin7062 | drray, you can pay dell for that too! =) |
05:30.58 | drray | youget one from dell, and trick it out the way you like, then you send them an ikmage |
05:31.07 | spackle | it's summer, hire a college student |
05:31.11 | drray | then every one you get after comes with it |
05:31.27 | drray | hell, find some punk in another department who thinks he knows IT |
05:32.00 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
05:32.05 | benjamin7062 | drray, those guys can give you excuses. (I have to do my other job) when they realize IT sux |
05:32.28 | drray | a monkey with a screwdriver can do that job |
05:32.43 | FuriousGeorge | knife fighting cable monkey |
05:32.51 | FuriousGeorge | ~furiousgeorge |
05:32.53 | jbot | from memory, furiousgeorge is a knife-fighting monkey last seen with The Man with the Yellow Bat |
05:33.11 | justinu | where is yellow bat man, anyways? |
05:33.14 | FuriousGeorge | jbot: no, furiousgeorge is a knife-fighting (cable) monkey last seen with The Man with the Yellow Bat |
05:33.18 | jbot | FuriousGeorge: okay |
05:33.18 | benjamin7062 | FuriousGeorge, I think you've found your man then... GO into the bathroom... look above the sink |
05:33.30 | FuriousGeorge | lol |
05:33.38 | justinu | ~seen r-evolution |
05:33.51 | jbot | i haven't seen 'r-evolution', justinu |
05:33.51 | P-NuT | has anyone compiled mpg123 on ubuntu dapper? |
05:33.52 | P-NuT | It's not working for em.. |
05:33.59 | justinu | ~seen r_evolution |
05:34.06 | jbot | r_evolution <i=_evoluti@208.251.203.246> was last seen on IRC in channel #asterisk, 15d 10h 3m 38s ago, saying: 'lest i flex my wrists back into a pair of cuffs ;)'. |
05:34.06 | drray | I think monkey with a machine gun would make a fine tv show |
05:34.15 | FuriousGeorge | shoot anything with a monkey... |
05:34.22 | justinu | sheehs, i really hope that's not what happened to him |
05:34.33 | FuriousGeorge | i'd make a show called "the monkies"... unless someone's done it |
05:34.40 | spackle | hehe |
05:34.49 | spackle | hay hay we're the monkeys |
05:34.54 | drray | even if they had, you could just spell it different |
05:35.02 | benjamin7062 | justinu, think positive.. maybe he's kinky |
05:35.06 | FuriousGeorge | settle down there daydream believer :) |
05:35.26 | justinu | benjamin7062: lol, problem is i know him well enough to be concerned |
05:35.30 | benjamin7062 | Stop! In the name of love! |
05:35.56 | benjamin7062 | justinu, Yuk |
05:36.02 | FuriousGeorge | my tunk, my trnk, my lovely asterisk trunk (check-it-out) |
05:36.12 | FuriousGeorge | that never gets old |
05:36.15 | justinu | either way, he'll be ok... i just hope it doesn't take years to find out |
05:36.49 | benjamin7062 | lets talk about pbx baby.. lets talk about you and me... lets talk about all the good things and the..... |
05:37.06 | justinu | benj, your robotic voice problem |
05:37.08 | spackle | ugh |
05:37.13 | FuriousGeorge | i treat it very nice-y, config my SIP device-y |
05:37.22 | justinu | check the RTP packetization value of your SIP devices |
05:37.24 | justinu | should be 20ms |
05:37.44 | justinu | anything else will make weird shit happen, like what you describe |
05:37.52 | FuriousGeorge | wachagonna do with all that zap, all that zap up in your trunk |
05:37.55 | benjamin7062 | my lighter quit working... Now I can't sit here and smoke... I don't think I can stay at work any longer |
05:37.57 | FuriousGeorge | i'll stop |
05:38.05 | justinu | benjamin7062: you can smoke at work? |
05:38.10 | benjamin7062 | justinu, yes |
05:38.17 | FuriousGeorge | i work from home too :) |
05:38.21 | justinu | i can too, but only because I work from home |
05:38.29 | benjamin7062 | no, I work from an office.. we just have smoking |
05:38.34 | justinu | weird, where's that? |
05:38.38 | drray | 1987 |
05:38.41 | FuriousGeorge | mexico? |
05:38.42 | benjamin7062 | Photodex |
05:38.48 | benjamin7062 | Austin, TX |
05:38.51 | FuriousGeorge | photomex? |
05:38.55 | justinu | wow... texas, land of the free |
05:38.58 | drray | you can smoke in austin? |
05:39.02 | drray | I grew up there, |
05:39.02 | benjamin7062 | no |
05:39.06 | benjamin7062 | but in Westlake |
05:39.07 | benjamin7062 | you can |
05:39.10 | benjamin7062 | =) |
05:39.14 | justinu | i hear austin is like, the only nice city in tx |
05:39.18 | benjamin7062 | our company moved because Austin changed there law last year |
05:39.19 | spackle | anybody use fxotune? |
05:39.26 | drray | austin is like the spec on top of the big pile of shit |
05:39.31 | justinu | heh |
05:39.33 | drray | it's the nicest part |
05:39.35 | justinu | so i hear |
05:39.45 | benjamin7062 | justinu, after katrina... it is... They shipped Lousianna to TX, and well, there was a different kind of people there. |
05:40.02 | benjamin7062 | Most didn't come to Austin, thank god |
05:40.04 | benjamin7062 | but some did |
05:40.17 | drray | My mom is lamenting here refugees |
05:40.19 | cryptnix | hmm, seeing 6 TB of ram and 119 processors ... then another shipment of the same the next day is quite invigorating |
05:41.18 | benjamin7062 | drray, Texas is actually pretty bad ass... LOTS of tech here... tons of money... Largest University in the nation (so hot chicks everywhere)... no state tax... and you can carry a gun |
05:41.21 | benjamin7062 | what more could you want |
05:41.23 | benjamin7062 | =) |
05:41.28 | justinu | benjamin7062: your mistake was letting your stog go out before lighting another off i |
05:41.29 | justinu | it |
05:41.41 | spackle | two guns? |
05:41.46 | drray | I spent 26 years inAustin |
05:41.50 | benjamin7062 | justinu, You're right about that.. but I didn't know I was going to lose a lighter |
05:41.52 | drray | I go back every spring |
05:41.57 | justinu | i think they should stop executing people |
05:42.07 | justinu | my biggest beef with texas |
05:42.18 | benjamin7062 | if someone killed your mother for fun.. you'd think otherwise |
05:42.23 | justinu | perhaps |
05:42.25 | drray | no |
05:42.35 | drray | because it would not bring her back |
05:42.36 | justinu | i can't say, because it hasn't happened |
05:43.12 | benjamin7062 | At least here if you 'catch' them.. you can shoot them and it's legal |
05:43.29 | justinu | no issues with that |
05:43.43 | drray | I believe you can do that anywhere |
05:43.57 | benjamin7062 | Not bringing her back is true... But that element of punishment hopefully makes people think twice. the only problem is.. you either have to do it WAY more often.. or not at all.. we have a SOFT death penalty at best. |
05:44.04 | justinu | it doesn't |
05:44.06 | justinu | that's the problem |
05:44.30 | benjamin7062 | It would if we executed in 2 months... EVERYONE that deserved it.. but we don't... there are hardly executions |
05:44.40 | benjamin7062 | It's not a 'fear' factor at all |
05:44.41 | justinu | anyways, this is a slippery slope that we shouldn't get into |
05:45.00 | drray | well, I vote we don't do it at all |
05:45.12 | justinu | agreed, and that's all i have to say about that :) |
05:45.32 | benjamin7062 | true... either you are an eye-for-eye person... or a forgiveness person... but like all things.. I can agree to disagree. =) |
05:45.49 | *** join/#asterisk RoyK[se] (n=roy@svg-acs.ipzone.no) |
05:46.02 | benjamin7062 | As long as you guys aren't white... stupid honkeys |
05:46.04 | benjamin7062 | err wait |
05:46.05 | benjamin7062 | I'm white |
05:46.06 | benjamin7062 | damn |
05:46.22 | benjamin7062 | Guess I'm not racist after all |
05:46.36 | drray | misanthropy is not racism |
05:46.50 | spackle | fxotune? anyone? |
05:48.34 | benjamin7062 | hell yes... boss had lighter fluid and I had a spare zippo around here |
05:48.35 | benjamin7062 | hawt |
05:48.45 | RoyK[se] | morning..... |
05:48.48 | justinu | night guys |
05:48.57 | benjamin7062 | night |
05:53.30 | benjamin7062 | hey.. the only includes that count are the ones from your original context right? so if you include a context with other includes... those aren't relavent? |
05:53.32 | benjamin7062 | is that true? |
05:54.38 | drray | you are speaking of direct references? |
05:55.42 | benjamin7062 | Well, for instance.. you make a sip call... it enters the context specified in sip.conf... lets say that points to => dialout ... but dialout includes other contexts... the dialout contexts aren't relavent? |
05:55.44 | benjamin7062 | right? |
05:55.46 | benjamin7062 | or wrong? |
05:56.22 | benjamin7062 | err dialout 'includes' don't apply |
06:12.45 | SwK | [top] include=> next [next] include => third [third] exten => foo |
06:12.55 | SwK | foo is accessable from [top] |
06:15.51 | benjamin7062 | SwK, That's what I was looking for.. thank you! |
06:15.53 | *** part/#asterisk seb- (n=seb@cpe-72-132-242-171.san.res.rr.com) |
06:16.06 | benjamin7062 | with a lower priority on match... but it 'is' accessible |
06:16.08 | benjamin7062 | kewl |
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06:29.00 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
06:29.14 | *** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
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06:29.37 | *** part/#asterisk littlejohn (n=little@host20-61.pool8711.interbusiness.it) |
06:30.29 | P-NuT | Hey all, moh-native for music on hold. |
06:30.39 | P-NuT | what are the files? |
06:30.41 | P-NuT | mp3's? |
06:30.43 | P-NuT | pcm? |
06:38.14 | *** join/#asterisk n3glv (n=Omega__@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
06:38.22 | kmilitzer | P-NuT: As far as I know it's mp3. At least that's the format that is coming with the asterisk-sources ... |
06:39.10 | n3glv | rehi all |
06:40.30 | n3glv | u taling abt MOH? |
06:40.36 | n3glv | talking |
06:42.56 | n3glv | hey, where is controlled the console behavior? the actvity that scrolls up the screen on the console screen when not logged in? |
06:43.02 | jayb | anyone know any wholesale australian providers with great mobile rates ? |
06:43.26 | n3glv | theres a few aussie sites listed in asteriskguru's list jayb |
06:43.56 | jayb | is that http://www.asteriskguru.com/ ?? |
06:44.04 | n3glv | yes sir |
06:44.17 | n3glv | didn't pay them much attn, since I am in usa |
06:44.27 | jayb | hehe no probs :) |
06:44.27 | jayb | thanks |
06:44.38 | n3glv | just found out that my provider only allows two pstn sessions at a time |
06:44.40 | n3glv | :-( |
06:44.44 | *** join/#asterisk zepmantra (i=mantra@203.215.100.96) |
06:44.46 | n3glv | bummer |
06:45.01 | n3glv | any combo of two, out or in etc. |
06:45.04 | benjamin7062 | PRI it... |
06:45.07 | benjamin7062 | =) |
06:45.13 | n3glv | yeah, for home use... |
06:45.26 | benjamin7062 | oh |
06:45.26 | n3glv | would prob be ok with 4 |
06:45.27 | benjamin7062 | ouch |
06:45.30 | n3glv | 2 sucks |
06:45.37 | benjamin7062 | Get 2 dif. accounts |
06:45.39 | benjamin7062 | =) |
06:45.43 | n3glv | yeah, looking at that |
06:45.55 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
06:45.57 | n3glv | if I buy a DID somewhere, that go's ip to my box yes? |
06:46.06 | benjamin7062 | yup |
06:46.17 | n3glv | and then depending on their system can do x number of inboubnds? |
06:46.18 | benjamin7062 | if you buy it from a sip provider that is |
06:46.30 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
06:46.32 | benjamin7062 | yes, that depends on their system |
06:46.53 | n3glv | like I said, could use like 2 more leggs in |
06:46.58 | benjamin7062 | technically, they can feed you and number of PSTN channels that they want. |
06:47.08 | n3glv | any number |
06:47.14 | benjamin7062 | if yours allows 2... 1 more would work |
06:47.22 | n3glv | well, mine has fail over |
06:47.24 | *** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
06:47.26 | benjamin7062 | just get another account with same provider. |
06:47.34 | n3glv | so if I put the 2nd DID as my fail over..... |
06:47.44 | Tili | i have 2 different IAX2 clients. when i use one I see cli sent to PSTN by * but when I use other I dont see CLI for same account. Note: CLI is set in conf and not on client |
06:48.06 | n3glv | if I get a 2nd DID from same provider, it depends on them if it's separate from the 2 count? |
06:48.26 | benjamin7062 | n3glv, yes... the provider controlls what you can/can't do.. definately not * |
06:48.33 | n3glv | yeah |
06:48.39 | benjamin7062 | * will support whatever they will feed you |
06:48.42 | n3glv | cause axvoice we did like 11 legs, testing |
06:48.49 | n3glv | this one (viatalk) does 2 |
06:49.22 | n3glv | so, I think I need a prepay that allows free incoming |
06:49.28 | n3glv | any suggestions? |
06:49.56 | n3glv | I can set my failover to the prepay incoming |
06:50.03 | benjamin7062 | have you tried http://www.voipdiscount.com/en/index.html |
06:50.07 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
06:51.12 | n3glv | just went there, are they a listing service? |
06:51.16 | n3glv | like pricewatch? |
06:51.37 | benjamin7062 | no |
06:51.40 | benjamin7062 | they provide service |
06:51.41 | n3glv | ol |
06:51.44 | benjamin7062 | but I just checked... |
06:51.44 | n3glv | ooops ok |
06:51.48 | benjamin7062 | may not be USA |
06:51.53 | benjamin7062 | or maybe that is just there free service |
06:52.05 | benjamin7062 | I'm pretty sure that is one of the ones I checked and they did USA but I could be crazy |
06:52.40 | n3glv | yes usa is there |
06:52.45 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
06:53.15 | muppetmaster | So, I have been running v1.2.9.1 for a while and after attending Astricon in London decided to give the SVN Trunk a try. |
06:53.16 | n3glv | since I have an outdial, rartes are not that important, need A:free incoming B:prepay C:usa number |
06:53.27 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
06:53.31 | muppetmaster | But nothing seems to compile... Is this just the state of things, or is there something 'special' I should be taking into consideration? |
06:54.10 | n3glv | so, u guys think someone could load * onto say a HP-Ipac running linux? |
06:54.23 | n3glv | would be a palm top pbx |
06:54.46 | benjamin7062 | n3glv, might be hard to get 'device' management working |
06:54.46 | n3glv | use some wifi phones and make a hell of a demo box for sales |
06:55.00 | n3glv | just wifi sip phones |
06:55.11 | benjamin7062 | ie... don't know how linux runs on a HP Ipac |
06:55.26 | n3glv | neither do I but it would be wild |
06:55.31 | muppetmaster | n3glv We are doing exactly that, with Asterisk inside of a VMWare session creating an ad-hoc WiFi network and then using WiFi SIP Phones. |
06:55.34 | muppetmaster | Works great |
06:55.44 | n3glv | jack into a clients lan with a cable and hand em a couple wifi handsets |
06:55.59 | Zeeek | muppetmaster how was LOndon ast? |
06:56.02 | n3glv | muppet:cool |
06:56.03 | benjamin7062 | The VMWARE thing works 'excellent' |
06:56.18 | n3glv | vmwarea on a M$ box? |
06:56.23 | muppetmaster | Zeeek Was great, except that yesterday the room was an absolute icebox, think everyone will get sick. |
06:56.23 | jayb | benjamin -- what vmware thing ?? |
06:56.36 | jayb | ben - im interested in * in VMWare also... please share |
06:56.52 | Zeeek | muppetmaster it's probably the only time they'll have seen "weather" |
06:56.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
06:56.59 | benjamin7062 | jayb, Running * in vmware works just fine so long as you get your trunking for PSTN via sip provider |
06:57.25 | muppetmaster | benjamin7062 Using bridged networking I have had no problem getting SIP trunks to work with a PSTN interconnect provider |
06:57.27 | n3glv | since there is a vmware for M$ I always ask |
06:57.36 | benjamin7062 | I ran 10 phones for demoing using VMWare |
06:57.59 | benjamin7062 | muppetmaster, well, my point being that you can't use 'hardware' for PSTN...ie, trunk cards through vmware |
06:58.03 | muppetmaster | So, anyone have any ideas on the state of trunk? Zaptel, Asterisk & Asterisk-addons all crap when compiling. |
06:58.33 | muppetmaster | benjamin7062 I have not ever tried that, as not necessary for demo purposes. Although I don't see why Zaptel would not work in a VMWare session if the hardware is present on the box. But will have to take your word for it. |
06:58.35 | n3glv | I am not a coder, does anyone know a wiki or url to upgrade my centos SMP kern? (trixbox zaptel broken) |
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06:59.34 | benjamin7062 | ~centos |
06:59.41 | jbot | centos is probably better than Fedora Core except for that silly bug, see ~centosbug for details |
06:59.53 | benjamin7062 | ~centosbot |
06:59.57 | benjamin7062 | ~centosbug |
06:59.58 | jbot | extra, extra, read all about it, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
06:59.58 | benjamin7062 | damnit |
07:00.01 | n3glv | I just threw hardware at it to get some juice out of it on single kern |
07:00.17 | n3glv | I did that paste and it ignored me |
07:00.23 | benjamin7062 | as root? |
07:00.27 | benjamin7062 | hmm? |
07:00.33 | n3glv | then it puked on the fact that the dir name is diff on the kern |
07:00.43 | n3glv | seems to be the issue |
07:01.15 | benjamin7062 | well, if you have a custom location for some of the above referenced stuff... you may need to modify as necessary |
07:01.17 | n3glv | that actually may be the problem loading the module |
07:01.25 | n3glv | I tihink it's looking in wrong dir for it |
07:02.25 | benjamin7062 | do some of the escape commands and see what they do on your system.. ie `uname -r` |
07:02.29 | benjamin7062 | uname - m |
07:02.38 | benjamin7062 | etc |
07:02.43 | benjamin7062 | make sure the kernels exist where they are suppose to be |
07:02.45 | n3glv | my kern dir is called 2.6.9-34.EL-smp-i686 |
07:03.07 | benjamin7062 | what does uname -r and uname -m return |
07:03.07 | n3glv | and I think it's trying to load from 2.6.9-34.ELsmp |
07:03.18 | benjamin7062 | ahh |
07:03.28 | n3glv | so, the .mo is not found |
07:03.46 | n3glv | on startup (I think that is what I am reading) |
07:04.07 | n3glv | should I see the module file in the kern dir? |
07:04.20 | benjamin7062 | do vi /path/to/spinlock.h then type :.,$ s/rw_lock/rwlock/i |
07:05.01 | x86 | re |
07:05.02 | n3glv | can we see if the module exsists but is in wrong dir? |
07:05.28 | tzafrir | benjamin: why not give a sed -i / ex command and be done with it? |
07:06.14 | benjamin7062 | tzafrir, either way |
07:06.46 | tzafrir | vi /path/to/spinlock.h +'%s/rw_lock/rwlock/' +x |
07:07.12 | tzafrir | (Though IIRC original vi did not support multiple + commands) |
07:07.22 | benjamin7062 | tzafrir, I personally rather be in the file to see what changes and backout if necessary |
07:07.38 | tzafrir | n3glv, ls is known to be handy |
07:07.45 | benjamin7062 | lol |
07:07.45 | tzafrir | find/locate also |
07:08.01 | benjamin7062 | and updatedb <-- or your boxens version |
07:08.10 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
07:08.22 | tzafrir | updatedb should be run nightly |
07:08.43 | tzafrir | and locate's db is hopefully a good approximation |
07:09.20 | benjamin7062 | that makes the assumption no changes were made recently -- and the paste above didn't work so I would imagine something's new or dif on his box |
07:10.36 | benjamin7062 | muppetmaster, I never tested directly access the local machines HW from VMWare, just not sure how you'd add the device to the vmware session .. maybe it's possible these days |
07:11.37 | stoffell | If I wanted to use spandsp 0.0.3pre22, there's no accompanying app_txfax or rxfax ? any idea which one I should use? |
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07:14.05 | littleball | hello, i don't understand how "hint" priority works. who can help? |
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07:17.56 | dangerarea | morning |
07:18.23 | dangerarea | i'm still having issues with ztdummy and my usb-uhci modules |
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07:33.37 | *** join/#asterisk yacyac (n=yac@202.189.231.82) |
07:34.02 | yacyac | hey |
07:34.05 | yacyac | anyone around ? |
07:35.27 | Zeeek | millions of us! |
07:35.35 | yacyac | hahaha |
07:35.47 | Zeeek | well, actually 241 lurkers |
07:35.50 | x86 | TRILLIONS HAHAHAHA ROFLMAO KTHX OMG |
07:35.57 | yacyac | Zeeek... dude... which distro is the best for running asterisk |
07:36.01 | yacyac | hahahaha |
07:36.06 | yacyac | hey x86 |
07:36.09 | x86 | heya |
07:36.11 | yacyac | any suggestion guys |
07:36.20 | littleball | hello, i don't understand how "hint" priority works. who can help? |
07:36.27 | masked | LFS |
07:36.28 | x86 | yacyac: any distro will run asterisk fairly well, at least on x86 or x86-64 hardware |
07:36.47 | x86 | yacyac: stay away from the BSD's and yellow dog linux, etc |
07:37.44 | yacyac | hahaha |
07:37.45 | yacyac | ok |
07:37.51 | yacyac | i was thinking about freebsd |
07:37.52 | yacyac | lol |
07:38.03 | yacyac | i guess i will need to kick it out of my mind then |
07:38.04 | yacyac | lol |
07:38.09 | yacyac | which one do you guys use |
07:38.33 | [hC] | I use debian. |
07:38.49 | stoffell | debian also.. 2 votes.. ;) |
07:39.00 | x86 | i use gentoo and ubuntu |
07:39.03 | [hC] | but, i compile asterisk from source.. |
07:39.08 | [hC] | i dont use debian packages. |
07:39.09 | yacyac | how about arch |
07:39.14 | x86 | yeah, you gotta compile asterisk from source :) |
07:39.24 | [hC] | plain x86 |
07:39.27 | x86 | yacyac: stick with x86 if you can, x86-64 if you must |
07:39.37 | yacyac | i have amd 64 machin |
07:40.01 | yacyac | so which is good 64bit edition of linux ? |
07:40.15 | masked | gentoo? |
07:40.33 | masked | or ubuntu? |
07:40.38 | yacyac | i dont like gentoo and ubuntu |
07:40.48 | masked | neither really |
07:40.50 | Zeeek | slackware |
07:40.50 | masked | use LFS |
07:40.57 | masked | yeah or slack! |
07:40.57 | yacyac | archlinux ? |
07:41.02 | yacyac | brb.... lunch time |
07:41.02 | yacyac | lol |
07:41.42 | [hC] | any of you guys play with hudlite? |
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07:54.35 | X-Rob_ | you stole the match! |
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08:05.37 | littleball | hello, i don't understand how "hint" priority works. who can help? |
08:05.48 | nettie | Hi, since a couple of days asterisk (1.2.6) started crashing like every 2 days. /var/log/asterisk/messages doesnt show anything, other than respawn it in inittab anyone know what I should check/enable to figure out what's broken please? |
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08:16.48 | *** join/#asterisk FreezeS (n=Gladius@82.208.156.94) |
08:16.58 | FreezeS | hello |
08:17.20 | FreezeS | do you know how can I make MoH to start from the beginning for every call ? |
08:19.19 | nettie | FreezeS |
08:19.19 | nettie | exten => 7601,1,Answer |
08:19.19 | nettie | exten => 7601,2,SetMusicOnHold(default) |
08:19.20 | nettie | exten => 7601,3,WaitMusicOnHold(3000) |
08:20.12 | benjamin7062 | Wouldn't you just do MusicOnHold instead of making 'em wait 3000 seconds? |
08:20.20 | benjamin7062 | Ouch |
08:21.40 | drray | or just play an audio file |
08:21.57 | FreezeS | sorry for being so imprecise. How can I do it for a queue ? |
08:23.55 | nettie | benjamin7062 that's my particular case, it's just to give him an idea |
08:26.43 | *** join/#asterisk yacyac (n=yac@202.189.231.82) |
08:27.02 | Zeeek | case dismissed! |
08:27.55 | *** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
08:28.18 | yacyac | Zeeek dude |
08:28.19 | yacyac | i am back |
08:30.26 | *** part/#asterisk x86 (n=x86@p3m/member/x86) |
08:31.18 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
08:31.49 | Zeeek | hey now |
08:32.07 | Zeeek | what was the question? (I have a life as well and it interrupted my IRC career) |
08:32.23 | giesen | man |
08:32.36 | giesen | I just did some of the dirtiest crap to my dialplan that you have ever seen |
08:32.57 | Zeeek | I would win the trohy for the dirtiest dialplan hands down |
08:33.06 | Zeeek | s/trohy/trophy/ |
08:33.14 | giesen | all in the name of setting caller id |
08:33.20 | yacyac | hahaha |
08:33.34 | giesen | well, it's a contextual caller id |
08:33.58 | giesen | so, if it's from an internal extension to one of our employees cell phones |
08:34.09 | giesen | it's set to 000000<ext> |
08:34.24 | giesen | it's a dirty, dirty hack. |
08:34.26 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
08:34.41 | yacyac | how much is the cost of the hardware used in asterisk |
08:34.59 | giesen | haha the best is when I set the caller id for a single extension twice |
08:35.08 | giesen | yacyac: depends what you want to plug it into |
08:35.23 | giesen | a single port analog fxo card is only like $20 |
08:35.31 | yacyac | what are my options |
08:35.33 | giesen | but the pri cards can get pretty pricey |
08:35.50 | yacyac | what does pri cards do |
08:35.53 | giesen | other than that, it's commodity pc hardware |
08:36.05 | giesen | yacyac: if you dont know, then you dont need it |
08:36.18 | giesen | if you just want to plug an asterisk box into an analog phone line |
08:36.25 | giesen | all you need is a single port analog fxo card |
08:36.25 | yacyac | is it pstn to voip adapter ? |
08:36.39 | yacyac | what does single port analog fxo card do |
08:37.02 | giesen | just allows you to plug your asterisk box into a plain jane analog phone line |
08:37.23 | giesen | if you dont even need that much |
08:37.35 | giesen | then all you really need is an ethernet card |
08:37.54 | giesen | in a standard pc |
08:38.05 | yacyac | asterisk handles incomming calls and fwds it ? |
08:38.38 | giesen | if you want to handle incoming (or make outgoing) calls on an analog phone line, you need an analog fxo card |
08:38.46 | yacyac | ok |
08:38.48 | yacyac | kool |
08:39.00 | giesen | if you want to do voip exclusively |
08:39.03 | giesen | then you dont need one |
08:39.18 | yacyac | that will handle 1 call at a time ? |
08:39.28 | giesen | assuming you either have IP phones or ATA (Analog Telephone Adapters) for your existing analog phones |
08:39.32 | giesen | yacyac: yes |
08:39.58 | Zeeek | yacyac what exactly do you want to do? |
08:40.17 | Zeeek | and in what country |
08:40.25 | yacyac | i am in india |
08:40.39 | yacyac | i want to recive and make calls |
08:40.43 | Zeeek | and you want to call/receive calls from where? |
08:40.50 | yacyac | recive and fwd to some other local number |
08:40.52 | yacyac | something like that |
08:40.56 | dangerarea | can anyone help with this please? |
08:40.57 | dangerarea | http://pastebin.com/734452 |
08:41.03 | yacyac | all around the world |
08:41.03 | yacyac | lol |
08:41.09 | Zeeek | rceive on a normal POTS phone line or a new number? |
08:41.48 | yacyac | what is pots phone line |
08:42.00 | giesen | pots = plain old telephone system |
08:42.05 | giesen | basic analog phone line |
08:42.05 | dangerarea | yacyac: a normal analog line |
08:42.11 | yacyac | yes |
08:42.21 | yacyac | i want to transfer to pots |
08:42.43 | giesen | the question is do you want to receive calls on your existing pots line |
08:42.49 | yacyac | yes |
08:42.50 | giesen | or are you fine with getting a new phone number |
08:43.10 | yacyac | yes |
08:43.24 | yacyac | new phone number from the telephone company ? |
08:43.34 | Zeeek | so you need a cheap X100P (used or clone) |
08:43.38 | giesen | well if you went for the second option |
08:43.45 | giesen | you would get the new number from your voip provider |
08:43.52 | Zeeek | that connects to one POTS phone line |
08:44.07 | yacyac | we dont have any voip provider here i guess |
08:44.18 | yacyac | what if i want to do that |
08:44.18 | giesen | there's definitely voip providers operating in india |
08:44.19 | Zeeek | for foreign activity, I'd recommend getting an account in a country where you call |
08:44.43 | Zeeek | or, get Skype ;) |
08:44.51 | yacyac | what if i dont want to involve voip providers then |
08:44.59 | yacyac | plain funda |
08:45.04 | yacyac | one asterisk server here |
08:45.07 | Zeeek | yacyac is this business or pleasure if I may be so indiscreet? |
08:45.27 | yacyac | one asterisk sever any country |
08:45.35 | yacyac | and i just link them uo |
08:45.38 | yacyac | up* |
08:45.42 | giesen | yacyac: you can do that, yes |
08:46.06 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
08:46.19 | yacyac | so from india i can call to pstn line of any other country where my box is |
08:46.40 | yacyac | and anyone can call to my pstn line my box |
08:46.54 | Zeeek | if that box has an FXO and a line or a voIP provider, yes |
08:47.26 | yacyac | something i wanna try out |
08:47.51 | giesen | yacyac: it's probably much more cost effective to get a voip provider in the country you want to call |
08:47.59 | giesen | than to setup an asterisk server in every country |
08:48.06 | yacyac | hmmmm |
08:48.12 | yacyac | what about in long run |
08:48.19 | yacyac | which is more good |
08:48.45 | yacyac | skype and other voip providers are good... how cheap can they go in bulk calling |
08:48.59 | giesen | I pay 1.1c/minute for calls |
08:49.16 | yacyac | kool |
08:49.19 | giesen | within my local calling area |
08:49.27 | giesen | or 2c/minute for anywhere in north america |
08:49.42 | yacyac | giesen ... whats the use of using asterisk if we use skype of anythinng |
08:49.55 | yacyac | kool |
08:49.57 | giesen | what's the use of any pbx |
08:49.58 | yacyac | vonage is good |
08:50.16 | giesen | allows you to share resources and perform cool stuff |
08:50.19 | yacyac | giesen ... sorry but i am new.. and my concepts are not very clear |
08:50.24 | yacyac | thansk |
08:50.50 | giesen | a bunch of people on a pbx can talk between eachother without consuming any resources, they can share phone lines, etc |
08:51.17 | yacyac | hmmm |
08:51.17 | giesen | you can have 100 people sharing 10 phone lines |
08:51.23 | giesen | rather than have one for each person |
08:51.23 | yacyac | kool |
08:51.29 | yacyac | ok |
08:51.54 | giesen | haha unless you work at my company |
08:51.54 | *** join/#asterisk gan- (n=nmuller@195.70.21.58) |
08:51.59 | yacyac | hahahahaha |
08:52.00 | giesen | in which case we need 2 phone lines for every person |
08:52.00 | yacyac | lol |
08:52.04 | yacyac | why |
08:52.15 | giesen | we spend a lot of time on conference calls, etc |
08:52.23 | yacyac | hahaha |
08:52.26 | giesen | so we're sometimes talking to 5 people at a time |
08:52.31 | yacyac | wow |
08:52.32 | yacyac | lol |
08:53.15 | yacyac | so you have a asterisk box up and running ? |
08:53.38 | *** join/#asterisk oscarh (n=oscar@host-87-74-0-243.bulldogdsl.com) |
08:53.58 | giesen | yep |
08:54.01 | gan- | hello. I'm using Asterisk with BRI and an HFC card, and I'm trying to set the MSN for out calls... I was able to do it with CAPI by doing "exten => _XXXXXXXXXX,1,Dial(CAPI/ISDN1/05:${EXTEN})" where "05" is my MSN, but I can't find a way to do the same thing with zaptel. Has anyone managed to do that? |
08:54.48 | yacyac | kool giesen |
08:54.49 | giesen | yacyac: I actually have 2 setup, in an active-standby cluster |
08:54.55 | yacyac | home or office |
08:54.57 | yacyac | kool |
08:55.00 | oscarh | hi, i am having a problem with asterisk + zap + sip. when a SIP phone has called a Zap phone and hung up on it, asterisk does no longer respond to the zap phones DTMF signals :( |
08:55.09 | yacyac | what do you do with 2 then |
08:55.22 | giesen | one is there in case the other one fails |
08:55.32 | yacyac | wow |
08:55.32 | giesen | so if one dies, the other takes over |
08:55.36 | yacyac | koool |
08:55.47 | yacyac | what is your use of asterisk ? |
08:55.47 | [hC] | was there a significant change in the way asterisk spits out manager data in the past couple versions? |
08:55.49 | giesen | and I can take one down for hardware upgrades, etc, without taking down my phone system |
08:56.00 | giesen | yacyac: company pbx |
08:56.07 | yacyac | oh |
08:56.09 | yacyac | koooll |
08:56.29 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
08:56.40 | yacyac | so which voip provider do you use |
08:57.08 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
08:57.10 | yacyac | i have heard alot about vonafe |
08:57.13 | yacyac | vonage |
08:57.26 | benjamin7062 | can * work with vonage? |
08:58.04 | giesen | yacyac: unlimitel.ca |
08:58.15 | giesen | benjamin7062: I think there's some hacks to make it work |
08:58.20 | *** join/#asterisk creadurx (n=creadure@196.82-134-19.bkkb.no) |
08:58.21 | giesen | though I've never tried |
08:58.35 | yacyac | kool |
08:59.03 | yacyac | and is there any setup cost for setting up a voip account with them ? |
08:59.15 | giesen | check the pricing page |
08:59.31 | giesen | for the setup I got, it was a minimum $50 to start |
08:59.45 | giesen | but I got the a la carte package |
08:59.49 | giesen | where I pay per minute |
08:59.58 | giesen | rather than a flat rate per line |
09:00.19 | yacyac | oh |
09:00.19 | yacyac | ok |
09:01.45 | yacyac | giesen which distro have you used for asterisk |
09:01.55 | giesen | gentoo |
09:02.22 | benjamin7062 | They do support * (or provide ATA rather) for bus customers starting at $150 |
09:02.54 | giesen | benjamin7062: either that or you can just use the ata they give you, can get fxo cards for your asterisk box |
09:03.03 | giesen | and use single line analog trunks between them |
09:03.06 | giesen | it's kinda dirty |
09:03.08 | giesen | but it works |
09:03.24 | yacyac | thanks a lot giesen |
09:03.29 | giesen | no problem |
09:03.42 | yacyac | any recommandation on using any specific distro |
09:03.45 | benjamin7062 | yeah... but that's .. well, bleh |
09:03.47 | benjamin7062 | =) |
09:03.56 | giesen | hey, I said it was dirty =) |
09:04.00 | benjamin7062 | I know -- teasing |
09:04.04 | yacyac | gentoo... i dont like it much.... never was able to hold it for long on my machin |
09:04.04 | giesen | almost as dirty as my dialplan |
09:04.17 | giesen | yacyac: have you tried asterisk@home? |
09:04.22 | giesen | it may be to your liking |
09:04.29 | giesen | automates some of the tough stuff in asterisk |
09:04.34 | giesen | with a point and click interface |
09:04.44 | benjamin7062 | yacyac, I have excellent success with debian testing (etch) |
09:04.47 | giesen | it's based off of centos |
09:04.47 | benjamin7062 | 2.6 kernel |
09:04.57 | yacyac | oh |
09:04.57 | yacyac | ok |
09:05.00 | benjamin7062 | trixbox is also something to play with |
09:05.09 | giesen | trixbox = asterisk@home |
09:05.11 | benjamin7062 | (which is asterisk@home) |
09:05.16 | yacyac | i have used debian .. it was too heavy on my laptop |
09:05.17 | yacyac | lol |
09:05.23 | yacyac | so i used arch |
09:05.25 | benjamin7062 | too heavy? |
09:05.28 | giesen | all depends on what you install |
09:05.30 | yacyac | now i am just on this windows |
09:05.30 | yacyac | lol |
09:05.37 | yacyac | yeah i had only 256mb ram |
09:05.47 | yacyac | and i wanted to run gnome |
09:05.47 | yacyac | lol |
09:06.01 | yacyac | arch was pretty fast |
09:06.05 | *** join/#asterisk bmg505 (n=leon@c1-199-1.rndf.isadsl.co.za) |
09:06.16 | giesen | benjamin7062: you have a decent enum macro? |
09:06.30 | giesen | I had to hack the one on voip-info.org up a bit to get it to work reliably |
09:07.02 | yacyac | giesen is it relaibly to use asterisk at corporate level ? |
09:07.11 | benjamin7062 | yacyac, my X session is 160megs right now pushing 6 screens (3 vid cards) ... just fyi |
09:07.29 | yacyac | benjamin7062 what do you run boy |
09:08.01 | giesen | yacyac: lots of people do |
09:08.06 | *** join/#asterisk n3glv (n=Omega__@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
09:08.10 | giesen | if you know what you're doing |
09:08.15 | yacyac | kool |
09:08.26 | giesen | I still have a few kinks Im working out in my system |
09:08.30 | giesen | but it's definitely getting there |
09:08.37 | yacyac | giesen: i am not a pro or something... but i just try to learn |
09:08.45 | yacyac | kool |
09:08.50 | yacyac | what are you trying this time |
09:09.12 | giesen | I've still got some bugs I have to work out with my music on hold and conference calling |
09:09.23 | yacyac | kool |
09:09.30 | giesen | and I dont like the way asterisk does queue |
09:09.32 | giesen | *queues |
09:09.35 | yacyac | why |
09:09.57 | giesen | because of the way it handles penalties |
09:10.05 | yacyac | hmmmmm |
09:10.28 | giesen | I had to use cascading queues |
09:10.40 | giesen | to get "priority" levels to work the way I want |
09:10.40 | yacyac | what is cascading queues |
09:10.45 | yacyac | oh |
09:10.46 | yacyac | kool |
09:10.50 | giesen | for example |
09:11.03 | giesen | Ive got a couple employees who are the front line tech support guys |
09:11.07 | benjamin7062 | giesen, for us, we aren't doing ENUM |
09:11.11 | yacyac | kool |
09:11.11 | benjamin7062 | inbound call center mostly |
09:11.17 | giesen | but if they dont pick up the phone |
09:11.24 | yacyac | oh yes |
09:11.25 | yacyac | i know |
09:11.25 | giesen | I want it to fail over to some other poeple |
09:11.35 | giesen | but asterisk doesnt do it in a useful fashion |
09:11.37 | giesen | at least not for me |
09:12.02 | giesen | so I actually had to create 3 different queues |
09:12.16 | benjamin7062 | giesen, I had the same issue, had to use a combination of Queue(|||timeout) with maxlen, and strategy strict... along with looping in the dialplan |
09:12.18 | giesen | and timeout from one queue to the next |
09:12.49 | giesen | I think I've got a workable solution now |
09:12.53 | yacyac | if i get this asterisk up and running like a sweet cake.... i am gonna approch some small outboud international call center and ask show them how cheap it can be to get on to askterisk |
09:12.57 | giesen | with cascading queues |
09:13.19 | giesen | yacyac: be prepared to spend a LOT of time getting your first asterisk server going |
09:13.25 | yacyac | yeah |
09:13.31 | yacyac | i know that lol |
09:13.45 | bmg505 | Hi guys, any of the dev's here? |
09:13.47 | benjamin7062 | giesen, We ended up writing a server daemon that simply checks the queue status and uses the manager API to add second level agents... HAX! =) |
09:14.02 | giesen | haha |
09:14.04 | giesen | nice |
09:14.08 | giesen | I dont have the resources to do that |
09:14.26 | giesen | and the queue/agent penalties were useless to me |
09:14.33 | yacyac | international call center in india use's one of the best queueing system |
09:14.48 | yacyac | you will never see a call drop in it |
09:14.51 | benjamin7062 | yacyac, spend $500.00 and get two hours of support from digium. They will walk-you-through a base install and build dialplans (the hard part) in front of you answering your questions. It's a RUSH through the more difficult parts. |
09:15.23 | giesen | I pity the fool who has to try and figure out what the hell my dialplan is doing |
09:15.26 | yacyac | benjamin7062: then its no fun .. if they do it for me.... |
09:15.31 | bmg505 | I have back ported the MixMonitor to 1.27.1 and 1.2.9.1 but have a coupel of hassles, any guys here that knows something about the mixmonitor subsystem? |
09:15.43 | benjamin7062 | yacyac, trust me... 2 hours isn't enough to do 'shite' for you. =) |
09:15.44 | bmg505 | MixMonitor for queues |
09:15.50 | yacyac | hahahahaha |
09:16.01 | giesen | yacyac: be prepared to spend like 20 or more hours |
09:16.05 | giesen | figuring stuff out |
09:16.15 | bmg505 | giesen, only 20 hours? |
09:16.19 | giesen | hehehe |
09:16.24 | yacyac | benjamin7062 lets see... first let me get thing started... if i am fucked i will catch hold of them |
09:16.25 | giesen | Ive probably spent about 150 so far on mine |
09:16.25 | benjamin7062 | giesen, Problem is... if you do cascading queues like that.. it breaks hold times I think... avg hold times.. etc.. the stats. |
09:16.31 | yacyac | hahaha |
09:16.31 | giesen | yeah |
09:16.36 | yacyac | time is no problem |
09:16.37 | yacyac | lol |
09:16.42 | giesen | benjamin7062: it's not that important for me right now |
09:16.49 | giesen | to have hold time stats |
09:16.53 | giesen | but it will be in the future |
09:17.22 | benjamin7062 | giesen, the server daemon is easy if you know perl... their manager interface is EASY.. clear text ascii with \r\n terminators |
09:17.30 | giesen | yeah |
09:17.37 | giesen | it sounds like a cool way of doing it |
09:18.03 | benjamin7062 | but I honestly can't confirm if using queue timeouts in fact DOES break avg hold times |
09:18.18 | benjamin7062 | it might build those by channel timers rather than queue timers. |
09:18.33 | giesen | actually, that would be very cool if it did use channel timers |
09:18.36 | giesen | but I doubt it |
09:18.39 | stoffell | benjamin7062, a queue timeout means you bail out of the queue, so you go somewhere else... end of call to that queue |
09:19.06 | benjamin7062 | stoffell, that's what I assumed.. so I just went the daemon route instead of figuring it out |
09:19.08 | benjamin7062 | =) |
09:19.11 | giesen | stoffell: yeah, which is a problem when you use cascading qeues |
09:19.23 | giesen | since the timers are probably reset every time you change queues |
09:19.42 | stoffell | giesen, and (out of interest) cascading queues, would be useful to do ... what ? |
09:19.47 | benjamin7062 | I suspect spillover 'stuff' will be coming soon |
09:20.02 | giesen | stoffell I have different tiers of tech support staff |
09:20.08 | giesen | so if my front-line guys are too busy to pick up |
09:20.11 | giesen | or unavailable |
09:20.11 | benjamin7062 | stoffell -- high priority agents that are only bugged if queue 1 is full |
09:20.13 | stoffell | giesen, yes, they are.. changing queue is going out of 1 and going into a new |
09:20.18 | giesen | it fails over to some other people |
09:20.55 | benjamin7062 | we had the same issue |
09:21.09 | stoffell | so if q1 (first line) is full, the calls go to q2 (2nd line support)... ? so that's correct? |
09:21.15 | giesen | yeah |
09:21.18 | Zeeek | maybe you guys need a real "engine" that can be configured to know which agent is which and choose them intelligently as a function of the origin of the call, the time of day, who's available, etc etc |
09:21.18 | benjamin7062 | yeah |
09:21.18 | yacyac | ;kool |
09:21.26 | giesen | full or unavailable, etc. |
09:21.43 | benjamin7062 | Zeeek, that's what I wrote instead of doing a tiered queuing structure. |
09:21.48 | stoffell | giesen, but the time doesn't get reset then, does it ? it goes straight to q2 ?? |
09:21.51 | benjamin7062 | because of the hold time situation |
09:22.04 | giesen | stoffell: the first queue times out, then it goes into a brand new queue |
09:22.08 | giesen | I dont know if it gets reset or not |
09:22.12 | giesen | I havent tested it yet |
09:22.12 | benjamin7062 | stoffell, you have to inject a call to queue 1 first to check if someone is available |
09:22.16 | giesen | but I would suspect it does |
09:22.43 | stoffell | benjamin7062, yes, correct, but it takes 1 second to see if it its, if it's full, it goes to q2 ... |
09:22.47 | benjamin7062 | right now there is now application or global var that gives status on QUEUE's... so you can't check prior to injection |
09:22.49 | stoffell | if it is ... |
09:22.56 | Zeeek | benjamin7062 so where do we download it? :) Or is this the dirty dialplan? |
09:23.06 | benjamin7062 | stoffell -- you can't use 1 sec.. because the timeout includes call termination |
09:23.20 | stoffell | benjamin7062, ah, yes, indeed.. very correct... good point.. (on having an overall queue status) |
09:23.43 | giesen | I actually like benjamin7062's idea of using the manager interface to do it |
09:23.44 | benjamin7062 | You have to leave the call in the queue long enough to 'try' extensions... |
09:23.45 | stoffell | benjamin7062, but you can say in queues; if full (limited callers), go to q2.. |
09:23.49 | giesen | you might also be able to do it with an AGI |
09:23.57 | benjamin7062 | what I did, since you can't set maxlen to 0.. is set it to 1 instead |
09:24.19 | benjamin7062 | then.. at least... you can't put a call in that queue if someone is already waiting.. so it jumps to 2 |
09:24.22 | benjamin7062 | immediately |
09:24.30 | stoffell | an agi could do it, you could poll the queue and the available agents.. but it's some work/debugging |
09:24.30 | benjamin7062 | and only 1 caller gets screwed with a wait time |
09:24.31 | benjamin7062 | =) |
09:24.45 | *** join/#asterisk mrq1 (n=mrq1@M489P008.dipool.highway.telekom.at) |
09:25.36 | *** join/#asterisk oej (n=olle@bkkb-gw.bitcon.no) |
09:25.37 | benjamin7062 | the manager is inrodes easier to solve this.. because you can simply turn on and off agents in a specific queue based on avg time or whatever you want |
09:25.41 | benjamin7062 | but it's hax |
09:25.41 | stoffell | benjamin7062, waiting is 'good', it's a queue! ;) |
09:26.20 | benjamin7062 | stoffell, you don't waiting if you have free reps (lvl 2 or 3) available... THEN you want to cycle back to queue 1 if all queues are full.. BUT |
09:26.24 | benjamin7062 | it ends up looping |
09:27.02 | stoffell | yep, that's right.. can't do it easily right now.. good idea though.. could use it here myself.. :) |
09:27.10 | benjamin7062 | Basically, right 'now' support for this isn't exactly there in a clean way.. but our company will probably pay to have digium add it so you guys will get the benefits over the next few months |
09:28.48 | benjamin7062 | BTW -- did I mention I LOVE the polycom 601's? |
09:28.49 | benjamin7062 | =) |
09:29.04 | yacyac | what is polycom 601's |
09:29.15 | dlynes_office | benjamin7062: spread the love...donate some polycom 601's to a good cause :) |
09:29.37 | Zeeek | yacyac phones you can't afford |
09:29.40 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
09:29.43 | Bert- | hello there |
09:29.50 | yacyac | Zeeek and they are |
09:30.03 | Zeeek | Very nice SIP phones |
09:30.17 | benjamin7062 | We evaluated Cisco 7960, 7970, Snom, and the Polycom 601 to deploy ... the polycom's won the battle... Mini browser can hook into the manager API (Cisco can kinda support this), sound quality is great, can configure dialplans in the phone that make it act 'just' like most digital systems |
09:30.22 | Bert- | I have this card : VIC2-2BRI-NT/TE (cisco). It is a 2 port card. Txo ports means 2 lignes or 2x2lines plz ? |
09:30.47 | Zeeek | benjamin7062 tell us about manager<--> Polycom |
09:30.52 | Zeeek | 601 |
09:31.11 | yacyac | hey all cards and everything works in india ? |
09:31.14 | drray | the polycom's are ugly |
09:31.16 | drray | IMO |
09:31.23 | Zeeek | get out! |
09:31.25 | benjamin7062 | Zeeek, sky is the limit... write a CGI that converts manager API to miniture version of xhtml... get 'whatever' you want |
09:31.30 | dlynes_office | yacyac: i would imagine so, yeah...they work in Lahore |
09:31.36 | yacyac | i had heard that most of the card cant |
09:31.45 | yacyac | dlynes_office : where you from |
09:31.51 | benjamin7062 | for instance, our call center manager can see all agents on calls from his mini browser... select an agent and start the call barge from the web browser |
09:31.51 | dlynes_office | yacyac: at least the two port digium cards do |
09:32.05 | *** join/#asterisk NLinington (n=nfl@82-69-27-212.dsl.in-addr.zen.co.uk) |
09:32.09 | yacyac | kool |
09:32.11 | dlynes_office | yacyac: I'm from vancouver, canada, but there's two or three guys on here from Pakistan |
09:32.18 | giesen | benjamin7062: that sounds really cool |
09:32.23 | yacyac | kool |
09:32.24 | giesen | I may do something like that for my cisco phones |
09:32.40 | dlynes_office | yacyac: you're setting up routes in India? |
09:32.41 | giesen | so I can manage some stuff that I can do with the SIP loads on them |
09:32.42 | benjamin7062 | Zeeek, i have the 301.. also, same quality... but obviously lacking functionality |
09:32.52 | giesen | *cant do |
09:33.04 | Zeeek | yep, sad ain't it? We're in the ghetto of Polycom low rent |
09:33.07 | yacyac | yep ... setting up a asterisk server in bombay |
09:33.10 | giesen | and chan_sccp is just dirty |
09:33.21 | benjamin7062 | giesen, cisco does 'some' of this depending on whether you are running sccp or sip... btw... if you wanna crash *.. create a queue... call it from sccp... terminate it to SIP... POOF! |
09:33.27 | dlynes_office | yacyac: ah...will you be doing anything in Punjab/Amritsar? |
09:33.34 | Zeeek | yacyac so the club would be basterisk ? (like bollywood) |
09:33.43 | giesen | benjamin7062: yeah, that's part of the reason I wont touch chan_sccp |
09:33.48 | giesen | that and the lack of dynamic reloads |
09:33.49 | benjamin7062 | yup |
09:33.59 | giesen | taking down my whole box to make a change to a phone |
09:34.04 | giesen | is not my idea of a feasible system |
09:34.17 | dlynes_office | benjamin7062: apparently chan_skinny is a lot more stable |
09:34.18 | benjamin7062 | =) |
09:34.33 | giesen | chan_skinny doesnt have the functionality |
09:34.36 | dlynes_office | benjamin7062: and if you're having problems with it, there's an active developer for chan_skinny |
09:34.45 | dlynes_office | benjamin7062: it's Qwell/Qwell[] |
09:34.50 | benjamin7062 | dlynes_home, yeah, but we tried that too... and, well, Digium was on my box when it crashed... I laughed... he sighed... we moved on |
09:35.04 | dlynes_office | cool |
09:35.13 | benjamin7062 | He logged it though |
09:35.14 | yacyac | dlynes_office i dont know |
09:35.16 | benjamin7062 | just a bug of some sort |
09:35.34 | yacyac | dlynes_office i have a friend who is there in amritsir who is going to help me |
09:35.44 | dlynes_office | benjamin7062: yeah...seems those intel epro100's don't just have a problem with digium cards |
09:35.55 | benjamin7062 | We decided away from the cisco due to the lack of xml support in the SIP images... |
09:35.56 | dlynes_office | benjamin7062: their drivers seem to be buggy in general |
09:36.09 | benjamin7062 | It works... but to a lesser degree per my reading of other's experiences |
09:36.15 | dlynes_office | yacyac: can you talk to him about maybe setting up a server in Amritsar? |
09:36.23 | yacyac | sure |
09:36.24 | yacyac | why |
09:36.24 | dlynes_office | yacyac: if you could set one up there, we'd love to peer with you |
09:36.25 | *** part/#asterisk mrq1 (n=mrq1@M489P008.dipool.highway.telekom.at) |
09:36.32 | benjamin7062 | Sangoma all the way... |
09:36.43 | yacyac | dlynes_office : we as in ? |
09:36.43 | benjamin7062 | The EC on this Sangoma is 'perfect' |
09:36.49 | giesen | benjamin7062: the sip loads do have xml |
09:36.53 | giesen | it's just somewhat broken |
09:37.05 | benjamin7062 | giesen, yeah -- that's what I meant. It's minimal support |
09:37.06 | dlynes_office | benjamin7062: actually running a sangoma card on a box with two epro100's, and got a kernel trap after three hours |
09:37.28 | benjamin7062 | one of the guys on here has great insite into it... they use only cisco.. he swears by 'em but acknowledges some of the bugs |
09:37.30 | dlynes_office | yacyac: our company; we provide voip service and pstn in Vancouver, Canada |
09:37.35 | benjamin7062 | xhtml is 'so' much easier to deal with |
09:37.39 | dlynes_office | yacyac: That's close to Surrey, Canada |
09:37.53 | yacyac | dlynes_office : cool |
09:37.53 | giesen | dlynes_office: you're in surrey? |
09:37.54 | benjamin7062 | dlynes_home, shush, that won't happen to me |
09:38.12 | dlynes_office | giesen: no, but we plan to start marketing in surrey to businesses shortly |
09:38.16 | giesen | ah |
09:38.19 | giesen | we're in toronto |
09:38.20 | yacyac | dlynes_office you already have a peer in bombay/mumbai ? |
09:38.20 | dlynes_office | giesen: the owner of our company is Indian |
09:38.28 | yacyac | oh |
09:38.29 | dlynes_office | yacyac: nope...don't need one atm, either |
09:38.34 | yacyac | hahaha |
09:38.43 | yacyac | dlynes_office whats your companys name |
09:38.48 | dlynes_office | yacyac: All of our Indian customers are from Punjab, or are of Punjabi descent |
09:38.52 | benjamin7062 | Is that how everyone offers so much 'free' local calls? Just a bunch of cooperative peers? |
09:39.01 | dlynes_office | yacyac: and even then, 95% of them are Jutts |
09:39.20 | dlynes_office | yacyac: 24/7 Communications |
09:39.25 | yacyac | dlynes_office: hahahhaha lol... |
09:39.30 | yacyac | got a site to look at ? |
09:39.37 | dlynes_office | yacyac: www.247communications.com |
09:39.42 | yacyac | dlynes_office are you an indian or canadina |
09:39.52 | dlynes_office | yacyac: I'm Canadian; the owner is Punjabi |
09:40.02 | dlynes_office | yacyac: but he moved here when he was 2 or 3 from Amritsar |
09:40.06 | yacyac | oh |
09:40.07 | benjamin7062 | I'm mutt? is that cool? |
09:40.08 | yacyac | kool |
09:40.36 | dlynes_office | yacyac: He's a Canadian citizen, but originally from Amritsar |
09:40.37 | yacyac | dlynes_office : if i setup a machin over there... will that guy who runs the machin will get any benifit ? |
09:40.53 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
09:41.06 | dlynes_office | yacyac: yeah...basically we can provide local dial tone in Vancouver for you, in exchange for local dial tone in Punjab for us |
09:41.37 | dlynes_office | yacyac: so that would even allow you to call north american toll-free numbers that are callable from Vancouver |
09:41.44 | benjamin7062 | dlynes_home, is that how everyone does it? A shit ton of peers? |
09:41.47 | yacyac | kool |
09:41.54 | dlynes_office | benjamin7062: I doubt it |
09:42.00 | benjamin7062 | oh |
09:42.06 | dlynes_office | benjamin7062: dundi |
09:42.26 | benjamin7062 | I read about that |
09:42.36 | dlynes_office | benjamin7062: dundi is for sharing your local extensions with the world |
09:42.40 | benjamin7062 | basically, you agree to allow people out to the PSTN through you.. if you can go through them? |
09:42.43 | dlynes_office | so that if anyone needs to call one of your customers |
09:42.43 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
09:42.55 | dlynes_office | they just dial into your switch, and dial them as a sip extension |
09:43.03 | dlynes_office | so the call never goes to pstn |
09:43.09 | benjamin7062 | ahhhhhhh |
09:43.11 | benjamin7062 | GOTCHA |
09:43.17 | dlynes_office | but that won't work for peering (pstn) |
09:43.32 | dlynes_office | that you need to set up a manual agreement for |
09:43.32 | benjamin7062 | wouldn't that be cool though. |
09:45.06 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
09:46.49 | benjamin7062 | Just wait till Dundi is 'huge'... |
09:47.10 | benjamin7062 | The US Gov... will attack it like napster for 'stealing' telecom from the bells.. =) |
09:48.40 | drray | I don't get dundi |
09:49.36 | benjamin7062 | Do you understand BGP? |
09:49.47 | benjamin7062 | It's similar |
09:50.26 | benjamin7062 | I thought it allowed PSTN sharing.. but apparently it's just peer->peer dial route propagation |
09:51.17 | benjamin7062 | if you are a dundi peer.. I'm a dundi peer.. if you call my phone number... it never hits the PSTN... it goes straight to me via SIP (or whatever) |
09:51.37 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
09:53.14 | *** join/#asterisk bobman (n=bobman@24-53-5-197.agstme.adelphia.net) |
09:53.56 | n3glv | what's dundi? |
09:54.18 | n3glv | if the gov will hate it, I will prob love it |
09:54.36 | n3glv | is that like sipbroker? or more like ENUM? |
09:54.40 | n3glv | sounds like enum |
09:56.06 | benjamin7062 | enum |
09:56.08 | benjamin7062 | pretty much |
09:56.13 | benjamin7062 | only distributed |
09:56.24 | n3glv | hmmm |
09:56.31 | n3glv | sounds like it's up my ally |
09:56.46 | n3glv | they do any iax support? u got a url? |
09:57.17 | n3glv | been wanting to get sipbroker running |
09:57.21 | benjamin7062 | http://www.dundi.com/dundi.txt |
09:57.42 | n3glv | enum never called me back to ver my number, but enum trunk works |
09:57.53 | L|NUX | for DUNDi need peering :( |
09:58.11 | benjamin7062 | true |
09:58.24 | benjamin7062 | but 'anyone' can be a peer |
09:58.33 | n3glv | hmm |
09:58.41 | L|NUX | well |
09:58.44 | benjamin7062 | Not that it's the only or best way.. but it's creative |
09:58.45 | benjamin7062 | =) |
09:58.49 | n3glv | they implement dunid in * yet? |
09:58.50 | L|NUX | if he/she allow your machine :) |
09:59.02 | benjamin7062 | dundi was written by the same guy that wrote * |
09:59.09 | n3glv | cool |
09:59.19 | n3glv | they got a wiki I wonder |
09:59.19 | benjamin7062 | Mark Spencer / Digium |
09:59.35 | AltnTab | Why Asterisk could not find recorded files from Monitor() thru external encoder executed with System() |
09:59.45 | AltnTab | the output path is correct shoun in CLI |
09:59.49 | benjamin7062 | Anyone know why my hold music sould sound like an alien sending static distortion? |
09:59.51 | AltnTab | err shown |
09:59.58 | benjamin7062 | or.. hmm, maybe that's just what it is? |
10:00.07 | n3glv | I have a dream of a Ham Radio voip that uses a peering cached callsign to ip system |
10:00.51 | benjamin7062 | Ham over Ethernet? |
10:00.54 | benjamin7062 | HoE? |
10:01.01 | n3glv | ben, is your zaptel stuff working? ztdummy etc? |
10:01.16 | n3glv | haha echolink.org for one |
10:01.22 | n3glv | eqso for another |
10:01.27 | Zeeek | n3glv the Digium distrubutor in Paris is a ham |
10:01.31 | n3glv | beeing done, but not very robust to failure |
10:01.37 | NLinington | Can somebody give me a hand getting incomming faxing working with app_rxfax. Everything looks like its working ok but faxes being received always fail. |
10:01.45 | dlynes_office | dundi and enum are almost the same thing |
10:02.20 | dlynes_office | just basically two competing standards, from what I understand |
10:02.20 | benjamin7062 | Hmm, perhaps that's my problem... I'm using a PRI... so figured timing would be handled by the card? |
10:02.20 | benjamin7062 | maybe not |
10:02.20 | Zeeek | NLinington it may be a question of version. About 2 out of three fail for me |
10:02.27 | Zeeek | on X100P |
10:02.37 | dlynes_office | NLinington: app_rxfax/app_txfax are both highly unreliable |
10:02.43 | n3glv | do a lsmod and make sure that the zaptel dummy is there |
10:02.48 | dlynes_office | Zeeek: you actually get it to work at all on an x100p? |
10:02.51 | n3glv | I'm no expert |
10:02.52 | dlynes_office | Zeeek: you must be blessed |
10:03.10 | n3glv | but I know MOH and Meetme and some other stuff relies on zaptel or ztdummy |
10:03.20 | dlynes_office | benjamin7062: is your pri working fine? |
10:03.38 | NLinington | I have a X100P and zaptel dummy is not loaded, do I need this? |
10:03.50 | dlynes_office | NLinington: no...don't load it |
10:03.59 | dlynes_office | NLinington: ztdummy is if you don't have another zaptel driver loaded |
10:04.00 | benjamin7062 | dlynes_office, yeah -- can make receive calls fine.. loaded ztdummy (modprobe) just to try it.. still static. |
10:04.02 | benjamin7062 | poop |
10:04.12 | benjamin7062 | sounds aweful |
10:04.14 | n3glv | ok |
10:04.17 | n3glv | sorry |
10:04.21 | dlynes_office | benjamin7062: yeah...you only need wct4xxp loaded or whatever...don't load ztdummy |
10:04.27 | Zeeek | Murphy's law: ALL spam faxes work impeccably! Only my customers faxes can't talk to Spandsp. the channel just tries for a while and drops. Once it didn't drop for 5 hours! |
10:04.30 | dlynes_office | benjamin7062: ztdummy will just interfere |
10:04.34 | n3glv | I am trying to compile new kernel, getting hardware errors |
10:04.35 | benjamin7062 | Not your fault |
10:04.45 | n3glv | new (used) parts here |
10:05.12 | n3glv | seems to get a bit farther each time |
10:05.16 | n3glv | so just hammering on it |
10:05.21 | benjamin7062 | actually.. nothing like wct is loaded either |
10:05.25 | n3glv | may be a ram problem |
10:05.25 | benjamin7062 | perhaps that is it |
10:05.34 | dlynes_office | benjamin7062: wait |
10:05.41 | dlynes_office | benjamin7062: you're using sangoma not digium, right? |
10:05.44 | NLinington | so whats the best ways to get faxing working via * |
10:05.46 | benjamin7062 | yes sir |
10:05.52 | dlynes_office | benjamin7062: yeah...one second |
10:06.04 | n3glv | NL I think the fax machine has to be a certain spec |
10:06.15 | n3glv | support some protocol |
10:06.20 | n3glv | read that somewhere |
10:06.27 | dlynes_office | n3glv: 9600 baud or lower for app_?xfax.so |
10:06.34 | benjamin7062 | t.136 or t.36.. or something |
10:06.58 | n3glv | yeah, knew I saw someting on that |
10:07.09 | benjamin7062 | zaptel and wanpipe modules loaded |
10:07.11 | NLinington | but it should handshake to that level? |
10:07.12 | n3glv | not high priority on my box |
10:07.19 | dlynes_office | benjamin7062: you should have af_wanpipe, wanpipe, wanrouter, zaptel, crc_ccitt, and sdladrv loaded |
10:07.56 | dlynes_office | n3glv: anyways, if you want something more reliable |
10:08.07 | dlynes_office | n3glv: use iaxmodem in conjunction with hylafax |
10:08.07 | benjamin7062 | thank you for that list |
10:08.10 | benjamin7062 | all okay |
10:08.11 | n3glv | it's 6am and my stomach is trying to talk me into baking a frozen pizza |
10:08.27 | benjamin7062 | Timing on both PRI channels set to 'NORMAL'.. cards work |
10:08.28 | benjamin7062 | hmmph |
10:08.46 | benjamin7062 | the mp3 player is build into * right? |
10:08.49 | dlynes_office | benjamin7062: wanrouter status -> CLK: EXT? |
10:09.26 | NLinington | ok I am going to have a go with iaxmodem again, last time I tried it looked like it had the same problem. |
10:09.28 | benjamin7062 | dlynes_office, yeah.. both PRI's.. |
10:09.58 | benjamin7062 | NLinington, I am going to try the Linksys analog->sip thing.. I'll let you know how many fail for us in a few days |
10:10.32 | n3glv | linksys PAP2 ? |
10:10.51 | benjamin7062 | yes sir |
10:11.12 | benjamin7062 | I don't have high hopes based on reading |
10:11.20 | benjamin7062 | but hell, it's $60.00 .. might as well try it |
10:11.43 | NLinington | benjamin7062, the idea was to get the fax to an e-mail, so connecting a fax machine to a sip adapter was not something I wanted to do. |
10:11.45 | benjamin7062 | if it sux.. we'll hang it on the wall and watch the blinky lights |
10:12.12 | benjamin7062 | NLinington, ahhh -- touche.. efax. =) |
10:12.37 | n3glv | well. that big ass V company uses a lot of PAP2's |
10:12.54 | n3glv | I have no 1st hand info on reliability |
10:13.04 | benjamin7062 | I will in a few days |
10:13.09 | dlynes_office | benjamin7062: your span line looks like span=1,1,0,esf,b8zs? |
10:13.29 | benjamin7062 | span=1,1,0,esf,b8zs |
10:13.29 | benjamin7062 | span=2,2,0,esf,b8zs |
10:13.29 | benjamin7062 | dchan=24,48 |
10:13.29 | benjamin7062 | bchan=1-23,25-47 |
10:14.21 | benjamin7062 | Maybe I need to specify somewhere to use 'one' for timing |
10:14.24 | dlynes_office | benjamin7062: what kinda chip is on the machine? |
10:14.37 | benjamin7062 | AMD 4000 i think |
10:14.45 | dlynes_office | and you've got plenty of memory i'm guessing? |
10:15.09 | benjamin7062 | Only 1gig.. but 900free |
10:15.11 | n3glv | I have heard that * is not very good on non intel |
10:15.23 | dlynes_office | n3glv: you heard wrong |
10:15.27 | benjamin7062 | n3glv, nonsense.. * lives in user land |
10:15.33 | dlynes_office | n3glv: lots of people are running fine on amd's |
10:15.37 | n3glv | ok |
10:15.46 | dlynes_office | n3glv: same for zaptel |
10:15.50 | n3glv | cool |
10:16.05 | dlynes_office | n3glv: however zaptel doesn't work well on non-Linux OS's |
10:16.11 | n3glv | just heard that tonight |
10:16.13 | dlynes_office | n3glv: or is completely non-existent |
10:16.17 | benjamin7062 | n3glv, expecially if you use a generic 386ish build on your kernel... the apps wouldn't know the difference. |
10:16.25 | dlynes_office | n3glv: it's only been ported to FreeBSD |
10:16.35 | n3glv | well, my SMP kern won't run ztdummy |
10:16.38 | dlynes_office | n3glv: and FreeBSD's porting effort is only considered beta quality at best |
10:16.47 | n3glv | hence my attempt at building a new kern |
10:16.56 | benjamin7062 | n3glv, SMP breaks a lot in linux sometimes.. hehe |
10:17.05 | n3glv | yeah |
10:17.22 | dlynes_office | benjamin7062: peoples' inability to install the right packages is what breaks in linux sometimes, not smp :) |
10:17.38 | n3glv | ran me around in circles untill I found u guys and found out about ztdummy needed for meetme |
10:17.44 | dlynes_office | n3glv: paste uname -a's output to the channel |
10:18.08 | n3glv | it's a trixbox, well known issue I think |
10:18.25 | dlynes_office | trixbox still runs asterisk under-the-hood |
10:18.32 | benjamin7062 | dlynes_office, I agree... but I know, for instance, a smp kernel with threaded perl can behave strangly ... and I've had forked c++ go nuts a couple times... with no explanation. Probably idiot developer (me) |
10:18.36 | dlynes_office | trixbox's config files are just all fubar |
10:18.53 | dlynes_office | benjamin7062: i would imagine it's the code, not the smb |
10:18.54 | benjamin7062 | trixbox is * with all the crap loaded for you... on CentOS... |
10:18.55 | dlynes_office | benjamin7062: i would imagine it's the code, not the smp |
10:19.07 | n3glv | Linux asterisk1.local 2.6.9-34.EL #1 Wed Mar 8 00:07:35 CST 2006 i686 i686 i386 GNU/Linux |
10:19.23 | dlynes_office | n3glv: ummm....that's not an SMP kernel |
10:19.36 | benjamin7062 | n3glv, normally smp kernels specify |
10:19.36 | n3glv | not running both cpu-s right now |
10:19.46 | n3glv | u want kern ver of smp kern? |
10:19.47 | benjamin7062 | dlynes_office, STOP TYPING FASTER THAN ME! o.O! |
10:19.50 | dlynes_office | n3glv: doesn't matter...it's not an smp kernel |
10:19.54 | benjamin7062 | hehe |
10:20.03 | dlynes_office | n3glv: what kernel-dev do you have installed? |
10:20.08 | *** join/#asterisk ASTERISKNEWBIEXX (n=chatzill@rrcs-67-52-187-18.west.biz.rr.com) |
10:20.18 | n3glv | 2.6.9-34.EL-smp-i686 |
10:20.26 | *** join/#asterisk SHad|Work (n=kvirc@84.255.228.2) |
10:20.37 | benjamin7062 | hmm? |
10:20.43 | dlynes_office | n3glv: that's your smp kernel version, or the version of kernel-dev you have installed? |
10:20.45 | n3glv | that is kern that was failing |
10:20.54 | SHad|Work | good day |
10:20.55 | dlynes_office | n3glv: what kernel-dev version do you have installed? |
10:20.59 | n3glv | how I find that? |
10:21.13 | dlynes_office | n3glv: beats the funk out of me...I don't use an rpm-based distro |
10:21.17 | ASTERISKNEWBIEXX | can someone fix my asterisk system ? Ill pay them some money if they can do it now. |
10:21.19 | SHad|Work | does anyone here have any experience with configuring multiple zaptel cards in a single server? |
10:21.23 | benjamin7062 | rpm -<some arguments> |
10:21.50 | benjamin7062 | ASTERISKNEWBIEXX, what's wrong? |
10:21.59 | dlynes_office | ASTERISKNEWBIEXX: type /nick asterisknewbiexx before I go blind, please |
10:22.07 | n3glv | yeah, I prefer debian most of the time |
10:22.19 | benjamin7062 | dlynes_office, What did you just say, ? I couldn't see it? |
10:22.36 | n3glv | haha, yeah, shouting |
10:22.37 | benjamin7062 | my eyes hurt |
10:22.40 | benjamin7062 | ouch |
10:22.46 | *** join/#asterisk MarcPtz (n=MarcPtz@18.Red-80-35-146.staticIP.rima-tde.net) |
10:23.03 | dlynes_office | ~suggestions |
10:23.08 | jbot | it has been said that suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be ... |
10:23.15 | benjamin7062 | Man, if I were hold music on my * box... I would wanna play clear... this hold music is rude |
10:23.34 | dlynes_office | benjamin7062: i'm thinking you've got other issues, though |
10:23.44 | dlynes_office | benjamin7062: i don't think it's a question of your drivers |
10:23.56 | dlynes_office | benjamin7062: what else are you running on that box besides asterisk? |
10:23.59 | n3glv | I finally managed (don't know how) to get my box to play something else (cat stevens) |
10:24.18 | benjamin7062 | dlynes_home, nothing.. clean install.. tftp, ftp, ssh, normal kernel poop, cron, etc |
10:24.38 | dlynes_office | hrm |
10:25.05 | dlynes_office | asterisknewbiexx: now, what seems to be your problem? |
10:25.14 | benjamin7062 | can you call US 800 numbers toll free? |
10:25.27 | asterisknewbiexx | i messed up my conf files |
10:25.28 | dlynes_office | benjamin7062: who? |
10:25.50 | dlynes_office | asterisknewbiexx: freepbx/trixbox/asterisk@home/amp? or regular asterisk? |
10:25.52 | n3glv | ben, I can |
10:25.53 | benjamin7062 | dlynes_home, you... you can hear it by calling 877-87-photo... |
10:26.02 | benjamin7062 | i'll have to go figure what those numbers are... |
10:26.13 | AltnTab | How come asterisk cannot find a file when the wxact location is specified !? |
10:26.23 | benjamin7062 | 8778774686 |
10:26.25 | n3glv | 48686 |
10:26.31 | dlynes_office | AltnTab: maybe your permissions are screwy? |
10:26.44 | n3glv | ivr is fine |
10:26.46 | asterisknewbiexx | fedoracore 4 /asterisk 1.2.9/astguiclient-1.1.11/vicidialer |
10:26.47 | benjamin7062 | AltnTab, Sometimes it's relative path also |
10:26.56 | AltnTab | dlynes_home, the permissions are default ones on Monitor() |
10:26.58 | dlynes_office | asterisknewbiexx: and which particular config files are screwed up? |
10:27.17 | dlynes_office | AltnTab: those aren't the ones I was talking about...those will only make it so it can't read it, not so it can't find it |
10:27.25 | n3glv | got female |
10:27.35 | n3glv | talking about call something for live talk |
10:27.38 | dlynes_office | AltnTab: i was thinking more that you might not have execution bits on one of the parent directories |
10:27.39 | n3glv | sounds fine |
10:27.47 | asterisknewbiexx | sip.conf ,iax.conf ,zapata.conf |
10:27.58 | benjamin7062 | WHAT?.. really.. one sec |
10:28.04 | AltnTab | dlynes_home, could be, i'll check |
10:28.18 | dlynes_office | The number you have called cannot be reached from your calling area |
10:28.22 | dlynes_office | bastards |
10:28.43 | dlynes_office | I guess you don't want any business from Canada :p |
10:28.54 | n3glv | haha, sounds 100% on my cell |
10:29.00 | benjamin7062 | damnit.. gave you wrong number.. 877(3)7photo |
10:29.09 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
10:29.09 | n3glv | if it's supposed to do a commercial for live talk that is |
10:29.15 | n3glv | haha |
10:29.17 | benjamin7062 | No, i'm just dumb |
10:29.39 | asterisknewbiexx | think I messed up my zap channnels |
10:29.55 | dlynes_office | "The number you have dialed is not available in your area at this time." |
10:29.56 | benjamin7062 | hmm |
10:30.22 | n3glv | can u do that one numeric too? |
10:30.59 | benjamin7062 | weird... our 800 number goes to some random guy |
10:31.04 | benjamin7062 | i'm pretty sure that's broken |
10:31.07 | benjamin7062 | damnit |
10:31.13 | n3glv | guy answered, had no idea what i was talking about |
10:31.44 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
10:31.48 | n3glv | do u have a failover set, like a no answer or busy number? |
10:31.50 | benjamin7062 | OMG -- our 877 number must forward to our president if our lines are broken |
10:31.57 | n3glv | YEAH |
10:32.00 | n3glv | I bet |
10:32.00 | benjamin7062 | i didn't set it up.. I wouldn't know |
10:32.04 | benjamin7062 | that's hawt |
10:32.09 | benjamin7062 | I just had you spam my boss... |
10:32.11 | n3glv | heeheehee |
10:32.13 | benjamin7062 | heh |
10:32.14 | benjamin7062 | cool |
10:32.15 | n3glv | woke his ass up I bet |
10:32.20 | benjamin7062 | it's only 5:30 am |
10:32.24 | benjamin7062 | better wake his ass up |
10:32.30 | n3glv | hope I didn't get u in trouble |
10:32.36 | benjamin7062 | nah |
10:32.39 | benjamin7062 | he'll live |
10:32.43 | n3glv | I said trying to test an asterisk box |
10:32.57 | benjamin7062 | weird that it goes to him |
10:33.02 | n3glv | pretty funny though |
10:33.06 | benjamin7062 | makes me wonder if perhaps ONE of our pri's is broken |
10:33.10 | n3glv | could be |
10:33.19 | benjamin7062 | they both show okay though |
10:33.21 | benjamin7062 | so weird |
10:33.58 | benjamin7062 | our direct dial works fine |
10:34.28 | n3glv | who feeds you your did? |
10:34.38 | benjamin7062 | time warner telecom |
10:36.05 | n3glv | ok, and it was working? just crappy singnal to noise ? |
10:36.09 | MarcPtz | Hi all , I'm trying to get the ANSWEREDTIME & DIALSTATUS variables after each CALL inside an AGI script,If the channel is not hangup by the side that originated the call all works fine and I get correct values otherwise they return empty strings , is this a know bug or I'm missing something? |
10:36.20 | benjamin7062 | n3glv, yup |
10:36.26 | benjamin7062 | calls sound excellent |
10:36.34 | benjamin7062 | hold music sounds like ass in a tunnel |
10:36.48 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
10:36.50 | n3glv | u have some echocancelation stuff turned on? |
10:36.56 | benjamin7062 | yes |
10:37.05 | n3glv | try with it off? |
10:37.08 | benjamin7062 | does that break hold music? |
10:37.17 | n3glv | grasping here |
10:37.31 | n3glv | but should or could affect audio qual |
10:37.58 | n3glv | starting to get into my area, (live sound engineer here) |
10:40.03 | benjamin7062 | whew |
10:40.05 | benjamin7062 | okay |
10:40.08 | benjamin7062 | same without EC |
10:40.10 | benjamin7062 | I'm glad |
10:40.16 | benjamin7062 | cause that would have sucked if that was the problem |
10:40.25 | benjamin7062 | thing is.. it plays the 'prompts' fine |
10:40.28 | benjamin7062 | it's just hold music |
10:40.57 | n3glv | hmm |
10:41.08 | n3glv | try other music files? |
10:41.24 | benjamin7062 | I've tried 3 |
10:41.30 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
10:41.31 | benjamin7062 | they play clear on my system |
10:41.32 | kpettit | good morning |
10:41.37 | benjamin7062 | what is the actual 'player' |
10:41.42 | benjamin7062 | maybe that's my problem |
10:42.23 | n3glv | think it depends on install, but a lot use mpg123 or something |
10:42.44 | *** join/#asterisk michael-i (n=michael@141.41.38.58) |
10:42.49 | benjamin7062 | hmm |
10:42.51 | benjamin7062 | tis isntalled |
10:42.53 | benjamin7062 | installed |
10:42.54 | n3glv | GM kpetit |
10:43.45 | kpettit | anybody know some good places to look to hire asterisk folk? |
10:43.48 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
10:43.52 | kpettit | I need to find some people inm Houston |
10:43.57 | benjamin7062 | www.digium.com |
10:43.59 | backblue | hi, morning all |
10:44.09 | backblue | does anyone it's trying to use hudlite? |
10:44.11 | benjamin7062 | <-- in austin tx |
10:44.19 | kpettit | wanna move :) |
10:44.32 | benjamin7062 | No way, not after Katrina... |
10:44.34 | n3glv | I'll move |
10:44.37 | kpettit | haha |
10:44.39 | benjamin7062 | Houston = Lousianna now |
10:44.42 | n3glv | not expret, but get by |
10:44.42 | benjamin7062 | yuk |
10:44.52 | backblue | hudlite anyone? |
10:45.00 | n3glv | expert that is <ha> |
10:45.06 | kay2 | How can I write something in the CLI from an AGI ? |
10:45.23 | *** join/#asterisk peterm22 (n=petermit@219-90-223-69.ip.adam.com.au) |
10:46.04 | benjamin7062 | kay2, easier using Manager |
10:46.33 | benjamin7062 | or maybe I'm lazy |
10:46.44 | *** join/#asterisk jpeeler (n=thepeel1@host81-149-2-72.in-addr.btopenworld.com) |
10:47.19 | kay2 | benjamin7062 ??? what u talking about |
10:47.40 | kay2 | benjamin7062: I just say can my AGI write something in the CLI |
10:48.04 | benjamin7062 | I know |
10:48.16 | benjamin7062 | was 'teasing' |
10:48.56 | benjamin7062 | It was sarcastic in my .. you can't read my mind? |
10:49.11 | benjamin7062 | [head] |
10:49.15 | benjamin7062 | man, I'm getting tired... |
10:49.30 | *** join/#asterisk duckz (n=duckz@193.192.47.26) |
10:50.49 | benjamin7062 | kay2 -- what language is your AGI in? |
10:51.18 | benjamin7062 | kay2, couldn't you drop to system and run asterisk -rx'CLI Command' |
10:51.19 | benjamin7062 | ? |
10:51.21 | kay2 | no matter |
10:51.32 | kpettit | why not use AMI? |
10:52.00 | benjamin7062 | I got smacked for saying that |
10:52.11 | dlynes_office | ami? |
10:52.19 | dlynes_office | oh...nvm |
10:52.20 | dlynes_office | duh |
10:52.42 | kpettit | benjamin7062, I like ami, works pretty well for me. Better than using asterisk -rx "" anyways |
10:52.44 | benjamin7062 | I assume that means * Manager Interface? |
10:52.50 | kpettit | yes |
10:53.00 | kpettit | it works nice with PHP which is mainly what i program in |
10:53.03 | benjamin7062 | kpettit, I agree... I said that... and she said I was crazy |
10:53.09 | benjamin7062 | That is what _I_ use |
10:53.16 | benjamin7062 | AMI |
10:53.25 | kpettit | I use it to get queue stuff, transfer/place calls, etc. |
10:53.30 | benjamin7062 | exactly |
10:53.41 | benjamin7062 | she apparently is writing something that needs a voice channel I imagine |
10:53.45 | benjamin7062 | thus, she is using the AGI |
10:53.57 | benjamin7062 | so I'm guessing she needs something for a voice prompt or something |
10:54.00 | benjamin7062 | She didn't say |
10:54.25 | benjamin7062 | But an alternative from the AGI would be `asterisk -rx'something'` since she didn't like my AMI suggestion |
10:55.12 | benjamin7062 | <kay2> How can I write something in the CLI from an AGI ? |
10:55.12 | benjamin7062 | * peterm22 (n=petermit@219-90-223-69.ip.adam.com.au) has joined #asterisk |
10:55.12 | benjamin7062 | * FaithX has quit (Connection reset by peer) |
10:55.12 | benjamin7062 | <benjamin7062> kay2, easier using Manager |
10:55.51 | kpettit | got ya |
10:56.23 | kpettit | My big thing i'm trying to program this week is seeing if I can log a agent in thro0ugh a web page |
10:56.33 | benjamin7062 | easy as pie |
10:56.39 | benjamin7062 | if you are using 1.2.9.1 |
10:56.42 | kpettit | I can't find any AMI or asterisk -rx type commands i can use to log a agent in. Logging out was easy enough |
10:56.52 | benjamin7062 | AgentCallBackLogin |
10:56.54 | kpettit | on some machines i am. |
10:57.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
10:57.42 | kpettit | http://www.voip-info.org/wiki-Asterisk+cmd+AgentCallbackLogin |
10:57.52 | benjamin7062 | http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+AgentCallBackLogin |
10:58.10 | peterm22 | hi everyone, quick q. what app can i use for receptionist to "set callerid name" |
10:58.48 | benjamin7062 | Few lines.. especially if you already wrote the functions to connect, login, and communicate which it sounds like you did |
10:58.49 | kpettit | peterm22, real time asterisk with a web interface for the sip section |
10:59.05 | kpettit | benjamin7062, sweet this looks nice |
10:59.23 | benjamin7062 | has to be 1.2.1+ |
10:59.37 | kpettit | benjamin7062, it's amazing how agents can bugger up the login/logout process so I want to have a web page so a manager can log in/out people |
10:59.55 | kpettit | oh yeah I've got that on 90% of my systems. I think everything I'm doing queus with is 1.2.4 or better |
11:00.03 | benjamin7062 | tis 'exactly' what we do from our managers polycom 601 xhtml browser.. =) |
11:00.17 | kpettit | you do that from a polycom??? |
11:00.21 | kpettit | that's freaking cool |
11:00.33 | benjamin7062 | I've written more code in 5 days than ever in my life at one time |
11:00.39 | kpettit | wow. |
11:00.56 | kpettit | how does that work? do you have to have the attachment module? |
11:01.15 | benjamin7062 | no... xhtml browser is on all polycom 601's |
11:01.24 | benjamin7062 | the expansion is kinda useless |
11:01.33 | kpettit | I was happy I just figured out how to get rid of the DND and foward buttons on the polycom. The agents kept doing bad things with those |
11:01.38 | benjamin7062 | most you can do is call 'status' for people.. not functions |
11:02.03 | benjamin7062 | kpettit, you mean in the sip.cfg? |
11:02.03 | kpettit | benjamin7062, I use those like nuts for call status. People seem to be so used to those old style key systems |
11:02.05 | benjamin7062 | features |
11:02.15 | kpettit | benjamin7062, either in there or the individual phones. |
11:02.26 | benjamin7062 | hmmph.. didn't know the phone would let me do that. |
11:02.28 | benjamin7062 | intersting |
11:02.34 | kpettit | I created a sip2.cfg that I referance in the MAC.cfg file for the phones i don't want to have DND and foward buttons |
11:02.44 | kpettit | benjamin7062, its' saved me soooo much heartache |
11:03.11 | kpettit | agents would login then press DND, or forward there phones to some random extension, wich of couse the phone dosen't check |
11:03.28 | benjamin7062 | I've been mildly dissapointed at the button flexibility on almost all the available sip phones... I wish you could make buttons do 'anything' |
11:03.43 | benjamin7062 | the phone assumes it's a new call so it doesn't allow on call functionality |
11:03.47 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
11:03.48 | benjamin7062 | that I have found thus far |
11:03.57 | kpettit | I know I can disable the buttons, which worked great. But that's as much as i've done with them |
11:04.13 | benjamin7062 | I want a 'park' button.. =) |
11:04.25 | kpettit | my pain in the ass right now is parking as well. |
11:04.58 | kpettit | doing a * pbx for a resturant. People that park a call aren't at the same phone for more than a few seconds. so if the person dosen't pick up the call rings back and there never at the phone |
11:05.14 | benjamin7062 | It works... I just wanna' BUTTON for it. But not possible I don't think |
11:05.24 | kpettit | so I'm trying to decided where the parked call ring's back too but i don't seem to ahve that ability. It always rings back to the sip peer that did the park |
11:05.25 | benjamin7062 | kpettit, you can bump the timer for parked calls to 1000000 |
11:05.34 | benjamin7062 | and then have a web interface on the 600 show parked calls. |
11:05.47 | kpettit | I thought aobut that, but in a resturant that's impossible |
11:05.53 | benjamin7062 | true |
11:05.54 | benjamin7062 | good point |
11:06.13 | kpettit | they guys park a call, page "you dude pick up the phone" and if they don't the call gets lost basically becuase there isn't somebody at that same phone to answer anymore |
11:06.37 | benjamin7062 | yup, and if you bump the timeout... they might forget about the call |
11:06.54 | benjamin7062 | have it call back there phone for 30 sec... then forward to park again? |
11:07.07 | kpettit | exactly. If I could control the where it rang back or even if it rang back to a extension rather than a sip peer I'd be ok |
11:07.24 | benjamin7062 | You could capture the ring back in your dialplan |
11:07.28 | benjamin7062 | with some trickery |
11:07.31 | kpettit | hows that |
11:07.32 | benjamin7062 | then do something else with the call |
11:07.52 | benjamin7062 | it will ring back the same phone.. but what you do AFTER that ring back doesn't have to be voicemail |
11:08.14 | kpettit | I created a context that rings all the phones in the store, If I can have a NO-ANSWER r something like that I could have it do that context that rings all the phones |
11:08.31 | kpettit | oh got ya |
11:08.42 | kpettit | ughh I see a painfull alternative. |
11:08.48 | kpettit | it could workthough |
11:09.18 | benjamin7062 | It would ring back one phone.. then if timeout occurs.. move to next step.. check (something)... if something exists... dial (all phones) |
11:09.28 | *** join/#asterisk loopt (n=pt@gw1.sanyo.hu) |
11:09.39 | benjamin7062 | otherwise, hit up voicemail |
11:09.41 | kpettit | when it rings back thogh where do i control that in the extensions.conf |
11:09.58 | kpettit | it's the page app that does the ring back, how can i set a second priority on that? |
11:10.04 | lilalinux | does anyone know a cheap way to connect a mobilephone as a channel? |
11:10.10 | peterm22 | kpettit, i only need to change incoming call callerid name, do i need realtime for that ? |
11:10.15 | benjamin7062 | well, i'd have to play a bit... not sure if the call/channel loses state on a park termination.. but you could set a global var.. and check if it exists |
11:10.38 | benjamin7062 | lilalinux, heh, call a channel? |
11:10.38 | kpettit | peterm22, unless you want ot had edit sip.conf everytime you want to change caller id |
11:10.43 | benjamin7062 | lilalinux, does that count? |
11:10.56 | benjamin7062 | lilalinux, just kidding |
11:11.04 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
11:11.36 | kpettit | benjamin7062, so on the park thing. Where in extensions.conf can i specificy a 2nd priority? It's a features.conf app so i'm not sure how to do that |
11:11.37 | lilalinux | benjamin7062: I wan to use the flatrate of my mobile in asterisk |
11:12.05 | *** join/#asterisk Joe__11 (n=develope@host217-114-154-220.pppoe.mark-itt.net) |
11:13.00 | kpettit | lilalinux, Id see if there was a way you can remotly change where your phone fowards too. If you can control that you can just have * send calls to the cell, and the cell will foward to whereever you set remotly. |
11:13.01 | benjamin7062 | when it comes outta park I 'think' it goes back into the same context as it was in.. Like I said, I'd have to play to check some stuff... But I'm pretty sure it rings back into a 'context'.. whatever it is... in that context, you'd check for a global var or 'something'... and if it exists.. don't go to voicemail |
11:14.14 | kpettit | benjamin7062, yeah I think some tinkering is in order. I know in features.conf I specify the context for "parkedcalls" to use |
11:14.19 | benjamin7062 | lilalinux, to make your mobile a channel for outbound would be hax.. =) you'd have to have drivers for your mobile in linux I'd assume and probably connect via a cable... and hmm, have no idea how you'd get it to talk to * |
11:14.21 | kpettit | just haven't tried putting that in extensions.conf yet |
11:14.42 | benjamin7062 | kpettit, yeah, but that context is different than the ring back I believe... let me test real quick |
11:14.43 | kpettit | or rather I've done the include but not put a [parkedcalls] in there |
11:15.56 | benjamin7062 | parkedcalls is just an include I think |
11:15.59 | kpettit | benjamin7062, I wonder if i could jsut have a [parkedcalls] with a exten => t,1,,,, |
11:16.01 | benjamin7062 | so you can include the parkinglot |
11:16.20 | kpettit | ? |
11:16.21 | benjamin7062 | i 'think' |
11:16.31 | benjamin7062 | one sec |
11:16.36 | n3glv | I could not get ringback to work for me |
11:16.46 | n3glv | am guessing I missed something |
11:18.14 | benjamin7062 | Hmm, I don't know what would happen if you defined stuff in the context of parkedcalls... thought that was reserved but never have tried that |
11:18.29 | n3glv | so, how do I set the level of verbosity on the disconnected cosole screen? |
11:18.30 | benjamin7062 | I'm trying to capture the context of the returned call from park |
11:18.37 | n3glv | the call progress monitor etc |
11:18.46 | benjamin7062 | set verbose 13 |
11:18.49 | benjamin7062 | set debug 13 |
11:18.56 | kpettit | ok cool. I'm experimenting with timeouts in the parkedcall context |
11:19.53 | kpettit | it looks like it's using "park-dial" to do the ring back |
11:20.16 | n3glv | got any ideas why I keep seting the server apear to try login 2 times to my host? |
11:20.35 | n3glv | register I mean |
11:21.24 | *** join/#asterisk op3r (n=op3r@58.69.210.140) |
11:21.24 | *** join/#asterisk tomtom_ (n=tom@83.217.70.166) |
11:21.26 | tomtom_ | hi |
11:21.31 | n3glv | REGISTER attempt 1 to 1412etc@neptune.vtnoc.net |
11:21.44 | n3glv | then agin |
11:21.47 | n3glv | again |
11:23.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
11:23.20 | *** join/#asterisk anna-- (n=nmuller@195.70.21.58) |
11:23.55 | anna-- | hello |
11:24.11 | kpettit | benjamin7062, I found something interesting. It sets a priority1 in parkedcalls. I'm trying to put a priority2 in my own parkedcalls to see if that will take |
11:24.14 | backblue | http://lists.digium.com/pipermail/asterisk-dev/2006-May/020904.html |
11:24.20 | anna-- | I'm using Asterisk with BRI and an HFC card, and I'm trying to set the MSN for out calls... I was able to do it with CAPI by doing "exten => _XXXXXXXXXX,1,Dial(CAPI/ISDN1/05:${EXTEN})" where "05" is my MSN, but I can't find a way to do the same thing with zaptel. Has anyone managed to do that? |
11:25.02 | backblue | anna--: define groups, and use it. |
11:25.08 | backblue | instted of MSN's |
11:25.17 | backblue | and define the msn on the group |
11:25.35 | anna-- | in zapata.conf you mean? |
11:26.25 | backblue | anna--: what driver are you using? |
11:26.34 | anna-- | I'm using the zaphfc module |
11:26.37 | backblue | why use CAPI? |
11:26.46 | backblue | so, you are using bristuff |
11:26.51 | anna-- | I was using CAPI before because I had a AVM card with chan_capi |
11:27.13 | anna-- | but now I'm using a HFC card, so i'm using bristuf and zaphfc |
11:27.21 | backblue | forget capi, forget bristuff, use misdn |
11:27.34 | backblue | anyway, you can do the same, on zapata.conf |
11:27.41 | backblue | if you want to use bristuff |
11:28.01 | anna-- | I've tried it with the AVM card but it kept saying "fatal error in mISDN something..." at boot time and the card was useless |
11:28.02 | backblue | from my experiance, misnd lot better. |
11:28.06 | tomtom_ | anyone any experience with ISDN BRI Gateways, such as Mediatrix, Vegastream or Voxtream Parlay .. any recommendations for use with Asterisk? |
11:28.19 | anna-- | never tried it with the HFC though... |
11:28.21 | backblue | anna--: lspci, and show me the card id. |
11:28.34 | backblue | hfc cards, works great |
11:28.38 | kpettit | benjamin7062, I figured it out!! |
11:28.42 | backblue | what card do you have? single S0? |
11:28.49 | anna-- | 00:0b.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
11:28.57 | backblue | yes, use misnd |
11:28.57 | benjamin7062 | kpettit, using the parkedcalls context? |
11:29.00 | backblue | will work great |
11:29.15 | backblue | i have plenty of machines with cards like that |
11:29.18 | kpettit | in the console you can see it setting a priority 1 extension |
11:29.23 | anna-- | and with mISDN I don't need to patch the asterisk sources? |
11:29.54 | benjamin7062 | yeah |
11:30.16 | kpettit | benjamin7062, in park-dial i set a exten => 701,2,SayUnixTime as a text and it worked |
11:30.32 | benjamin7062 | Ahh... I see |
11:30.36 | kpettit | bad thing is the timeout. It's got to ring for like 60 seconds before it goes to the priority 2 |
11:30.53 | kpettit | I have no idea how I can set the priority 1 ringback timeout. |
11:30.54 | *** join/#asterisk BertZ (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
11:30.57 | BertZ | hi again |
11:31.32 | kpettit | benjamin7062, without hardcoding what asterisk is already doing automatically with parked calls |
11:31.35 | benjamin7062 | change the timeout for the extension it is calling |
11:31.51 | kpettit | it's not calling a extension but the sip peer directly |
11:31.52 | benjamin7062 | see if that affects this? |
11:32.02 | benjamin7062 | oh, duh |
11:32.03 | benjamin7062 | right |
11:32.16 | kpettit | I wish I could make it dial a exten |
11:32.28 | benjamin7062 | I didn't even catch that |
11:32.34 | benjamin7062 | my original thoughts wouldn't be valid |
11:32.44 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
11:33.23 | kpettit | benjamin7062, I was with you on tha torigionall. I actually created a second sip pressensce on each polycom phone. And if anybody call any of the extension fo any of the phones it would do my ringall context. |
11:33.34 | *** join/#asterisk kesa (i=shawarm@203.177.220.220) |
11:33.35 | kpettit | but sense it never rang back the extension my ideal didnt woprk |
11:33.47 | kpettit | this kind of works but the timeout is awfull |
11:34.10 | benjamin7062 | wonder if you can override priority 1? |
11:34.14 | kpettit | I guess it's nce at least to be able to send a 2nd priority at all |
11:34.18 | kpettit | benjamin7062, that would be nice. |
11:34.22 | kpettit | or rather modify it |
11:34.33 | benjamin7062 | what if you specify a priority of 1..would that overrule? |
11:34.38 | kpettit | actually I'm going to try that. I wonder if it'll bugger a parked call |
11:34.44 | kpettit | testing... |
11:34.47 | kesa | Hi. I'm new to Asterisk and would like to know if it has the capability of forwarding a call from one computer to another in a computer network? |
11:35.13 | benjamin7062 | This is a scenario I would have never tested so this is good for me to know! =) |
11:35.31 | Zeeek | kesa you mean with software phones on each computer? |
11:35.38 | Zeeek | In that case, absolutely |
11:35.44 | kpettit | benjamin7062, well it's still parking.. so that's good |
11:35.49 | benjamin7062 | heh |
11:36.52 | kpettit | bugger, it's still doing Timeout for Zap/1-1 parked on 701. Returning to park-dial,SIP/2009,1 on timeout |
11:36.59 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
11:37.03 | benjamin7062 | hmm |
11:37.09 | benjamin7062 | HAS to be a way to configure that |
11:37.15 | kpettit | It must write that on the fly and overwrite what I do |
11:37.41 | kesa | Zeeek: So one computer could act as a switching board and just forward calls to different computers through the use of Asterisk? |
11:37.47 | kpettit | when that priority 1 failes it's still doing my SayUnixTime for priority 2 though |
11:38.23 | Zeeek | kesa that's something it does very, very well assuming you have softphones on each PC with headsets or USB phones |
11:38.25 | benjamin7062 | so the dial back (park-dial) is hard coded as a Dial command |
11:38.26 | benjamin7062 | beh |
11:38.38 | redax | hi |
11:38.59 | benjamin7062 | I'm going to peak at the source |
11:39.10 | redax | is the Queue is suitable to |
11:39.36 | redax | use when too many incoming calls, and no internal to ring? |
11:39.44 | redax | without Answer() of course |
11:39.46 | kpettit | benjamin7062, yes it seems to be. I added my own park-dial context so I could add the second priority |
11:40.04 | kpettit | benjamin7062, so I could try to match what it seems asterisk was automatically doing |
11:40.48 | kpettit | this is what it does when the parked call "times out" by default... |
11:40.58 | kesa | Zeeek: I see. Thanks! :) So that function is built-in already? |
11:41.04 | kpettit | <PROTECTED> |
11:41.04 | kpettit | <PROTECTED> |
11:41.04 | kpettit | <PROTECTED> |
11:41.43 | benjamin7062 | I'm looking for park-dial in the source |
11:41.49 | Zeeek | kesa look here: http://asteriskdocs.org - read the intro of the book there |
11:42.12 | kesa | Zeeek: Alright. Thanks again :) |
11:42.26 | kpettit | is this a valid exten? exten => SIP/2009,1,SayUnixTime ?? |
11:42.38 | Zeeek | also there is no better FIRST introduction to asterisk than this: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
11:42.51 | Zeeek | it's old, but very easy to understand |
11:43.11 | benjamin7062 | Umm, ?.. I honestly can't say... Also have not tried that. I have yet to see that exist |
11:43.21 | benjamin7062 | where you specify the channel for an extension |
11:43.23 | benjamin7062 | but maybe? |
11:43.26 | kpettit | that's what it looks like it's doing. I'm testing... |
11:43.49 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
11:44.39 | benjamin7062 | it actually creates that context on the fly (park-dial) |
11:45.20 | kpettit | benjamin7062, yeah but it listens to what I put in there. it just overwrites what I set as priority1 |
11:45.34 | kpettit | humm setting a exten => Sip/2009 didn't work for me |
11:46.00 | kpettit | <PROTECTED> |
11:46.08 | kpettit | that's the part asterisk doing that I need to overwrite |
11:46.27 | benjamin7062 | you could modify the source in res_features.c to do what you want |
11:46.57 | kpettit | I think I'll have to tell the customer ot go to hell first. haha |
11:47.10 | benjamin7062 | it doesn't look that bad actually |
11:47.16 | kpettit | I've got to many boxes to keep track of to figure out different source code modificatoins |
11:47.52 | dlynes_office | kpettit: use a centralized svn server and create a separate repository for each server |
11:48.13 | dlynes_office | kpettit: have them download new configurations every night |
11:48.26 | dlynes_office | kpettit: and then restart when convenient after the upgrade |
11:48.48 | *** part/#asterisk NLinington (n=nfl@82-69-27-212.dsl.in-addr.zen.co.uk) |
11:48.50 | kpettit | dlynes_office, that's a possibillity. I'm just trying to figure out weather it's better to tell the customer. "Hey this phone system can't do that" or spend a bunch of man horus creating a system to maintain special code for one feature |
11:49.17 | benjamin7062 | looks like you'd change this one line |
11:49.19 | benjamin7062 | ast_add_extension2(con, 1, peername, 1, NULL, NULL, "Dial", strdup(returnexten), FREE, registrar) |
11:49.21 | dlynes_office | kpettit: custom programming always warrants a higher per hour rate, too :) |
11:49.25 | kpettit | dlynes_office, I know there is other junk I'll want to modify, it's just a scary game to get into |
11:49.42 | kpettit | dlynes_office, I wish. We just charge by system not by features. |
11:49.58 | dlynes_office | kpettit: tell them that feature's only included in the deluxe edition |
11:50.08 | dlynes_office | kpettit: and the deluxe edition is a higher price |
11:50.20 | kpettit | dlynes_office, for the extra $$godawfullammout fee |
11:50.27 | dlynes_office | kpettit: exactly |
11:50.32 | *** join/#asterisk userdefined (i=jr000430@shell1.phx.gblx.net) |
11:50.34 | dlynes_office | kpettit: now you catch my drift :) |
11:50.41 | kpettit | benjamin7062, did that specify a timeout? |
11:50.48 | dlynes_office | benjamin7062: ummm....no??? |
11:51.27 | kpettit | benjamin7062, so it looks like it's just ringing back. I bet there has to be a way to set a generic sip ring timeout, or something to that effect |
11:51.44 | Zeeek | kesa I'm there |
11:52.19 | Mw3 | hi. do you know about some analog -> isdn converter. i have 2 analog gsm adapters and a bri card in my asterisk server. i would like to convert the 2 analog gsm to a bri and plug into my bri card |
11:52.19 | benjamin7062 | dlynes_home, You don't think that'd work? |
11:52.36 | dlynes_office | benjamin7062: use ast_strdupa, or you're going to have a memory leak |
11:52.37 | benjamin7062 | dlynes_office, dang, wrong name |
11:53.02 | benjamin7062 | =) I just copied that line from the source.. didn't modify it |
11:53.10 | benjamin7062 | So, they might have a leak |
11:53.16 | dlynes_office | benjamin7062: strdup allocates memory for the pointer and returns a pointer to the malloc'ed memory |
11:53.49 | dlynes_office | benjamin7062: ast_strdupa allocates memory on the stack, not the heap |
11:53.55 | dlynes_office | benjamin7062: and so no need to free it |
11:54.30 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
11:54.32 | benjamin7062 | right now I'm trying to comprehend what they are doing and why |
11:54.51 | benjamin7062 | but I think that method probably has the ability to set a timeout |
11:55.04 | benjamin7062 | assuming it's the "add a call to the queue" method |
11:55.05 | benjamin7062 | not sure |
11:55.06 | dlynes_office | ah wow |
11:55.10 | dlynes_office | whoever wrote that should be shot |
11:55.19 | dlynes_office | or wait |
11:55.23 | kpettit | benjamin7062, I'm going to check in setting a timeout in the sip peer |
11:55.31 | benjamin7062 | heh |
11:55.31 | kpettit | I wonder if that's doable |
11:55.53 | benjamin7062 | kpettit, hmm? suppose that'd be phone specific? |
11:56.00 | kpettit | http://www.voip-info.org/wiki/index.php?page=Functions |
11:56.10 | kpettit | I'm wondering if I can use the TIMEOUT type function specified here |
11:56.30 | kpettit | whoops wrong page |
11:57.07 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
11:57.27 | kpettit | http://www.voip-info.org/wiki/view/Global+Varialbes |
11:57.43 | dlynes_office | ah...nvm |
11:57.44 | kpettit | if not in extensions.conf maybe in sip.conf. |
11:57.52 | dlynes_office | I think they free the memory later on |
11:58.05 | benjamin7062 | kpettit, it might be that one of those NULL values is the timeout. Not sure... if I found another example of call termination I might be able to tell |
11:58.20 | benjamin7062 | I don't have ctags on this machine |
11:58.27 | kpettit | ah |
11:58.27 | dlynes_office | normally though, unless you need the memory to persist past the current function, you should only use ast_strdupa instead of strdup |
11:58.30 | benjamin7062 | so jumping around is .. poop |
11:58.57 | kpettit | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+rtpholdtimeout |
11:58.59 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
11:59.02 | kpettit | what do you tihink of that var? |
11:59.08 | dlynes_office | kpettit: what is it you're trying to do? |
11:59.12 | benjamin7062 | dlynes_office, I think someone should sign their name to each piece of code so we have a target to shoot at |
11:59.28 | dlynes_office | benjamin7062: nah...i think it was probably valid for them to use strdup there |
11:59.30 | *** join/#asterisk alucard064 (n=alucard0@ABayonne-152-1-72-90.w83-200.abo.wanadoo.fr) |
11:59.35 | benjamin7062 | dlynes_home, hook a return call from park and give it an action other than return the call |
11:59.35 | alucard064 | re all |
11:59.43 | kpettit | the overall goal is to define where to send a parked call that time's out and gets sent back. I want to define where I want that returning call to go |
11:59.50 | alucard064 | i try to find a software |
12:00.02 | alucard064 | its name is ipswitchboard |
12:00.02 | dlynes_office | kpettit: ah |
12:00.13 | dlynes_office | alucard064: yeah...it's a .NET client application |
12:00.17 | alucard064 | someone have this software |
12:00.18 | alucard064 | yes |
12:00.27 | alucard064 | its a free client |
12:00.28 | kpettit | dlynes_office, asterisk sends the parked call back to the sip peer that made it. Which in a resturant enviornemt where they aren't in front of the phone for more than 5 seconds at a time a bad thing |
12:00.35 | dlynes_office | alucard064: not quite |
12:00.42 | dlynes_office | alucard064: he might be charging for it in the future |
12:00.49 | dlynes_office | alucard064: but it's free for now, but not opensource |
12:00.54 | alucard064 | yes |
12:00.58 | alucard064 | it s free |
12:01.01 | *** join/#asterisk nothinman (i=shakey@aczw177.neoplus.adsl.tpnet.pl) |
12:01.06 | alucard064 | but i want it to try it |
12:01.08 | kpettit | dlynes_office, I figured out how to set a second priority for the returning parked call if the sip peer dosen't answer the the timeout is looooong |
12:01.12 | nothinman | hry |
12:01.15 | nothinman | hey* |
12:01.19 | alucard064 | but i cna find him |
12:01.30 | alucard064 | i cant find it |
12:01.51 | alucard064 | so if someone have a version of this software |
12:01.55 | alucard064 | please send me |
12:02.04 | *** part/#asterisk Joe__11 (n=develope@host217-114-154-220.pppoe.mark-itt.net) |
12:02.09 | dlynes_office | alucard064: http://ipswitchboard.thorben.dk/ |
12:02.16 | alucard064 | lol |
12:02.23 | alucard064 | yes i know this adress |
12:02.26 | dlynes_office | That's the official homepage for it |
12:02.39 | nothinman | guys, how can I make asterisk to execute Dial() without bridging the call with current call? I'm calling system and after pressing 1 I want asterisk to call the number say something and disconnect. But I can't find any info how to do it... |
12:02.40 | dlynes_office | If you know the address, why haven't you downloaded it? |
12:02.47 | alucard064 | but th software its not avaible and in return it give me the links of easypbx |
12:03.08 | alucard064 | there is no links to download the software |
12:03.32 | alucard064 | so i thought that someone can give me ipswitchboard |
12:03.38 | alucard064 | if it s possible |
12:04.02 | dlynes_office | alucard064: go to easypabx.com, and then click on 'Login' |
12:04.04 | znoG | question: i'm setting up hunt groups in * using just standard dialplan config, and I'm wondering how to deal with loops. eg. extension 1000 is set to forward to 1005 after 20 seconds. but 1005 is also set to forward to 1000 after 20 seconds. What would make sense is that when it hits the first extension (1000) it adds to a variable called "callroute" or something, so when 1005 gets the call, before forwarding to 1000 it checks if it was dialed p |
12:04.13 | dlynes_office | alucard064: It's a link in the upper righthand corner |
12:04.14 | znoG | oops, sorry, typed a bit too much for one message. |
12:04.24 | alucard064 | yok |
12:04.26 | benjamin7062 | kpettit, beh... that method doesn't allow timeouts |
12:04.34 | alucard064 | but when can i download it |
12:04.42 | kpettit | benjamin7062, bummer |
12:04.43 | dlynes_office | alucard064: no idea...i haven't signed up |
12:04.47 | dlynes_office | alucard064: i dont' use windows |
12:04.53 | kpettit | benjamin7062, so without changing the source I'm kinda screwed then it looks like |
12:04.56 | alucard064 | because i want to developp a software as example ipswitchboard |
12:05.18 | benjamin7062 | only 50... you still have a 60 second solution |
12:05.19 | kpettit | is there any good RTA web interfaces out there? |
12:05.23 | kpettit | I hven't checked in awhile |
12:05.27 | benjamin7062 | make the parked timeout like 15 secs. |
12:05.28 | benjamin7062 | heh |
12:05.53 | kpettit | benjamin7062, that just sends it back to the sip peer in 15 seconds, which will ring that sip peer for ever |
12:06.08 | benjamin7062 | not if there's a priority 2, right? |
12:06.08 | kpettit | until it finaly fails over to my 2nd priority which rings all the phones again |
12:06.16 | alucard064 | so |
12:06.26 | benjamin7062 | so 75 seconds... and they are back on the phone |
12:06.30 | alucard064 | if someone have ipswitchboard |
12:06.35 | benjamin7062 | There's another hax you could do |
12:06.40 | kpettit | ? |
12:06.42 | alucard064 | can he give me the software |
12:06.45 | alucard064 | thanks |
12:06.57 | benjamin7062 | But you'd only kinda have 1 call park |
12:07.06 | benjamin7062 | or 1 per button |
12:07.13 | benjamin7062 | this is REALLY hax though |
12:07.15 | kpettit | I have to leave the call parked for a couple of mintues to give them time to page and find the person that should pick up the parked call. Then if they don't... that's what i'm having a hard time with |
12:07.29 | kpettit | go on.... sounding good.,... |
12:07.52 | benjamin7062 | create a few 'park' queues... create a speed dial that does an auto login to the queue... and create a daemon that monitors agent logins that logs them out after the call terminates |
12:07.55 | kpettit | benjamin7062, paging through the ip501's and 601's kicks ass by the way. I sooo love that |
12:08.05 | benjamin7062 | when they hit the 'call park' button.. it basically does a queue login |
12:08.10 | backblue | anyone with hudlite? |
12:08.11 | benjamin7062 | agent login rather |
12:08.18 | benjamin7062 | they get the call.. |
12:08.23 | benjamin7062 | make the queues max len = 1 |
12:08.28 | benjamin7062 | heh |
12:08.32 | benjamin7062 | it's hax though |
12:09.33 | kpettit | ah I got ya. I'm buzzing that to see if that's optional there. |
12:09.36 | kpettit | I love queues though |
12:09.37 | dlynes_office | alucard064: try here: http://ipswitchboard.thorben.dk/index.php?option=com_simpleboard&Itemid=42 |
12:09.49 | benjamin7062 | It would accomplish your park goal |
12:09.52 | benjamin7062 | but it's uglyt |
12:09.53 | benjamin7062 | ugly |
12:10.08 | dlynes_office | benjamin7062: and asterisk source code isn't? :) |
12:10.14 | benjamin7062 | lol |
12:10.16 | benjamin7062 | touche |
12:10.18 | kpettit | man they just need a key system. these old bastards that can't learn new tricks drive me nuts |
12:10.34 | dlynes_office | kpettit: so why do you need to do all this parking crapola then? |
12:10.52 | dlynes_office | Just use autoattendant, if they don't feel like answering the phone, throw it into voicemail |
12:11.07 | dlynes_office | And use hinting on the extensions, so they can see each others' statii |
12:11.19 | kpettit | dlynes_office, it's a resturant. A call comes in, it rings all phones. Call is for Joe, they page over the phones. "Joe you have a call on 701" whatever phone joe happens to be by he cal call 701 |
12:11.39 | dlynes_office | kpettit: ok |
12:11.54 | benjamin7062 | hell, better yet... create an extension that does a 'barge'.. or a call connect.. can't remember the function... but just put the call in a WAIT state... and then 'barge' the call |
12:12.03 | benjamin7062 | I know that's not the right word for it but you get the point |
12:12.04 | kpettit | so that's why the timeout and ringing back to a phone where a person is not at for more than a few seocnds at a time is a problem |
12:12.10 | dlynes_office | and it's just timing out on the park, instead of beeping you to remind you that you have a parked call? |
12:12.29 | benjamin7062 | he wants it to ring back to 'all' sip phones |
12:12.32 | kpettit | benjamin7062, I haven't heard that term before |
12:12.39 | dlynes_office | benjamin7062: ah |
12:12.48 | benjamin7062 | there is a way to force a channel to connect to another channel |
12:12.48 | alucard064 | thanks dlynes_office |
12:13.01 | benjamin7062 | but i'm going to have to dig in my cobwebs |
12:13.05 | kpettit | dlynes_office, if it just rings back to the one sip phone that made the park, chances are that person isn't there anymore |
12:13.06 | benjamin7062 | (the back of my brain) |
12:13.09 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
12:13.23 | kpettit | benjamin7062, is this like remote call pickup? |
12:13.24 | dlynes_office | kpettit: and how does it do that? sorry...i'm not familiar with parking |
12:13.31 | dlynes_office | kpettit: does it ring a Local extension? |
12:13.50 | benjamin7062 | it's hard coded to ring back the sip/<ext> so you can't capture the extension even |
12:14.01 | benjamin7062 | it rings the channel directly |
12:14.04 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
12:14.08 | dlynes_office | ic |
12:14.10 | kpettit | dlynes_office, Basically you get a call, you don't know where the erson is you want to connect to so you park it. Parking you transfer the call to extension 700 (defined in features.conf) |
12:14.26 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
12:14.59 | kpettit | then the system tells you what extension the call is parked on. Defaults to exten 701-705. There is a timeout so if somebody dosen't pickup that parked call in X seconds in rings back to the Sip peer that placed it in park |
12:15.01 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
12:16.55 | kpettit | dlynes_office, benjamin7062 the part that's bugging me is the default timeout for sip. According to the console and the code benjamin7062 found it's not setting a timeout when it rings back the Sip peer |
12:17.44 | kpettit | so if I just do Dial(Sip/2009) without setting a time, how long will that go and can I set a sip.conf global for a timeout before going to second priority? |
12:17.46 | creadurx | i <3 click 2 dial |
12:18.12 | *** join/#asterisk hwt (n=hwt@195.139.204.157) |
12:18.13 | kpettit | I have no idea where it's getting that onw |
12:18.14 | hwt | what does Jun 28 14:17:01 WARNING[5531]: res_musiconhold.c:848 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. mean? |
12:18.27 | hwt | i have compiled zaptel, and it is confirmed loaded. |
12:18.28 | benjamin7062 | Don't hold me to that... there were some variables in there I didn't track down but best I could tell by skimming... I didn't see a timeout |
12:18.30 | hwt | the kernel module. |
12:18.47 | kpettit | hwt, sounds like you don't have a zaptel card or zaptel modules loaded. I'm just guessing though |
12:19.04 | benjamin7062 | that function is defined in pbx.c if you want to peak |
12:19.13 | hwt | kpettit: it is confirmed loaded. |
12:19.15 | *** join/#asterisk nortex (n=nortex@64.136.65.142) |
12:19.18 | kpettit | benjamin7062, the park or the sip timeout? |
12:19.24 | hwt | kpettit: i don't have a zaptel card, though. |
12:19.31 | benjamin7062 | the park-dial timeout |
12:19.32 | kpettit | hwt, you dong zt_dummy? |
12:19.37 | kpettit | ah. |
12:20.20 | dlynes_office | kpettit: rtptimeout=30 |
12:20.34 | dlynes_office | erm 60 i mean |
12:20.36 | tomtom_ | so no one any experience with bri gateways |
12:20.57 | dlynes_office | tomtom_: it sounded like you wanted an isdn->gsm gateway |
12:21.03 | benjamin7062 | I know I saw a way to take a channel and force termination to another channel... or force a transfer |
12:21.15 | dlynes_office | benjamin7062: there is |
12:21.19 | kpettit | hwt, I'm not sure. It just just be yoru mpg123 version or a sound file recorded at the wrong settings |
12:21.42 | hwt | kpettit: maybe, i'll look into it. |
12:22.06 | kpettit | dlynes_office, I just tried that one actually |
12:22.26 | hwt | can i have extensions loaded both from extensions.conf AND realtime? |
12:22.33 | kpettit | hwt, yes |
12:22.34 | dlynes_office | kpettit: »·»·»·»·»·»·snprintf(returnexten, sizeof(returnexten), "%s||t", peername); can be changed so that you can read in an override extension setting from features.conf, and put that in, instead of peername here |
12:22.37 | hwt | or does that only work with sip peers. |
12:22.40 | hwt | kpettit: k, thanks. |
12:22.44 | hwt | cool. |
12:23.23 | dlynes_office | kpettit: so basically check to see if the setting exists, if it does, use that setting; otherwise, use peername |
12:23.37 | *** join/#asterisk Dovid (n=none@barak.cellcom.co.il) |
12:23.40 | kpettit | dlynes_office, I'm alittle lost |
12:23.53 | dlynes_office | kpettit: i guess you're not a c coder? |
12:23.55 | kpettit | what do you mean by peername |
12:24.06 | kpettit | PHP is and basic C is about all I have in me right now |
12:24.43 | Dovid | How do I start a console debug ? |
12:24.55 | dlynes_office | kpettit: say like SIP/2009 |
12:25.02 | dlynes_office | Dovid: set debug 9999 |
12:25.15 | dlynes_office | Dovid: then sip debug |
12:25.21 | benjamin7062 | Dovid, or asterisk -vvvvvvvvvvvvvvvvvvvddddddddddddddddddddr |
12:25.21 | dlynes_office | Dovid: or iax2 debug |
12:25.22 | kpettit | rtpholdtimeout I'm going to try that. A parked call is on hold... Havne' done that one yet |
12:25.29 | dlynes_office | Dovid: or pri intense debug |
12:25.39 | kpettit | dlynes_office, got ya |
12:25.46 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
12:26.13 | benjamin7062 | Driving me crazy.. I can't find the 'thing' that I saw that would force two channels to join/terminate |
12:26.19 | dlynes_office | kpettit: so in your case, you would have a line in your features.conf that looked like 'ringbackext=Local/ringall' |
12:26.30 | dlynes_office | kpettit: and use that instead of peername if it's set to anything |
12:26.52 | kpettit | but I'd have to do the C code to have that right? |
12:26.54 | dlynes_office | kpettit: it's a pretty simple change to res_features.c anyways |
12:27.02 | dlynes_office | kpettit: correct |
12:27.21 | kpettit | ok |
12:27.39 | dlynes_office | and then once you're happy with it |
12:27.46 | tomtom_ | dlynes_office: no, not an isdn->gsm gateway, basically a voip->bri gateway, instead of using bri pci cards |
12:27.49 | dlynes_office | you might want to post it to bugs.digium.com and share it with others |
12:27.52 | benjamin7062 | in your case, you could just hard code the return ext into res_features.c for that particular install |
12:28.09 | dlynes_office | benjamin7062: yeah, but it's better to do it the proper way |
12:28.23 | dlynes_office | benjamin7062: and then kpettit doesn't have to keep patching the code with every new release |
12:28.36 | benjamin7062 | yeah.. but I just realised.. that function doesn't take 'extension'.. it takes 'channel'.. anyway, so I'm wrong either way |
12:29.22 | benjamin7062 | wonder if it'd accept sip/234&sip/234&sip/234 |
12:29.30 | dlynes_office | benjamin7062: of course |
12:29.36 | dlynes_office | benjamin7062: it's only parameters to the dial command |
12:29.47 | dlynes_office | benjamin7062: but to keep it simple |
12:29.59 | dlynes_office | benjamin7062: i'd still do ringbackext=Local/ringall |
12:30.16 | benjamin7062 | dlynes_office, I didn't follow it through what it was doing |
12:30.25 | benjamin7062 | just poked and skimmed |
12:30.29 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
12:30.30 | kpettit | Id' be nice if I could jus tdo exten => sip/2009,1,,,, type of thing |
12:30.39 | benjamin7062 | lol |
12:30.43 | dlynes_office | benjamin7062: and then in [default], do exten => ringall,1,Dial(SIP/2009&SIP/2010&SIP/2011) |
12:30.44 | kpettit | so if it called sip/2009 but I think that's not going to happen |
12:30.47 | dlynes_office | erm |
12:30.51 | kpettit | exactly |
12:30.52 | benjamin7062 | you can if you're willing to accept 60sec timeout |
12:30.57 | kpettit | hah |
12:30.57 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
12:31.00 | *** join/#asterisk graphyx (n=mkling@24.144.57.161) |
12:31.00 | dlynes_office | benjamin7062: and then in [default], do exten => ringall,1,Dial(SIP/2009&SIP/2010&SIP/2011,,t) |
12:31.02 | kpettit | yeah that's what I've got now |
12:31.09 | graphyx | anyone familiar with sivus? |
12:31.36 | dlynes_office | kpettit: what is? |
12:31.50 | graphyx | The Sip Protocol security scanner. |
12:32.16 | kpettit | i've got a second priority that goes after 60 seoncds that does a ringall to all the phones |
12:32.46 | kpettit | the first priority is dynamic from * so I can't control that, or the sip timeout when it rings so that's what I'm stuck with right now |
12:32.59 | kpettit | unless I do some C hacking that is :) |
12:33.49 | benjamin7062 | it's application transfer it think... |
12:34.06 | benjamin7062 | if the call were sitting in a queue... you could have a button that 'transferred the call the that phone' upon pressing it |
12:34.09 | dlynes_office | kpettit: well, what i suggested is probably exactly what you want |
12:34.10 | benjamin7062 | there you go |
12:34.13 | benjamin7062 | =) |
12:34.18 | dlynes_office | kpettit: and then you can change it easily, too |
12:34.23 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
12:34.24 | benjamin7062 | that completes my UGLY ass way to accomplish this |
12:34.36 | dlynes_office | kpettit: you don't have to modify the code every time you want to change the extensions it rings |
12:34.43 | dlynes_office | kpettit: or the dial pattern, or whatever |
12:35.25 | benjamin7062 | kpettit, dlynes_office's solution is far more accurate and elegant than mine... =) |
12:35.25 | kpettit | dlynes_office, I understandthat part. I just don't think changing code is going to be a good option |
12:35.56 | dlynes_office | kpettit: well, to each his own, i suppose |
12:36.14 | kpettit | dlynes_office, I wish I could. I just really short on people right now |
12:36.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
12:36.33 | dlynes_office | kpettit: yeah...but something like that would take about 1/2 hour of C hacking, if that |
12:36.40 | kpettit | and we don't have antying in place now to track source changes. so it would be a new thing and we've got over 40 PBX's out there |
12:36.46 | *** join/#asterisk Lord_Drachenblut (n=Lord@74.129.228.28) |
12:36.51 | Lord_Drachenblut | hello |
12:37.00 | dlynes_office | yeah...that's something you'd change on your test machine only, until you're happy with it |
12:37.06 | dlynes_office | then when you're happy with it |
12:37.07 | Lord_Drachenblut | have a question to ask if you guys have the time |
12:37.14 | dlynes_office | repackage your asterisk and redeploy |
12:37.21 | dlynes_office | ~suggestions |
12:37.34 | jbot | from memory, suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite ... |
12:37.34 | kpettit | dlynes_office, I apprecaite it. |
12:37.47 | kpettit | dlynes_office, I need to find another body to help me out with this stuff. |
12:38.03 | Lord_Drachenblut | i am trying from cli to use the playback command to play an audio file but i keep getting -bash: syntax error near unexpected token `(' |
12:38.10 | dlynes_office | kpettit: heh...i'd help, but i'm already overloaded with work as it is |
12:38.29 | kpettit | dlynes_office, I sooo know how that goes. I'm doing 60hr weeks ever week. |
12:38.34 | *** join/#asterisk xxttxxttxxtt (n=chatzill@rrcs-67-52-187-18.west.biz.rr.com) |
12:38.38 | dlynes_office | Lord_Drachenblut: ummm...why are you trying to use playback from the cli? |
12:39.10 | benjamin7062 | <-- just worked another 40 hr shift |
12:39.14 | benjamin7062 | and I have to stay all day |
12:39.16 | benjamin7062 | and stay late |
12:39.16 | benjamin7062 | sigh |
12:39.17 | Lord_Drachenblut | dlynes_office, i am working on a graphical frontend that when you hit a button will call a person and playback an audio file |
12:39.20 | dlynes_office | benjamin7062: ring a ding |
12:39.28 | dlynes_office | benjamin7062: i've been up since 8am yesterday morning |
12:39.36 | benjamin7062 | Me too! =) |
12:39.40 | dlynes_office | benjamin7062: that's how it goes in the it industry |
12:39.47 | benjamin7062 | I wondered why you were always in chat while I am! |
12:39.53 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
12:40.01 | benjamin7062 | thought maybe you were in Australia or something |
12:40.02 | dlynes_office | It's 5:40am here |
12:40.15 | graphyx | so nobody has tried out the sivus scanner at all? |
12:40.16 | benjamin7062 | 8:00 here |
12:40.26 | dlynes_office | I'm busy building a bunch of packages to deploy for troubleshooting some buggy aastra firmware |
12:40.31 | benjamin7062 | well.. 7:40 |
12:40.32 | mut | thats only how it goes when ya don't know what you're doing |
12:40.43 | dlynes_office | stupid companies that make windows-only voip debugging solutions |
12:40.51 | dlynes_office | like wtf? |
12:40.58 | benjamin7062 | vmware! |
12:40.59 | kpettit | dlynes_office, benjamin7062 must just be this line of work |
12:41.02 | benjamin7062 | that's how I run windows |
12:41.07 | dlynes_office | how the hell are you supposed to run Windows apps remotely? |
12:41.13 | benjamin7062 | kpettit, and doctors! |
12:41.14 | creadurx | what better way to run windows than in a window |
12:41.31 | dlynes_office | especially when you don't have access to do port mapping? |
12:41.52 | benjamin7062 | dlynes_home, oh, i see.. the windows app is on someone 'elses' machine |
12:41.58 | dlynes_office | no |
12:42.00 | kpettit | oh god I had the works sip bug that was driving me freaking nuts for weeks |
12:42.02 | benjamin7062 | VNC? |
12:42.14 | dlynes_office | I have to run a .net app to access an aastra phone that's sitting behind a firewall at a remote site |
12:42.23 | kpettit | our T-1 provider is our SIP provider whcih is nice (fax works great over sip), but DTMF wasn't working with somepeople |
12:42.24 | benjamin7062 | lol |
12:42.26 | benjamin7062 | hawt |
12:42.47 | dlynes_office | and obviously I'm not going to be able to run it on the client's machine because they're going to want to use their machine, and they won't trust me to be on their machine when they're not there |
12:42.56 | kpettit | dtmfmod=auto and every other option, and we did all other sorts of debuging, but the fix ended up being setting canreinvite=yes |
12:43.16 | benjamin7062 | I've been running unix on my desktop so long I'm outta the loop... that's why I threw vmware out there... just assumed everyone was like me... which is like, 0 people |
12:43.20 | kpettit | that was killing me. I guess the telco didn't like asterisk sending the dtmf so doing canreinvite=yes fixed it |
12:43.32 | dlynes_office | kpettit: canreinvite=yes makes asterisk get its ass out of the media path |
12:43.51 | kpettit | benjamin7062, I'm with yha man. The closest thing to a windows box I have in work or house is SuSE |
12:43.55 | dlynes_office | benjamin7062: ummm |
12:44.02 | *** join/#asterisk Dovid_Laptop (n=none@barak.cellcom.co.il) |
12:44.09 | Lord_Drachenblut | dlynes_office, no idea's? |
12:44.10 | dlynes_office | benjamin7062: i haven't used a windows desktop on any of my machines for as long as i can remember |
12:44.13 | dlynes_office | Lord_Drachenblut: ? |
12:44.22 | kpettit | dlynes_office, exactly, which I guess made all the difference in the world for DTMF. which is odd becuase it only effected about 5% of the calls |
12:44.29 | benjamin7062 | I feel so... validated! |
12:44.30 | dlynes_office | Lord_Drachenblut: oh....why not use fop? |
12:44.56 | dlynes_office | Lord_Drachenblut: I think fop will allow you to do all that |
12:45.02 | dlynes_office | Lord_Drachenblut: no point reinventing the wheel |
12:45.20 | kpettit | benjamin7062, I feel guilty when I use SuSE, instead of a source distro. Like I took the easy way out or something. haha |
12:45.21 | benjamin7062 | now that multimonitor support works well in X I can't go back. I can push WAY more apps on these 6 screens vs windows... which pukes at just processing that much video |
12:45.21 | *** join/#asterisk coppice (n=chatzill@223.193.17.210.dyn.pacific.net.hk) |
12:45.24 | Lord_Drachenblut | dlynes_office, there is a point but thanks for the help |
12:45.29 | benjamin7062 | and windows has decent multi mon support |
12:45.33 | *** part/#asterisk graphyx (n=mkling@24.144.57.161) |
12:45.46 | benjamin7062 | don't get me wrong |
12:46.05 | benjamin7062 | but just starts to die after a while.. no matter how much ram you throw at it |
12:46.10 | kpettit | benjamin7062, I like the vnc hack so I can use the same monitor keyboard and run it from one side of my screen to multiple other computers |
12:46.13 | dlynes_office | kpettit: what source distros is there besides Gentoo, Sourcemage, and FreeBSD ports collection? |
12:46.27 | benjamin7062 | x11vnc you mean? |
12:46.39 | kpettit | dlynes_office, I'm not sure, it seems to change daily. Gentoo is what i use for work servers |
12:47.00 | kpettit | benjamin7062, I thnk that's it. it's just the keyboard and mouse that is shared |
12:47.02 | dlynes_office | kpettit: ok...was just making sure you weren't including slackware in that ugly list :p |
12:47.24 | kpettit | i haven't used that one in forever |
12:47.24 | benjamin7062 | <-- still a debian vagina |
12:47.28 | kpettit | haha |
12:47.39 | kpettit | I like doing the Knoppix debian thing |
12:47.50 | dlynes_office | yeah...Knoppix is cool |
12:47.51 | kpettit | I switch dekstop distro's every few months it seems. |
12:47.56 | dlynes_office | it's my favorite rescue cd distro |
12:48.05 | kpettit | the knoppix install is pretty sweet |
12:48.09 | benjamin7062 | I'm so lazy... I compile production services always... but for tools.. I hate compiling.. (like tcpdump, less, etc) |
12:48.15 | dlynes_office | kpettit: i've never installed it |
12:48.15 | *** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org) |
12:48.20 | kpettit | then you have a pre-installed working system that you can apt-get install... |
12:48.20 | dlynes_office | kpettit: only ran it on the live cd |
12:48.34 | kpettit | I installed it and used it for a few months when the first DVD version came out |
12:48.34 | tzafrir | kpettit, if you refer to something you can use on a server, count knoppix out. You can use it, but not maintain it |
12:48.39 | benjamin7062 | You guys know about systemrescuecd.iso right? |
12:48.45 | kpettit | tzafrir, I'm just talking desktop |
12:49.02 | kpettit | benjamin7062, Yeah I like that partionmagic clone on there. that's pretty sweet |
12:49.06 | dlynes_office | benjamin7062: no idea, but my main use for a rescue cd is for backing up windows filesystems that got hosed |
12:49.30 | dlynes_office | benjamin7062: because somebody clicked "OK! Clean the 56 viruses from my system!" |
12:49.37 | benjamin7062 | hahahahha |
12:49.43 | tzafrir | it's a demo. Not a desktop. The problem is that your only upgrade option is debian Sid. If you're a newbe you'll end up not upgrading at all |
12:49.45 | benjamin7062 | I laugh.. because you speak truth |
12:49.55 | dlynes_office | benjamin7062: i know i speak the truth |
12:50.12 | dlynes_office | benjamin7062: that's a popup window from internet explorer asking you to click ok to install the latest spyware |
12:50.12 | kpettit | I did gentoo KDE desktops for all my sales guys |
12:50.35 | kpettit | did the kororaa KDE install thing which worked out nice. |
12:50.35 | Dovid_Laptop | Besides for these lines what do I have to put in to meetme.conf ? |
12:50.35 | Dovid_Laptop | http://astcc.dovid.net/meet.txt |
12:50.54 | benjamin7062 | I run etch on servers... instead of sid |
12:51.12 | kpettit | etch? |
12:51.15 | benjamin7062 | OMG -- when they broke pam on unstable a while back... I wanted to spit |
12:51.18 | benjamin7062 | etch = testing |
12:51.21 | benjamin7062 | it's in the middle |
12:51.22 | op3r | does anyone know agentcallbacklogin here? |
12:51.29 | benjamin7062 | stable is always 'OLD' |
12:51.29 | dlynes_office | Dovid_Laptop: none of those belong in meetme.conf; those are all for extensions.conf |
12:51.38 | kpettit | op3r, I wish, that's on my todo list |
12:51.41 | Dovid_Laptop | I know |
12:51.46 | Dovid_Laptop | Its in exten.conf |
12:51.51 | benjamin7062 | op3r, yes, what do you want to know? |
12:51.52 | kpettit | op3r, http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+AgentCallBackLogin |
12:51.53 | tzafrir | Dovid_Laptop, do you have an existing meetme root for every number that will get there? |
12:51.55 | Dovid_Laptop | My question is what do I add to meetme.conf ? |
12:51.56 | dlynes_office | Dovid_Laptop: no such file in asterisk |
12:52.12 | Dovid_Laptop | tzafrir: can I pm ? |
12:52.23 | tzafrir | yes |
12:52.56 | kpettit | benjamin7062, haha i know what you mean. Some of the new 1.2.9 bugs are kind of funny |
12:53.10 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
12:53.19 | kpettit | benjamin7062, asterisk -rx "sip show peers" or basically any command just shows the first line |
12:53.24 | dlynes_office | littleball: you had a problem earlier....i forget what it was |
12:53.33 | dlynes_office | littleball: but you logged off before i had a chance to reply |
12:53.34 | kpettit | maybe that's fixed now, but on all the 1.2.9's I have it has the problems. kind of anoying |
12:53.34 | littleball | hi dlynes_office |
12:53.39 | op3r | benjamin7062: because I am having this problem that the agent can login twice |
12:53.41 | hwt | is realtime stable enough for production use now? roughly 1000 users. |
12:53.56 | op3r | benjamin7062: even if he is already logged in on another extensions |
12:54.08 | littleball | hi dlynes_office, i forgot the problem, maybe it is related to "hint" prioroty |
12:54.10 | kpettit | op3r, if they sit at the same phone I'd hardcode the sip peer to be logged into the queue. |
12:54.19 | kpettit | op3r, agents have a way of buggering uip the login |
12:54.20 | dlynes_office | littleball: oh yeah...that's what it was |
12:54.30 | dlynes_office | littleball: what problem exactly were you having? |
12:54.32 | littleball | can explain to me? |
12:54.36 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
12:54.36 | *** mode/#asterisk [+o anthm] by ChanServ |
12:54.38 | benjamin7062 | kpettit, hrmm? I get all lines? weird? |
12:54.45 | op3r | kpettit: so the best way is to hard code the phones to queues? |
12:54.47 | dlynes_office | good morning, anthony |
12:55.00 | kpettit | benjamin7062, which release do you have? |
12:55.05 | op3r | kpettit: because need to be able to monitor the agent login logout using qmetrics |
12:55.12 | trelane_ | what are the general thoughts on asterisk and preempt? does the lower latency from preempt help? |
12:55.13 | benjamin7062 | kpettit, 1.2.9.1 |
12:55.16 | littleball | i just don't understand when i read http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
12:55.18 | anthm | hi |
12:55.20 | kpettit | op3r, in queues.conf you specify |
12:55.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
12:55.30 | dlynes_office | littleball: what don't you understand? |
12:55.34 | kpettit | op3r, pm me and i'll paste you a example |
12:55.43 | nfi|ermes | where can i read about spacial characters, like "_" "." ecc |
12:55.47 | nfi|ermes | ? |
12:56.02 | benjamin7062 | kpettit, do agents only have one station? |
12:56.11 | littleball | dlynes_office, start from the begining what is the purpose of "hint"? |
12:56.19 | kpettit | op3r, basically in queues.conf just define member => Sip/206 |
12:56.21 | nortex | Strange behavior, If I set SipHeader(Alert_Info : XXX) before sending the call to a queue of 3 Polycom phones the phones autoanswer. But the alert_Info class is only for a specific ringtone in the sip.cfg on the phones. Any ideas why? |
12:56.23 | benjamin7062 | sorry.. op3r do agents only have one station |
12:56.27 | dlynes_office | littleball: blf (busy lamp field) |
12:56.40 | dlynes_office | littleball: well, presence detection |
12:56.47 | op3r | benjamin7062: the reason why i need them to be able to use agentcallbacklogin is that they are all outbound agents (meaning they are making calls not taking calls) |
12:56.49 | dlynes_office | littleball: in sip's case, it's called BLF (busy lamp field) |
12:56.58 | dlynes_office | erm presence awareness |
12:56.59 | littleball | exten => 200,hint,SIP/phone1, in this case, how to understand this |
12:57.08 | benjamin7062 | op3r, right.. but do they have ONE desk... |
12:57.14 | benjamin7062 | like, is a person assigned to a desk? |
12:57.16 | *** join/#asterisk Lord_Drachenblut (n=Lord@74.129.228.28) |
12:57.24 | littleball | dlynes_office, in jabber, it is called presence, in SIP, it is called BLF? |
12:57.31 | op3r | benjamin7062: nope its first come first serve |
12:58.02 | dlynes_office | littleball: that means that when SIP/phone1 is busy, anyone that's checking for hint extension 200 will turn their busy lamp field on for that extension |
12:58.12 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
12:58.14 | trelane_ | With regard to asterisk performance on a system I'm having hardware problems with (the digium card is being mean to me! :( ) what are some thoughts on using preempt as I recompile the kernel |
12:58.33 | dlynes_office | littleball: it also means if someone hits the button associated with that busy lamp field, it'll dial SIP/phone1 |
12:58.47 | littleball | dlynes_office, ok. understood this. |
12:59.23 | littleball | dlynes_office, next question, subscribecontext= |
12:59.26 | benjamin7062 | op3r, perhaps you should have them auto log them out before logging them in? |
12:59.32 | littleball | why subscribecontext= is needed? |
12:59.38 | benjamin7062 | even if they aren't logged in |
12:59.43 | *** part/#asterisk oscarh (n=oscar@host-87-74-0-243.bulldogdsl.com) |
13:00.07 | dlynes_office | littleball: three phones I know of off the top of my head that support it are Aastra 9133i/Aastra 480i/Aastra 480iCT(BLF), Grandstream GXP2000(BLF), Polycom 501/601(buddy list) |
13:00.08 | *** join/#asterisk McLazarus (n=mcallist@pool-72-78-138-105.phlapa.east.verizon.net) |
13:00.08 | benjamin7062 | op3r, run the logout prior to login... for your login extension |
13:00.21 | op3r | hmm |
13:00.24 | benjamin7062 | op3r, not sure if that would work... |
13:00.26 | *** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
13:00.27 | benjamin7062 | but maybe |
13:00.33 | dlynes_office | littleball: subscribecontext is needed so that asterisk knows which dialplan context to look in for hint subscriptions |
13:00.50 | benjamin7062 | I'll try real quick |
13:00.50 | op3r | benjamin7062: can i pm you my agentcallbacklogin info from my extensions.conf? |
13:00.50 | hwt | are there any good web-based management interfaces for realtime? |
13:00.59 | op3r | hwt: fop? |
13:01.04 | benjamin7062 | one sec |
13:01.09 | benjamin7062 | let me check if this even works |
13:01.15 | dlynes_office | op3r: fop isn't a realtime web based management interface |
13:01.17 | hwt | i don't want to use A@home or AMP. |
13:01.31 | dlynes_office | hwt: fop doesn't require A@home or amp |
13:01.38 | hwt | op3r: no, to edit dialplans, extensions, sip users, etc. |
13:01.39 | dlynes_office | hwt: but it's not a realtime management interface, either |
13:01.47 | hwt | dlynes_office: yup, i know. |
13:01.49 | littleball | dlynes_office, let me think first. not understand this yet |
13:02.05 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
13:02.28 | kpettit | op3r, check out my agent logoff, it's the ugliest logoff hack ever. |
13:02.32 | kpettit | <PROTECTED> |
13:02.44 | nortex | Strange behavior, If I set SipHeader(Alert_Info : XXX) before sending the call to a queue of 3 Polycom phones the phones autoanswer. But the alert_Info class is only for a specific ringtone in the sip.cfg on the phones. Any ideas why? |
13:02.51 | *** join/#asterisk goof (n=goof@81.199.100.163) |
13:03.14 | littleball | dlynes_office, can u describe how a sip phone can subscribe another sip phone's presence |
13:03.15 | op3r | hwt: check out the php edit config that was used by trixbox |
13:03.17 | littleball | ? |
13:04.10 | littleball | dlynes_office, the whole flow is important for understanding the whole sip presence thing. :-) |
13:05.32 | benjamin7062 | kpettit, actually -- that's what I do too... since there is no way to auto log someone off |
13:05.34 | dlynes_office | littleball: one sec...just bringing up a page |
13:05.40 | littleball | thanks |
13:06.34 | xxttxxttxxtt | Hey dlynes_office , Do you know where I could insert the IVR into my dial plan ? |
13:07.01 | dlynes_office | littleball: try this link...it's a wiki on there that I've started but haven't had a chance to finish yet |
13:07.04 | dlynes_office | littleball: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+subscribecontext |
13:07.22 | dlynes_office | littleball: it involves a few screenshots of Aastra 9133i's for setting up SIP presence (BLF's) |
13:07.30 | dlynes_office | xxttxxttxxtt: wherever you want |
13:07.51 | littleball | thanks |
13:07.58 | hwt | op3r: so trixbox is the new name of AMP? |
13:07.58 | xxttxxttxxtt | do i need ne extra commands |
13:08.08 | kpettit | benjamin7062, that's funny, I thought that was soo ugly |
13:08.11 | dlynes_office | hwt: new name of Asterisk@Home |
13:08.17 | hwt | dlynes_office: ah, right. |
13:08.19 | op3r | hwt: no its the new name of A@H |
13:08.22 | dlynes_office | hwt: FreePBX is the new name of AMP |
13:08.29 | hwt | it's probably easier to just write something yourself. |
13:08.31 | benjamin7062 | kpettit, really, it's the only way to auto do it.. or create an AGI that calls the AMI |
13:09.07 | kpettit | it's just funny you can't code that directly I think |
13:09.32 | *** join/#asterisk }btorch{ (n=btorch@208.63.19.179) |
13:09.48 | benjamin7062 | op3r, it works... do exactly what kpettit did above but changing to your 'login' method... have that run prior to a login... if they are logged in, it will log them off,... then ask for a password to login.. if they aren't logged in.. it will still log them in... now they can't double queue. |
13:09.55 | *** join/#asterisk bkw__ (n=brian@asterisk/friend-and-developer/bkw) |
13:10.28 | benjamin7062 | kpettit, the person who coded it had a one track mind.. only invisioned people would want to do the login/logoff via phone... not dialplan |
13:11.03 | benjamin7062 | kpettit, same thing; you can't log someone in automatically without prompting for a password |
13:11.38 | benjamin7062 | I understand you may want security but it should be 'my' decision .. not set in stone... but there are always hax around the system |
13:11.40 | benjamin7062 | =) |
13:11.41 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
13:11.54 | benjamin7062 | like a custom perl app that calls the AMI |
13:11.58 | *** join/#asterisk Katty (n=aisaacs@64.82.232.54) |
13:12.31 | Katty | hey dlynes (= |
13:12.38 | op3r | benjamin7062: nice |
13:12.40 | *** join/#asterisk speedwagon (n=Ariel@70.46.87.158) |
13:12.46 | benjamin7062 | op3r, work for you? |
13:13.16 | op3r | benjamin7062: trying it on a dev server |
13:14.19 | }btorch{ | hi , is there a way to change the voicemail app so that it doens't play such a long intro before each ne msg ? |
13:15.06 | }btorch{ | all my users complain that it takes so long to hear the actual msg |
13:15.24 | kay2 | if I have two channels in my AMI, of let say two people that are in MusicOnHold(), how can I from the AMI make one talk to the otherone ? |
13:15.50 | SheriF_WorK | i want to understand something about buying the G729 codecs what dose it mean per channel ? |
13:16.33 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcd.mn.charter.com) |
13:16.40 | dlynes_office | alucard064: you still there? |
13:16.43 | }btorch{ | any way to do that ? |
13:18.24 | benjamin7062 | kay2, perhaps transfer them both to a meetme? |
13:19.48 | dlynes_office | }btorch{: maxgreet, saycid=no, sayduration=no, saydurationm=5, envelope=no |
13:20.05 | dlynes_office | }btorch{: check your sample voicemail.conf file for more info on those five options |
13:20.13 | *** join/#asterisk ambriento (n=1@200.190.197.66) |
13:22.12 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
13:23.02 | *** join/#asterisk boddy (n=e@212.58.24.138) |
13:23.40 | Katty | ariel_: (= |
13:24.05 | X-Gen | is there a chan skype yet ? |
13:24.09 | X-Gen | Skype-to-Asterisk |
13:24.17 | boddy | hii all I configure extensions conf --> _1XXX,1,Dial,Zap/1/${EXTEN} |
13:24.27 | boddy | _5XXX,1,Dial,Zap/1/${EXTEN} |
13:24.49 | znoG | question: i'm setting up hunt groups in * using just standard dialplan config, and I'm wondering how to deal with loops. eg. extension 1000 is set to forward to 1005 after 20 seconds. but 1005 is also set to forward to 1000 after 20 seconds. What would make sense is that when it hits the first extension (1000) it adds to a variable called "callroute" or something, so when 1005 gets the call, before forwarding to 1000 it checks if it was dialed p |
13:24.55 | boddy | I can call telephones that begin with 1 |
13:25.02 | boddy | but I cant 5 |
13:25.29 | Katty | are they 4 digits long, starting with 5? |
13:25.43 | ariel_ | Katty, morning |
13:25.49 | Katty | ariel_: how's it goin, hun? |
13:26.01 | }btorch{ | dlynes_office, thanks |
13:26.08 | ariel_ | ruff been very busy... but not enough income.. normal |
13:26.11 | Katty | hey iDunno (= |
13:26.12 | boddy | Katty you sayin me ? |
13:26.19 | Katty | ariel_: aww, i'm sorry to hear that )= |
13:26.38 | Katty | ariel_: stuffs a little tight around here too...windshield got cracked beyond repair. |
13:26.49 | Katty | ariel_: and my license renewal is up soon, so it had to be replaced right away ;) |
13:27.11 | Katty | boddy: yes. |
13:27.22 | *** join/#asterisk a20060628 (n=fazakasc@86.35.34.63) |
13:27.33 | *** join/#asterisk Persilon (n=ajolodov@200.123.112.152) |
13:27.38 | Persilon | Hi |
13:27.42 | benjamin7062 | boddy, do you have anything else in your dial plan that could possibly be catching _5XXX first? |
13:27.44 | boddy | I just want redirecet telephon numbers start with 5 to zap |
13:27.46 | ariel_ | Katty, well in my state windshields are required to be changed and paid for by the insurance co. without any deductable to the driver. |
13:28.11 | a20060628 | how can i find the dial command result? |
13:28.27 | Katty | ariel_: wow, that's pretty nice. |
13:28.35 | Katty | ariel_: i have glass coverage, but my deductable is 500. |
13:28.42 | kpettit | I'm tryijng to parse through a cdr record to see how many voicemail's were left for a extension |
13:28.43 | Persilon | I'm having troubles with Playtones(), I'm building an ivr and it only works whitin another extention, not on the ivr extention |
13:28.46 | ariel_ | You should check with your state most of them have the same rule. |
13:28.53 | Katty | ariel_: i already did check. |
13:28.55 | littleball | anyone using firefly? i can call out from firefly, but cannot call to firefly. everything works for xlite |
13:28.58 | kpettit | any easy way to do that. Voicemail's have all been cleared out now, but I want to make sure that they were being left correctly. |
13:29.00 | ariel_ | argh |
13:29.02 | Katty | ariel_: i'd have to pay 500 to get a 200 job done ;) |
13:29.09 | Katty | ariel_: and it's already taken care of now anyway. |
13:29.13 | kpettit | Grepping for the voicemail extension in Master.csv dosen't seem to do the trick |
13:29.14 | ariel_ | I see |
13:29.26 | boddy | ? |
13:29.32 | nortex | Strange behavior, If I set SipHeader(Alert_Info : XXX) before sending the call to a queue of 3 Polycom phones the phones autoanswer. But the alert_Info class is only for a specific ringtone in the sip.cfg on the phones. Any ideas why? |
13:29.40 | benjamin7062 | kpettit, http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+MailboxCount |
13:30.08 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
13:30.13 | benjamin7062 | kpettit, as an alternative |
13:30.52 | kpettit | This is more of a immediate thing. They already clearned out the INBOX but I need to show that messages were being send to the inbox. |
13:30.57 | kpettit | not sure the best way to go about that. |
13:31.19 | kpettit | I know the number people were calling and the voicemail box. but i'm not sure how to tell from Master.csv if it actually left a voicemail. |
13:31.20 | Katty | ariel_: you any good with mounting win2k3 shares? (smbfs) |
13:31.50 | creadurx | is there anyway i can tell my ip10s SIP phone to answer a channel that is ringing? via the manager interface.. |
13:31.51 | ariel_ | you mean setting up samba on a linux box? |
13:31.58 | Katty | ariel_: nono. |
13:32.05 | Katty | ariel_: it technically /mounts/ |
13:32.10 | Katty | ariel_: but gives me errors about stale nfs |
13:32.29 | a20060628 | can u help me, pls ? |
13:32.31 | ariel_ | I don't deal much with that... |
13:32.33 | *** part/#asterisk jpeeler (n=thepeel1@host81-149-2-72.in-addr.btopenworld.com) |
13:32.37 | Katty | ariel_: m'kay. |
13:32.45 | Katty | ariel_: i'll find someone else to pester, don't worry ;) |
13:32.52 | nortex | Katty, Is it still not working? |
13:32.56 | a20060628 | how can i find the asterisk dial command result |
13:32.59 | Katty | nortex: no, no it's not. |
13:33.00 | a20060628 | ??? |
13:33.11 | nortex | Katty, but it did mount |
13:33.16 | Katty | nortex: still at the same point i was yesterday. mounts but gives me a stale nfs error on dir |
13:33.21 | Katty | nortex: it /has/ mounted. |
13:33.22 | ariel_ | a20060628, more info is needed. I don't understand |
13:33.27 | Katty | nortex: always has. |
13:33.31 | Katty | nortex: but just not properly. |
13:33.38 | nothinman | guys, how can I make asterisk to execute Dial() without bridging the call with current call? I'm calling system on Zap/1-1 and after pressing 1 I want asterisk to call the number using Zap/2-1, say something and disconnect. But it's hanging up if I hang up or bridging the call when I don't. Any ideas? :/ |
13:33.58 | Katty | anthm, dear, are you around? |
13:33.59 | benjamin7062 | kpettit, so, you can see that the voicemail answered.. but you can't see if they actually left a message? |
13:34.08 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
13:34.11 | anthm | yes hon |
13:34.21 | a20060628 | in php i use the command exec("dial","") ..., and i want to know what is the result of this |
13:34.38 | kpettit | benjamin7062, I can see calls in the cdr but I don't know how to verify if a voicemail was left |
13:34.41 | a20060628 | somebody answered, busy somthing like that |
13:35.08 | Katty | anthm: ever heard of a "stale nfs file handle" when you mount a win2k3 share? |
13:35.22 | anthm | yah |
13:35.36 | ariel_ | a20060628, there is very good dialparties.agi out there on the wiki that has allot of info about what your looking for. |
13:35.38 | Katty | anthm: can you put that into kat for me so i can understand what it's trying to tell me? |
13:35.41 | benjamin7062 | kpettit, hrrm, probably the only way would be .. hmm.. cranking up logging and grabbing it from a syslog? if that even works |
13:35.45 | *** join/#asterisk MACscr (n=MACscr@66.73.154.70) |
13:35.46 | BertZ | Hmm |
13:35.49 | MACscr | hello everyone |
13:36.22 | anthm | it means the nfs link was lost while something was using the file |
13:36.28 | MACscr | anyone know of a managed asterisk hosting provider? |
13:36.29 | a20060628 | ariel_ can you send me a link or something ? |
13:36.40 | BertZ | I'm trying to use queues. I don' use agents, only extensions (eg 2001,2002). I created my queue, but now how to statically put some users into my queue please ? |
13:36.41 | anthm | and windows is prtty lame about letting go of file locks |
13:36.46 | Katty | anthm: hmm. |
13:36.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
13:36.48 | ariel_ | ~google dailparties.agi |
13:36.57 | a20060628 | thanks |
13:37.08 | Hmmhesays | oh man these e2's kickass |
13:37.10 | anthm | so you gotta close every application that may have been using a file on that |
13:37.18 | anthm | before it will let you mount it again |
13:37.21 | Katty | anthm: i wasn't technically trying to open a file. i mount the directory, then change directories and do a dir. when i try to put a file in the directory and i get a permission denied, even though i'm root. |
13:37.25 | anthm | or reboot the box |
13:37.32 | Katty | anthm: oh ah. |
13:37.36 | Katty | anthm: that's something to try. i'll do ti. |
13:37.45 | anthm | nfs over windows is a pita |
13:37.46 | Hmmhesays | $100 bucks a set then damn well better be kickass |
13:37.59 | Katty | anthm: i'll give that a try. thanks for the heads up hun |
13:38.03 | anthm | they usually mount nfs as a weak user |
13:38.05 | BertZ | ~queues |
13:38.09 | *** join/#asterisk m4rk___ (i=mark@mir.stevecole.org) |
13:38.12 | BertZ | ~queue |
13:38.14 | jbot | Innovative load-balancing/batch-processing system and rsh replacement. URL: http://bioinfo.mbb.yale.edu/~wkrebs/queue.html |
13:38.14 | ariel_ | Hmmhesays, 2e's |
13:38.27 | anthm | you have to do some more stuff to mount it root in windows iirc it's bneen a while mine is still setup fro 6 years |
13:38.29 | Hmmhesays | Shure E2 ear monitors |
13:39.04 | benjamin7062 | anthm, not so bad if you get smb to auth using secure=domain or whatever... |
13:39.05 | Katty | anthm: well it /was/ working fine...was being the operative word. i was dumping various folders onto a win2k box with the same mount command. then we replaced that server with a win2k3 server... |
13:39.14 | *** join/#asterisk cef_ (n=cef@38.119.128.203) |
13:39.26 | benjamin7062 | anthm -- don't remember exactly but if it auths against AD basically |
13:39.29 | anthm | so you are using the windows disk from unix then ? |
13:39.31 | Katty | anthm: still trying to hash out the problems... i simply presumed it was a win2k/2k3 difference in permissions or something. |
13:40.15 | Katty | anthm: it's a cron job that mounts a win2k3 share, copies various asterisk folders to the mount point, then umounts it. then the 2k3 box does its usual backup |
13:40.15 | ariel_ | Hmmhesays, expensive ear devices. |
13:40.28 | Hmmhesays | ariel_: well worth it |
13:40.36 | anthm | yah it's gonna be 2k3 being lame |
13:40.39 | anthm | in that case |
13:40.48 | Katty | mew )= |
13:40.51 | m4rk___ | if i'm sending calls through a sip -> pstn gateway to a number that is busy should i be explicitly sending a busy back to my sip peer or will asterisk do it for me? |
13:41.08 | m4rk___ | what i mean is: can i just do a Dial() and have asterisk handle passing on the call progress |
13:41.21 | cef_ | wondering if anyone can help me with a trascodinbg problem in thr latest trunk... I am trying to use itso i can get jingle support |
13:41.25 | *** join/#asterisk fulgas (n=fulgas@82.102.2.30) |
13:41.30 | anthm | i was dumb enough to use 2k3 back in 2k3 so i havent looked at it much since |
13:41.38 | Katty | anthm: i'm going to try a few things, and google a bit. if i don't make any headway i'll pastebin some stuff and see if you can give me a hand. |
13:41.45 | dlynes_office | alucard064: anyways...if you're still there, try these links: http://www.ipdanmark.dk/ and http://www.ipdanmark.dk/IPSwitchBoard/IPswitchBoard%20Manual.pdf |
13:41.50 | redax | hi. |
13:41.57 | cef_ | apparently transcoding gets messed up in this version even for sip-sip |
13:42.11 | *** join/#asterisk userdefined (n=jross@cpe-24-169-142-23.rochester.res.rr.com) |
13:42.31 | cef_ | i nmy case i am trying to go from ilbc to g.711 alaw.. but the g.711 loops back and the ilbc is mute in both direction |
13:42.37 | Katty | anthm: i'm really getting sick of 2k3...but the company wants me to keep using it since i'm mcped on 2k3 stuff |
13:42.40 | redax | what's wrong with this: s,2,Dial(mISDN/g:Intern/101) ; s,3,Dial(mISDN/g:Intern/102) |
13:42.48 | Katty | anthm: i appear to be doomed ;) |
13:42.53 | redax | it should ring the ext 102, if the 101 is busy, right? |
13:43.14 | dlynes_office | cef_: i've only found that to be the case when you have canreinvite=no |
13:43.36 | dlynes_office | cef_: oh...nvm |
13:43.51 | dlynes_office | cef_: you're actually able to convert...it just doesn't sound right |
13:44.09 | dlynes_office | cef_: the problem i was thinking of was where it doesn't autonegotiate codecs |
13:44.18 | cef_ | yup.. i don't think that's my problem |
13:44.49 | dlynes_office | and anthm will probably say but freeswitch does that, no problems :) |
13:44.51 | cef_ | asterisk seems to know what i'm trying to do a 'show channel xxx' indicates the correct info including transcoding |
13:45.08 | dlynes_office | or not :) |
13:45.55 | cef_ | i verified this works fine in the branches/1.2 release |
13:46.03 | Persilon | can anyone help me with playtones ? |
13:46.04 | cef_ | it only brakes in the trunk |
13:47.31 | redax | or should I use the "j" parameter in Dial() to jump +101 if the Dialed party is busy ? |
13:48.21 | littleball | hello, who knows how to force an sip peer expire in asterisk CLI console? |
13:49.02 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
13:49.32 | *** join/#asterisk anna-- (n=nmuller@195.70.21.58) |
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13:50.19 | *** join/#asterisk ToyMan (n=stuq@74-32-9-135.dsl1.mdl.ny.frontiernet.net) |
13:50.46 | znoG | are my messages too long to bother reading or nobody has any suggestions? |
13:51.31 | fulgas | hi |
13:52.19 | Lord_Drachenblut | where can i find info on callfiles at |
13:52.52 | ariel_ | znoG, ??? |
13:53.07 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
13:53.36 | ManxPower | Lord_Drachenblut, Google the mailing lists, the Wiki, sample.call in the Asterisk source |
13:53.43 | znoG | ariel: i'm setting up hunt groups in * using just standard dialplan config, and I'm wondering how to deal with loops. eg. extension 1000 is set to forward to 1005 after 20 seconds. but 1005 is also set to forward to 1000 after 20 seconds. What would make sense is that when it hits the first extension (1000) it adds to a variable called "callroute" or something, so when 1005 gets the call, before forwarding to 1000 it checks if it was dialed prev |
13:53.45 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
13:54.16 | ManxPower | znoG, I think it's so location specific that we reallly can't tell you. your idea makes some sense. |
13:54.21 | asteriskmonkey | anyone know of some asterisk mail client software |
13:54.27 | nortex | Can some one help me diagnose why some polycom phones are auto answering. |
13:54.35 | ariel_ | znoG, you need what I call a rollover macro |
13:54.40 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
13:54.45 | ManxPower | nortex, no. Polycoms do not auto answer unless you tell them to. |
13:54.56 | ManxPower | the Wiki and the mailing list archives have info on making them autoanswer |
13:55.15 | ariel_ | ManxPower, morning |
13:55.17 | asteriskmonkey | avc was a good stand alone java client for asterisk but is no longer made |
13:55.17 | nortex | ManxPower, Oh, but they do in a queue with a sipheader. |
13:55.41 | ManxPower | nortex, um, adding a SIP header would be considered "telling them to" |
13:55.41 | *** join/#asterisk dec_ (n=tom@ppp164-96.lns3.adl4.internode.on.net) |
13:55.41 | *** join/#asterisk anna-- (n=nmuller@195.70.21.58) |
13:55.43 | znoG | ariel_: right, which would do more or less what I proposed? |
13:56.41 | ariel_ | znoG, do you have one you need help with? |
13:57.23 | nortex | Manx I add the same sipheader to the same phones before a dial command and they just ring. The sip.cfg class is set to ring not ring-answer. The phone handles it different when the sipheader is followed by a queue application. |
13:57.33 | znoG | ManxPower: yeah, but i thought more than one person would have ran into this. For example in a place where there are 2 receptions... say extension 5 and 10... if I dial 5 and they dont pick up, it calls ext 10... if 10 doesn't pick up, the call should go to voicemail (and vice versa if I ring 10 first). But if 10 is set to forward to 5, it would create an endless loop. |
13:57.47 | znoG | ariel_: not yet, haven't really implemented the idea, yet. |
13:57.51 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
13:58.18 | znoG | ariel_: i would have to run an AGI script, I guess, and keep adding to a variable like callpath ... ie. callpath=10,20,30 .. and split on comma, and check if the extension being dialed is part of the list or not. |
13:58.34 | ariel_ | znoG, here is an old one I did about 3 years ago. It will help you get started and kinda get the idea: http://pastebin.ca/73956 |
13:58.58 | ariel_ | znoG, use group counts in a macro |
13:59.04 | ariel_ | you don't need an agi for it. |
13:59.23 | *** join/#asterisk DrkShdw (n=DrkShdw@fl-209-26-20-205.sta.embarqhsd.net) |
13:59.33 | ManxPower | znoG, we don't let our users forward to internal extensions. End of problem |
14:00.12 | ManxPower | nortex, what makes you think the Queue app even sends the header you are setting? |
14:00.17 | ManxPower | I thought only DIAL did that. |
14:00.25 | znoG | ariel_: interesting, time to go to the url you pasted |
14:00.30 | znoG | ariel_: thanks |
14:00.38 | nortex | znoG, I have 3 receptionist all in a queue with ring-all. |
14:00.59 | ManxPower | Queue looks smart. It is not. |
14:01.06 | nortex | ManxPower, Becuase the ring tone is changed on the phone. |
14:01.21 | znoG | nortex: yeah, that could work I guess |
14:01.25 | ManxPower | nortex, then perhaps the PHONE config is different. |
14:03.23 | nortex | ManxPower, I checked it, I swear. Rebooted all 3 phones in the queue after changing the config and verified the same alert_info I use to distinguish a DID call was being used for calls from the queue. |
14:04.19 | nortex | ManxPower, If I called the did it worked perfect, ringtone 5 and no autoanswer, If I called the queue all 3 rang once then one randomly answered the call. Nobody was at the phone. |
14:04.21 | ManxPower | nortex, and you confirmed that the polycom config files on the FTP server are set up the same? you did a factory reset on the phones to clear all configs and have the phones download their config again? |
14:04.25 | iq | Good Morning |
14:04.35 | Katty | hey iq (= |
14:05.06 | iq | hey Katty - whats up |
14:06.36 | nortex | ManxPower, I did not do a factory reset, but all 3 phones use the same sip.cofg file and I rebooted them using the sip notify command which AFAIK checks the config and reloads it. I have an extra 601 I'm going to try to recreate the problem on. |
14:06.44 | *** join/#asterisk nXOR (n=drade@pdpc/supporter/sustaining/nXOR) |
14:06.53 | *** part/#asterisk a20060628 (n=fazakasc@86.35.34.63) |
14:06.59 | *** join/#asterisk anonymouz666 (i=anonymou@200.218.196.5) |
14:07.03 | Hmmhesays | gxp-2000 thoughts? |
14:07.05 | nortex | ManxPower, I mean the same sip.cfg (Polycom file) |
14:07.08 | nXOR | hi ppl is someone here to help me with a little problem |
14:07.21 | nXOR | i knwo its prolly pebkac but i need enlightning on the matter |
14:07.26 | ManxPower | nortex, Well, make sure all phones are loading that file, yes |
14:07.41 | ManxPower | Customer: We are on our way to Jackson MS, will you be available in 2 hrs? |
14:07.51 | ManxPower | It might be nice if I had less than 2hrs notice. |
14:07.54 | *** join/#asterisk eles (n=ls@dsl-145-238-228.telkomadsl.co.za) |
14:08.11 | dlynes_office | ManxPower: don't you mean more than? |
14:08.32 | ManxPower | dlynes_home, Correct. Apparently I didn't have enough coffee yet. |
14:08.37 | Katty | iq: nada, just trying to get some work done ;) |
14:08.43 | dlynes_office | ManxPower: and i've had too much |
14:08.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
14:08.51 | dlynes_office | ManxPower: still up from yesterday morning :( |
14:09.00 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
14:09.00 | *** mode/#asterisk [+o russellb] by ChanServ |
14:09.03 | nXOR | i have an asterisk configured two ip phones and 12 soft phones, internally i can place calls however when i try to make an outside call i get this error in debug asterisk cli: Everyone is busy/congested at thsi time |
14:09.08 | iq | Katty: :) |
14:09.11 | dlynes_office | Good morning, Russell! |
14:09.16 | ManxPower | dlynes_home, I don't do that anymore without large wads of cash in hand. |
14:09.19 | nXOR | and my soft client shows NAT/Firewall of an unknown type |
14:09.20 | russellb | good morning :) |
14:09.36 | nXOR | any pointers would be much appreciated |
14:09.41 | ManxPower | nXOR, you want to tell the SIP client there is no NAT. |
14:09.49 | ManxPower | Then use asterisk's built in NAT features |
14:09.52 | nXOR | ManxPower,nat=no ? |
14:10.09 | nXOR | ManxPower, shed some light on this please |
14:10.18 | ManxPower | nXOR, I don't know how to configure your softphone. All I can tell you is how to configure Asterisk. |
14:10.24 | nXOR | tell me that |
14:10.34 | nXOR | or at least point me to a resource |
14:10.43 | ManxPower | nXOR, Is Asterisk behind NAT? |
14:10.51 | nXOR | well no not really |
14:10.59 | *** join/#asterisk [TK]D-Fender (n=joe@CPE000d3a2c3061-CM00080d8dba84.cpe.net.cable.rogers.com) |
14:11.02 | nXOR | its like this --isdn line ----- asterisk box ----- lan |
14:11.11 | jbalcomb | ~websae |
14:11.15 | ManxPower | nXOR, NAT is like being pregnant. You can't be "sort of" |
14:11.17 | jbalcomb | ~seen websae |
14:12.00 | jbot | websae is currently on #asterisk (37m 52s), last said: 'get a VoIP provider and ATA/SIP phone'. |
14:12.02 | nXOR | maagic,no there is no nat, because nat is handled by my cisco dsl router |
14:12.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
14:12.02 | jbalcomb | nXOR NAT or PAT? |
14:12.02 | nXOR | ManxPower,the asterisk box taps directly to an isdn line via an TA |
14:12.03 | ManxPower | nXOR, and ALL phones and softphones are on the LAN |
14:12.03 | nXOR | ManxPower,indeed |
14:12.04 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
14:12.05 | nXOR | jbalcomb, neither for asterisk box |
14:12.06 | Dandre | Hello, |
14:12.08 | ManxPower | nXOR, then you do not have NAT. |
14:12.12 | nortex | ManxPower, If it should work, I mean set the alert_info then call the queue and have all the phones ring distinctively for that queue, then I am probably just having problems with the Polycom configs. |
14:12.25 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
14:12.27 | ManxPower | nortex, correct. |
14:12.32 | nXOR | ManxPower, so everythign should work ? |
14:12.51 | nXOR | but somehow it doesnt ...... maybe its the TA ...... |
14:12.54 | Dandre | I need some help with grandstream bt100 firmware update. Is there any one here that culd help me? |
14:12.56 | ManxPower | nXOR, Assuming you have all the other billion things that need to be set up correctly. |
14:12.58 | jbalcomb | nortex: you have your system set up to ring distinctively per queue? |
14:13.00 | nXOR | but ui see visdn raising up the visdn0 interface |
14:13.21 | nXOR | then closing it ..... |
14:13.30 | jbalcomb | Dandre: No one can know until you tell us what the problem is. |
14:13.32 | ManxPower | nXOR, So you have Softphone makes a call -> Asterisk -> HOW DOES ASTERISK DIAL THE PSTN? |
14:13.56 | nXOR | extensions.conf has the dialplan |
14:13.57 | ManxPower | nXOR, looks to me like you have an asterisk/ISDN problem that has nothing to do with the phones. |
14:14.24 | ManxPower | nXOR, and I can't help you with ISDN since I'm in the USA. |
14:14.25 | nXOR | yeah i think so too, but i cant really pin point it, do you knwo of any good visdn + asterisk tuts ? |
14:14.40 | nortex | jbalcomb, I only have one queue, but that is what I hoped for. |
14:14.50 | *** part/#asterisk m4rk___ (i=mark@mir.stevecole.org) |
14:15.00 | nXOR | i cant fidn any good ones online, ones that i find are scarce |
14:15.21 | Dandre | jbalcomb: I have downloaded the latest firmware from grandstreaam website for bt100, put it on my tftp server and rebooted my bt100. Then I am unable to connect to the http config server on the bt100 |
14:15.22 | ManxPower | nXOR, No. I'm in the USA. I think there are 3 people in the entire country that have any kind of ISDN BRI. |
14:15.36 | eles | lo all, does anyone know if it is possible to transfer a call in progress to another extension via console ? |
14:15.38 | nXOR | should i forward traffic from lan1 to visdn with iptables ? |
14:15.45 | Dandre | I have a blank |
14:15.48 | Dandre | page |
14:15.53 | littleball | hello, for sip, CLI#sip show peers. what is the difference for status : UNKNOW and UNREACHABLE? what packets change the status from "UNREACHABLE" to "OK"? |
14:15.53 | ManxPower | eles, you cannot. |
14:16.13 | ManxPower | eles, you can via the manager interface (aka AMI) |
14:16.23 | nortex | jbalcomb, Our operators tend to ignore the first few rings on phone calls forwarded to them because it rings the same as a call from outside the company. |
14:16.50 | ManxPower | littleball, unknown = phone is not registered. Unreachable = phone was registered, but now it's not responding, OK = phone is registered and is responding |
14:17.11 | anthm | eles, app_changrab from http://www.pbxfreeware.org |
14:17.17 | ManxPower | unrechable and OK only happen when you use the qualify= option |
14:17.31 | littleball | ManxPower, how to define "Unreachable" and not responding? Responding to what? |
14:18.07 | ManxPower | littleball, not responding to the SIP OPTIONS packet Asterisk sent |
14:18.31 | littleball | ManxPower, thanks. I think the same and let me test now |
14:19.37 | eles | ManxPower: know where i can maybe get a reference or some documentation for the AMI ? I have tried looking but havent really found anything nice |
14:19.41 | jbalcomb | nortex: yes, i understand completely. I would very much like to set that up for my call centers. Is there a page on the wiki that covers this setup or do you have some information? |
14:19.43 | eles | anthm: tanks i will take a look |
14:20.04 | ManxPower | eles, that would be on the Wiki and in the Asterisk source, prolly /path/to/src/asterisk/docs |
14:20.10 | littleball | ManxPower, i found that OPTIONS from asterisk to peer packet has the same CSeq: 102 |
14:20.11 | jbalcomb | Dandre: Does the phone work otherwise? Have you port scanned the phone to see if anything is responding? |
14:20.14 | littleball | why? |
14:20.57 | littleball | ManxPower, i found that OPTIONS from asterisk to peer packet has the same CSeq: 102. For all options to a specific peer, the CSeqs have the same value |
14:21.02 | Dandre | jbalcomb: the phone seems to work as before but doesn't register to my asterisk box |
14:21.36 | anonymouz666 | anthm: lots of apps in that site |
14:22.33 | nortex | jbalcomb, I used some of the information about the Polycom autoanswer to do distinctive ring on my polycom 501/601 phones. But I have hit a wall of sorts when I use the the sipheader and queues, but I have not figured out why. |
14:22.41 | anthm | yah that's where i keep my out of tree apps |
14:22.41 | *** join/#asterisk snowy_owl (i=0@200.218.196.2) |
14:23.18 | nortex | jbalcomb, I can pastebin what I have at this point if you want. |
14:23.24 | snowy_owl | Those who know me have no need of my name |
14:23.26 | jbalcomb | nortex: Hrmm.. can I get a copy of the relevant configs? |
14:24.31 | jbalcomb | nortex: yeah, a pastebin would be great |
14:25.00 | *** join/#asterisk anna-- (n=nmuller@195.70.21.58) |
14:27.40 | anonymouz666 | snowy_owl: that's for sure |
14:29.19 | jbalcomb | nortex: if pastebin.com is still down you can use http://sial.org/pbot/ |
14:30.48 | Dandre | jbalcomb: only 80 and 4144 ports respond to a port scan |
14:31.48 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
14:31.49 | a1fa | yo |
14:32.11 | a1fa | i found a site a month ago that offered $1/did + $0.01 per minute |
14:32.20 | a1fa | 25 calls max at one time |
14:32.34 | a1fa | anybody know the site i am talking about |
14:33.11 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
14:33.57 | MACscr | alfa, you might want to check your browser history or next time bookmark it =P |
14:34.42 | dlynes_office | MACscr: dood...don't be making sense...that's not allowed |
14:35.01 | MACscr | lol, sry =P |
14:35.17 | a1fa | lol |
14:35.23 | a1fa | nah.. i dont know where i put it |
14:36.05 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
14:38.19 | *** join/#asterisk mog (n=mogorman@gateway.digium.com) |
14:39.05 | MACscr | you cant search your browser history? |
14:40.06 | MACscr | dlynes: you seem pretty logical, have you heard of anyone offering asterisk hosting and management? |
14:40.20 | MACscr | basically "managed asterisk hosting" |
14:40.22 | a1fa | MACscr : this was 2 months ago |
14:40.28 | a1fa | MACscr : i do |
14:40.33 | MACscr | google hasnt turned up anything |
14:40.36 | a1fa | MACscr : $5/month |
14:40.45 | a1fa | msg |
14:40.47 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
14:40.57 | dlynes_office | MACscr: nope |
14:41.09 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.94.Dial1.SanJose1.Level3.net) |
14:41.15 | dlynes_office | MACscr: however, i have heard of people doing xen hosting |
14:41.24 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.94.Dial1.SanJose1.Level3.net) |
14:41.27 | jbalcomb | Dandre: does the phone work? |
14:41.30 | dlynes_office | MACscr: it's not well suited to asterisk, but some people have it working |
14:41.41 | *** join/#asterisk blebleble (n=ble@d149-67-99-160.col.wideopenwest.com) |
14:42.11 | Dandre | jbalcomb: it doesn't register to my asterisk box but I can place calls |
14:42.21 | littleball | what is xen? |
14:42.34 | jbalcomb | Xen is a virtual server technology |
14:42.35 | dlynes_office | littleball: virtual machines |
14:42.48 | littleball | ok |
14:43.00 | MACscr | my issue is that i dont have the time or patience to setup asterisk they way i want it |
14:43.09 | dlynes_office | littleball: as is uml and vps |
14:43.19 | jbalcomb | MACscr so just contract [TK]D-Fender to do it for you |
14:43.22 | MACscr | and im needing to custom of a configuration to do any vanilla service |
14:43.22 | dlynes_office | littleball: user mode linux, and virtual private server |
14:43.50 | dlynes_office | MACscr: why do you need to custom configure everything? |
14:44.30 | jbalcomb | Dandre: I assume you have rebooted the phone since? |
14:44.40 | Dandre | sure |
14:44.48 | littleball | dlynes_office, does it mean xen runing on a few clustered hardware? |
14:44.57 | dlynes_office | littleball: not necessarily |
14:45.03 | dlynes_office | littleball: could be all on one machine |
14:45.04 | *** part/#asterisk bkw_ (n=bkw_@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net) |
14:45.07 | MACscr | well, 1) im running multiple companies with the same staff, which the staff are remote |
14:45.18 | dlynes_office | MACscr: ok |
14:45.24 | dlynes_office | MACscr: and? |
14:45.27 | littleball | dlynes_office, does it mean xen runing on a few clustered hardware? i knoow not necessary. but want to know whether can run on clusterd multiple server. |
14:45.32 | MACscr | so i need caller id rewritten so the staff member knows what company he is answering for |
14:45.38 | MACscr | i also need to record all the calls |
14:45.49 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
14:45.49 | dlynes_office | MACscr: how many different companies? |
14:45.55 | MACscr | only 2 right now |
14:46.06 | MACscr | and only a handful of extensions |
14:46.07 | jbalcomb | littleball: Xen allows you to run multiple servers on one piece of hardware |
14:46.07 | dlynes_office | MACscr: do you expect it to be more than three? |
14:46.11 | MACscr | as in 6 at most right now |
14:46.21 | dlynes_office | MACscr: i.e. more than three companies? |
14:46.25 | MACscr | actually i only have 2 for sure |
14:46.28 | MACscr | not right now |
14:46.34 | MACscr | but i definately see it happening |
14:46.44 | dlynes_office | MACscr: you can handle that with three lines if you want, too |
14:46.48 | jbalcomb | MACscr: I have 5 companies on my system but only 7 remote users. Also, have two call centers with 17 call queues |
14:46.56 | dlynes_office | MACscr: one company per phone line |
14:47.06 | dlynes_office | MACscr: so when you see line 1 ringing, you know it's for company #1 |
14:47.09 | MACscr | right, i was planning on going with 3 lines |
14:47.20 | MACscr | 2 lines for 2 companies and 1 for fax |
14:47.36 | dlynes_office | MACscr: a line on a voip phone is not the same as an analog line |
14:47.39 | littleball | jbalcomb, running multiple servers on one piece of hardware is not so urgent. because hardware is so cheap. maybe clustering is more important to support big deployment and take away the scalability issue from normal design |
14:47.44 | dlynes_office | MACscr: it's not a 1:1 mapping |
14:47.44 | jbalcomb | dlynes_home MACscr: Why different lines? Why not just different phone numbers and match the DID? |
14:48.24 | dlynes_office | jbalcomb: then the remote user knows which company the call is for at a glance (or by the sound of the ring tone), without having to look at the caller id |
14:48.32 | MACscr | sry, meant lines |
14:48.34 | MACscr | woops |
14:48.37 | MACscr | numbers i mean |
14:48.37 | jbalcomb | littleball: I can't say for you there. I have two Xen servers that have 4 virtual servers each, dchp, iDNS, eDNS, and LDAP. |
14:48.46 | [TK]D-Fender | MACscr : Have you picked your phone already? |
14:48.59 | littleball | jbalcomb, yes.cater different requirement |
14:49.07 | MACscr | Fender, i went cheap and got a GXP-2000 for myself |
14:49.16 | jbalcomb | littleball: Saves 2 to 4 grand on machines, saves on rediculous wastes of hardware, saves on rack space, and lightens the heat load on the AC. |
14:49.20 | littleball | jbalcomb, reliable? |
14:49.22 | MACscr | but i got a polycom 501 right beside me and im quite impressed |
14:49.33 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
14:49.43 | [TK]D-Fender | MACscr : You should be able to configure individual registrations against the line keys and just have it ring on a distinct line thereby identifying the kind of caller |
14:49.49 | a1fa | jbalcomb : xeon is good |
14:49.56 | a1fa | jbalcomb : i got couple of xeon servers |
14:50.09 | a1fa | <PROTECTED> |
14:50.20 | jbalcomb | dlynes_office MACscr: yeah but you can do that based on the DID just as easy as line |
14:50.25 | dlynes_office | jbalcomb: you can do all that by setting four instances each of each server, listening on different interfaces, too |
14:50.32 | littleball | alfa, jbalcomb, is it reliable? and is it compatible with existing linux applications? |
14:50.32 | peterm22 | anyone using the new sipura one phones ? |
14:50.34 | a1fa | of course |
14:50.36 | a1fa | it is i386 |
14:50.38 | [TK]D-Fender | MACscr : Line 1-2 = Company A, 3-4 = Company B |
14:50.51 | dlynes_office | jbalcomb: yeah...I was just thinking doing it by the lines is less kludgy |
14:50.56 | *** join/#asterisk N3WWN (n=N3WWN@ns1.futuretek.cx) |
14:51.02 | dlynes_office | jbalcomb: then the caller id is left untouched |
14:51.05 | littleball | alfa, you mean the virtuam server is exactly one i386 box? |
14:51.16 | a1fa | well yeah |
14:51.21 | jbalcomb | dlynes_office: but if you do it based on DID then each company can use all the lines |
14:51.22 | a1fa | its like running wmvare on your box |
14:51.29 | a1fa | vmware* |
14:51.36 | a1fa | same thing |
14:51.48 | a1fa | on a 3ghz zeon with 10gb of ram |
14:51.49 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) |
14:51.50 | nortex | jbalcomb, http://pastebin.com/734897 |
14:51.56 | a1fa | you can run 20 virtual servers if not more |
14:52.00 | [TK]D-Fender | jbalcomb : not by PSTN lines, but PHONE "lines" or "appearances" |
14:52.13 | jbalcomb | littleball: it is quite reliable and natively support by Intel and AMDs VT extensions. If you have the right CPU you can run windows and linux virtual servers on the same machine. |
14:52.13 | dlynes_office | jbalcomb: you've lost me...how is doing it by did, not modifying the callerid for incoming calls? |
14:52.28 | nortex | jbalcomb, I just had it work on a fresh phone. |
14:52.40 | jbalcomb | [TK]D-Fender: ah, guess I'm tryign to do too much at once to keep it all straight. |
14:52.49 | dlynes_office | jbalcomb: or are you getting call appearances and pstn lines mixed up? |
14:52.51 | littleball | alfa, how about hardware resource like udp/tcp port? |
14:53.12 | a1fa | littleball : depends, you can assign them each individual ethernet |
14:53.22 | Dandre | jbalcomb: I have put 1.0.8.12 version on my tftp server and now I have recoverd admin console but it shows: |
14:53.24 | Dandre | Program-- 1.0.6.8 Bootloader-- 1.0.8.9 HTML-- 1.0.6.8 VOC-- 1.0.1.0 |
14:53.25 | littleball | alfa, true |
14:53.27 | a1fa | or you can use 1 ethernet -> nat/routing -> |
14:53.36 | dlynes_office | MACscr: these are all inbound calls, right? |
14:53.38 | a1fa | Dandre : reboot your phone again |
14:53.45 | a1fa | Dandre : 1.0.8.23 is out |
14:53.57 | a1fa | Dandre : i hate grandstream.. so many issues with their firmware |
14:54.04 | littleball | anyone using Firefly? it seems it doesn't reply OPTIONS packet sent by asterisk |
14:54.08 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
14:54.15 | littleball | s/reply/response |
14:54.31 | znoG | s/response/respond |
14:54.37 | znoG | :) |
14:54.40 | littleball | hehe |
14:54.41 | Dandre | but I have had issues with the admin console with 1.0.8.23: I had no access to it |
14:54.57 | a1fa | they dont recomend you to downgrade |
14:54.58 | MACscr | dlynes: 98% of traffic will be inbound |
14:55.19 | littleball | znoG, so, other peer cannot call Firefly if it is behind firewall |
14:55.22 | N3WWN | Anyone having "channel.c:787 channel_find_locked" errors with 1.2.8 and the vpb channel driver? |
14:55.28 | dlynes_office | MACscr: so you don't mind prefixxing outbound calls with a certain digit to differentiate between different phone lines then, right? |
14:55.39 | jbalcomb | a1fa: are you the guy that was working on an auto-provisioning system for IP phones? |
14:55.54 | dlynes_office | MACscr: i.e. to force asterisk to dial out on a specific analog line? |
14:56.16 | MACscr | dlynes: i dont see that being an issue, but im also not going to be using any analog lines |
14:56.16 | a1fa | jbalcomb : no |
14:56.25 | dlynes_office | MACscr: oh, ok |
14:56.36 | [TK]D-Fender | dlynes_office : Again not needed with seperate call appearances. |
14:56.37 | dlynes_office | MACscr: what kind of lines will they be then? pri, or voip? |
14:56.47 | MACscr | sry, i come from a PTSN background when it comes to phone systems |
14:56.59 | a1fa | jbalcomb : i did upgrade firmware on my remote locations in austria and bosnia :) |
14:56.59 | MACscr | so i apologize if i dont use the right logic sometimes |
14:57.06 | [TK]D-Fender | dlynes_office : My home IP 501 has 1 line key / customer of mine I'm working with actively supporting 4 calls at a time each. |
14:57.08 | littleball | who can recommend a cheap but good :-) router to seperate voice traffic and data traffic in the office? |
14:57.26 | a1fa | jbalcomb : both phones were behind two firewalls and two nats.. nasty, but i was able to upgrade them remotely |
14:57.28 | dlynes_office | [TK]D-Fender: yeah, i know what call appearances are |
14:57.49 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
14:57.51 | dlynes_office | [TK]D-Fender: but at the same time, sometimes you just want to grab the first available line, too |
14:58.01 | dlynes_office | [TK]D-Fender: and you don't care which one it is |
14:58.07 | *** join/#asterisk masonf (n=masonf@dungle.vineyard.net) |
14:58.09 | dlynes_office | [TK]D-Fender: other times you might want to grab a specific line |
14:58.11 | a1fa | intelligent call routing |
14:58.24 | masonf | did asterisk cvs change recently? |
14:58.25 | dlynes_office | a1fa: yes, i already use that |
14:58.29 | a1fa | if cost <> "" { use } |
14:58.41 | dlynes_office | a1fa: not an issue for cost |
14:58.50 | a1fa | well, availability and realibility |
14:58.51 | dlynes_office | a1fa: for an issue of which did you want showing up on the customer's display |
14:59.00 | [TK]D-Fender | dlynes_office : not so effective when you want to set your outbound callerID to indicate which business division you are calling from... |
14:59.00 | *** join/#asterisk jhb (n=joerg@yes.mediathek.de) |
14:59.08 | dlynes_office | a1fa: so that they know which business you're calling from |
14:59.13 | tzanger | cool, there's a virtual tour of my house |
14:59.20 | a1fa | dlynes_home: callerid not enuff? |
14:59.27 | tzanger | http://www.venturehomes.ca/VirtualTour.asp?TourID=5956 |
14:59.33 | littleball | who can recommend a reasonable router to seperate voice traffic and data traffic in the office? |
14:59.36 | dlynes_office | [TK]D-Fender: yeah, but if you're using all analog lines, and you cannot set your callerid, then it is an issue |
14:59.50 | a1fa | littleball : cheap? |
14:59.53 | a1fa | or good? |
15:00.02 | littleball | reasonable, alfa |
15:00.12 | dlynes_office | littleball: any router that supports vlans |
15:00.13 | a1fa | no such thing :) sacrifice something |
15:00.25 | a1fa | any router that, well routes |
15:00.26 | littleball | the budget is below 500USD |
15:00.31 | a1fa | littleball : hahah |
15:00.39 | a1fa | littleball : you can use QOS if you need to |
15:00.51 | a1fa | routing+qos via packetshaping router |
15:00.52 | dlynes_office | a1fa: qos doesn't separate it...it only controls it |
15:00.57 | littleball | small office (30 persons). i need to improve the voice quality |
15:01.10 | a1fa | dlynes_home: right, qos to improve pass-through |
15:01.13 | nortex | tzanger, I have in-laws in ontario whose house is almost identical, must be a Canadian thing. |
15:01.21 | a1fa | you can use anyswitch that supports VLANS |
15:01.23 | dlynes_office | littleball: qos to control it, vlans to separate it, and sip jitterbuffer if you've got huge files going across |
15:01.27 | a1fa | you dont even need routers |
15:01.39 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
15:01.50 | [TK]D-Fender | dlynes_home : And you also don't want to mess with kludgy prefixes to choose your "identity" either.. that forgoes the idea of "just grab and dial" as well.. and adds to the list of things to remember |
15:01.50 | a1fa | dlynes_home: he wont need a router.. pick up a good cisco switch, or HP switch |
15:01.54 | tzanger | nortex: probably just a common house plan for that time |
15:02.00 | a1fa | that supports .q and qos |
15:02.04 | littleball | alfa, the problem is that if a few people retrieve emails, the voice quality become bad |
15:02.17 | a1fa | what firewall you got? |
15:02.24 | a1fa | maybe you can qos through your firewall |
15:02.27 | a1fa | TOS/QOS |
15:02.31 | littleball | no idea, it is in customers side |
15:02.34 | dlynes_office | littleball: with huge attachments? |
15:02.35 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
15:02.38 | littleball | yes |
15:02.44 | a1fa | block email access |
15:02.46 | a1fa | :) |
15:02.47 | littleball | not huge, 2-to 5 m |
15:02.57 | a1fa | littleball : let me guess, the customer has DSL/CABLE? |
15:02.59 | dlynes_office | littleball: look into patching in the sip jitterbuffer from trunk |
15:03.14 | a1fa | i'm out |
15:03.15 | a1fa | meeting |
15:03.19 | a1fa | bbl |
15:03.28 | littleball | DSL |
15:03.39 | a1fa | littleball : DSL and what kind of phone? |
15:03.49 | a1fa | littleball : sipura firmware supports QOS in box |
15:03.53 | littleball | no branding :-) get from Taiwan |
15:04.01 | a1fa | ok, that could be your problem |
15:04.13 | a1fa | my linksys router supports QOS |
15:04.28 | a1fa | too bad halflife2 is QOS-ed on my home network |
15:04.51 | a1fa | no need to QOS voip sinc i got 6mbits :) |
15:04.51 | N3WWN | littleball: have you looked into MikroTik? |
15:05.00 | littleball | does i need to configure QOS? |
15:05.07 | a1fa | you may have too |
15:05.15 | littleball | so that voice has higher priority than data |
15:05.17 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
15:05.34 | MACscr | alfa, you have 6mbits upload? |
15:06.09 | littleball | N3WWN, what is MikroTik? :-( google now |
15:06.54 | N3WWN | littleball: Linux based router options. You can get embedded linux systems (RouterBOARD 500, etc) made by MikroTik or install the software on a PC |
15:06.58 | littleball | i see. no, i cannot change router for customers until they agree |
15:07.11 | littleball | maybe try it out |
15:07.12 | *** join/#asterisk assert_true (n=Sunil@59.176.57.79) |
15:07.34 | N3WWN | littleball: SOHO license is free and they work like a charm! |
15:07.59 | littleball | N3WWN, does it support wifi? |
15:08.20 | N3WWN | yup... AP, client, WDS, bridge, etc |
15:08.38 | littleball | did u try it before? |
15:08.40 | littleball | good? |
15:09.07 | littleball | how much is the total cost for such a router? |
15:09.27 | littleball | i like the wifi option. it is interesting |
15:09.35 | N3WWN | littleball: you may need a pay license for wifi abilities, but my company uses them quite a lot. We use them for customer router, core routers, edge routers, remote network admin/troubleshooting devices, etc |
15:09.52 | *** join/#asterisk fourcheeze (n=rich@82.153.23.79) |
15:10.08 | fourcheeze | Hmmhesays: got your mixer working yet? |
15:10.11 | *** join/#asterisk glLoadIdentity (n=tyn@81.214.255.57) |
15:10.18 | Hmmhesays | didn't work on it last night |
15:10.26 | Hmmhesays | sex0red the girlfriend and play some disc golf |
15:10.30 | fourcheeze | you've made me want to get one now |
15:10.32 | littleball | N3WWN, did your company pay for wifi abilities? |
15:10.33 | N3WWN | littleball: I use a distributor in the US and can get a RB534 (3 eth + 1 or 2 mini-PCI) with case and power supply for about $250 |
15:10.45 | fourcheeze | (mixer not girlfriend) |
15:10.47 | littleball | cheap |
15:11.12 | Hmmhesays | lol |
15:11.16 | N3WWN | If you want to take more about them, can you msg me offline? We're off-topic here ;) |
15:11.34 | littleball | with QoS? N3WWN. i need to improve voice quality urgently |
15:12.25 | N3WWN | littleball: yes, you can prioritize many ways |
15:12.33 | Nugget | I've found QoS to be invaluable on my consumer dsl (768 up, 6m down) |
15:12.40 | Nugget | I wouldn't try to use voip without it |
15:13.04 | Persilon | is there a s-UNAVAILABLE extension ? |
15:13.45 | littleball | N3WWN, thanks. i will check and get one |
15:14.01 | N3WWN | No problem, littleball... |
15:14.06 | littleball | Nugget, must i go to customer size to configure the QoS? |
15:14.17 | littleball | i need to check with them about their router |
15:14.19 | Nugget | I don't understand your question. |
15:14.36 | littleball | Because i have no idea what router the customer using now. |
15:14.51 | N3WWN | Anyone have experience with VPB channels? |
15:15.13 | littleball | maybe their router has QOS. I am not familiar with such thing. So just ask whether QOS need to be configured |
15:15.27 | Nugget | How would I know? |
15:22.16 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
15:24.51 | *** join/#asterisk florin2703 (n=aaa@florin-jurma.tm.ew.ro) |
15:28.44 | *** join/#asterisk Vorondil (n=cwaldeck@mail.yhamerica.com) |
15:29.46 | Persilon | I need some help with s-NOANSWER and s-BUSY, I can't get them to work |
15:30.13 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
15:30.21 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-92-225.cybersurf.com) |
15:31.08 | znoG | exten => s-NOANSWER,1,.... doesnt work? |
15:31.15 | znoG | Persilon: be more specific |
15:31.35 | *** join/#asterisk salviadud (n=ralfalfa@201.145.29.99) |
15:31.47 | Persilon | znoG: in my dialplan I have exten => s-NOANSWER,1,Voicemail(${MACRO_EXTEN}@default,b) |
15:32.24 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
15:32.36 | Persilon | znoG: but it doesn't seem to work... if I don't pickup, it hangs up |
15:32.55 | Qwell | Persilon: Do you have a Goto(s-${DIALSTATUS}) anywhere? |
15:33.38 | Persilon | Qwell: nope, it should go at the top of the dialplan ? |
15:33.57 | Qwell | Persilon: read the example in macro-stdexten |
15:35.09 | Persilon | Qwell: thank you |
15:35.16 | znoG | Persilon: you probably need to do the Goto |
15:35.20 | znoG | oh, Qwell jumped in :) |
15:35.29 | Persilon | znoG: yes, thank you :) |
15:35.35 | *** join/#asterisk tamp4x (n=tampon@64.201.13.51) |
15:35.47 | Persilon | I have another question regarding playtones() I can't get it to work either, when I call from a pstn phone |
15:37.46 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
15:39.55 | *** join/#asterisk brijn (n=brijnier@204.244.176.116.net-conex.com) |
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15:43.27 | Persilon | Qwell: I put the s-${DIALSTATUS} but when I dial I get this on the console: Goto (macro-interns,s-,1) |
15:44.59 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
15:45.04 | lilalinux | does the register command support md5 passwords, too? |
15:45.05 | Persilon | Qwell: nevermid, solved :) |
15:45.22 | smackus | has anyone had an issue with mixmonitor killing the asterisk server? |
15:45.33 | BekokBau | can someone help me what is the problem with my configuration? here is my SIP debug http://pastebin.ca/74022 |
15:45.39 | dlynes_office | smackus: yes |
15:45.46 | dlynes_office | smackus: get over it...go back to using monitor with sox |
15:47.14 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
15:47.55 | [TK]D-Fender | Persilon : pastebin your macro |
15:47.58 | [TK]D-Fender | ~pb |
15:48.00 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
15:48.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
15:48.28 | Persilon | [TK]D-Fender: I forgot to uncomment the dial statement, it's solved |
15:48.56 | Persilon | [TK]D-Fender: is there anyway of make rings when a channel is unavailable ? I can't get playtones to work |
15:50.31 | [TK]D-Fender | Persilon : "Ringing" |
15:51.01 | Persilon | [TK]D-Fender: let me try, but the manual said it didn't make a ring on the phone, playtones was for that |
15:51.08 | *** part/#asterisk InfraRed (n=subhi@bb-87-81-46-122.ukonline.co.uk) |
15:53.55 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
15:54.38 | znoG | i wonder if there are any T.38 fax providers with DIDs in Argentina... could make a neat solution to offer a fax-to-email solution |
15:55.08 | jbalcomb | I need a project name for my IP phone provisioning system. Ideas? |
15:55.22 | anna-- | how can I force the MSN when I make a dial-out call through the Zap interface? I've tried to put msn=xxx in zapata.conf but it still uses the other MSN (I got 1 BRI/2 MSN) |
15:55.44 | masonf | does anyone know the asterisk cvs servers address? |
15:56.08 | Qwell | masonf: the asterisk cvs servers are dead, and removed from dns |
15:56.11 | symlink | we don't use CVS anymore |
15:56.13 | Qwell | svn.digium.com |
15:56.59 | znoG | could be an interesting idea... get a solution together that can relay T.38 and offer it in this country |
15:57.33 | coppice | T.38 seems to be flavour of the month |
15:57.39 | *** join/#asterisk sangee (n=rkuru@206.191.114.66) |
15:58.06 | znoG | yea, don't know why people insist on using faxes |
15:58.08 | smackus | where do i put the m flag for the monitor command? |
15:58.23 | smackus | nevermind... found it |
15:58.30 | coppice | znoG: low IQ |
15:59.38 | Persilon | is there any way of knowing how many parameters were passed when calling a macro ?? |
15:59.57 | sangee | I am using database (realtime) registration but when i issue the sip show peer username, it does not show the user, so i can not call to that extension, anyone know how to fix it? |
16:00.11 | giesen | yay |
16:00.19 | sparkleytone | are the asterisk people the same people who program the GXP-2000 firmware? |
16:00.21 | symlink | sangee: sip show peer <name> load |
16:00.23 | jbalcomb | "ZIPP: Zee IP Phone Provisioner" |
16:00.28 | jbalcomb | sparkleytone no |
16:00.30 | sparkleytone | i just noted that it invites you to this channel on its page |
16:00.34 | sparkleytone | on voip-info |
16:00.40 | giesen | it looks like some punk at e164.org decided it'd be a good idea to add a wildcard for toll free numbers to go through voipmich |
16:00.56 | jbalcomb | sparkleytone voip-info is not grandstreams page |
16:01.07 | sparkleytone | yes i know jbalcomb ...i meant the wiki page |
16:01.11 | sangee | ok |
16:01.18 | sparkleytone | which has WAY more info than grandstream's site ;) |
16:01.38 | sparkleytone | i want to like this phone...but the firmware makes it impossible |
16:01.59 | sangee | when i dial to that extension it does not dialing? may be i need to update the database? |
16:02.00 | jbalcomb | sparkleytone the phone makes it improssible |
16:02.01 | nortex | Is there a way to build meetme rooms dynamicly in 1.2.9.1 ? |
16:02.20 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
16:02.51 | sparkleytone | jbalcomb: touch? i guess...the hardware must be crap... |
16:02.53 | sangee | it registered okay, i can make outbound call to someone, but when someone call to that extension it does not dialing |
16:03.18 | jbalcomb | ok, unless there are any objections, I'm going to name my provisioning system ZIPP (Z IP Phone Provisioner) |
16:03.32 | *** join/#asterisk viler (i=1000@200.114.70.228) |
16:03.38 | jbalcomb | sparkleytone: the Polycom SoundPoint IP 501 is 95% better |
16:03.44 | sparkleytone | i object...its way too good of a name for anything based on OSS ;) |
16:03.55 | sparkleytone | jbalcomb: how much is it, ballpark? |
16:04.01 | jbalcomb | sparkleytone 200 |
16:04.06 | jbalcomb | with PoE |
16:04.13 | sparkleytone | yeah...cisco phones cost that much |
16:04.29 | sparkleytone | meaning my boss would just make us buy the cisco phone |
16:04.36 | sparkleytone | i'll look it up tho |
16:04.39 | jbalcomb | sparkleytone the cisco phones cost more |
16:05.13 | jbalcomb | sparkleytone of course you might check out /linksys/sipura |
16:06.06 | sparkleytone | yeah |
16:06.15 | jbalcomb | sparkleytone plus it depends on how many lines you need, whether or not speakerphone is important, and what you voice quality requirements are. |
16:06.17 | sparkleytone | i saw they are finally getting some cheaper products in |
16:06.42 | jbalcomb | ~seen avibani |
16:06.45 | jbot | i haven't seen 'avibani', jbalcomb |
16:06.54 | sparkleytone | i wish we could just rip out the entire existing phone system and replace it with a system designed from scratch |
16:06.57 | [TK]D-Fender | IP430 = $170 incl PoE and suits most peoples needs |
16:07.05 | nortex | sparkleytone, Cisco phone plus the license cost much more then Polycom. |
16:07.07 | jbalcomb | [TK]D-Fender polycom? |
16:07.11 | [TK]D-Fender | ~[av]bani |
16:07.14 | sparkleytone | having to work with the existing black-magic makes it ridiculous |
16:07.15 | [TK]D-Fender | jbalcomb : Correct. |
16:07.32 | [TK]D-Fender | jbalcomb : Go check it out... amazing little deal |
16:07.33 | nortex | sparkleytone, Plus 30 bucks for a power adapter. |
16:07.39 | jbalcomb | ~seen [av]bani |
16:07.41 | jbot | [av]bani <n=[av]bani@washuu.anime.net> was last seen on IRC in channel #asterisk, 72d 17h 42m 6s ago, saying: 'robin_sz: how are they?'. |
16:07.47 | sparkleytone | when i saw 'we' i mean...everyone...as in get rid of PSTN all together and redesign the entire communications system. |
16:07.53 | jbalcomb | wow, that guy disappeared |
16:08.32 | jbalcomb | he was working on auto configuring gxp-2000s |
16:09.02 | nortex | jbalcomb, When will ZIPP be aVALIABLE ??? |
16:09.30 | Katty | hey mister fender (= |
16:09.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
16:10.02 | *** part/#asterisk moonwick (n=moonwick@core.dump.net) |
16:10.12 | jbalcomb | nortex: if i have to do it all by myself it'll probably take 3 months |
16:11.09 | stephane_ | re |
16:11.31 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
16:12.04 | jbalcomb | feel free to join #ZIPP for more discussion and colaboration on ZIPP |
16:12.14 | nortex | jbalcomb, :( I figured since you were picking a name it was almost done. |
16:12.56 | jbalcomb | nortex: HA! Picking the name comes first so as to encourage the procrastination! |
16:13.15 | jbalcomb | "can't start this project until I come up with a great name!" |
16:13.40 | [TK]D-Fender | Katty: Mew. |
16:14.36 | *** join/#asterisk saftsack (n=saftsack@p54A7DF23.dip.t-dialin.net) |
16:15.40 | saftsack | hi |
16:15.46 | saftsack | where to find hylafax experts? |
16:16.18 | dlynes_office | saftsack: maybe try #hylafax? |
16:17.43 | saftsack | there isnt such channel ^^ |
16:21.02 | Mw3 | hi. do you know about some analog -> isdn converter. i have 2 analog gsm adapters and a bri card in my asterisk server. i would like to convert the 2 analog gsm to a bri and plug into my bri card |
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16:27.20 | jbalcomb | the IP430 is only $10 cheaper than the IP501 |
16:28.01 | jbalcomb | Does this "Integrated IEEE 802.3af Power over Ethernet support" mean I dont need the $20 PoE cable? |
16:28.38 | Katty | [TK]D-Fender: how's it goin, hun? any better? |
16:29.56 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
16:30.14 | *** join/#asterisk userdefined (i=jr000430@shell1.phx.gblx.net) |
16:30.33 | [TK]D-Fender | Katty : Somewhat. Come to the conclusion that I have to change soon and "just do it". |
16:30.43 | [TK]D-Fender | Katty : on vacation now, and just vegging away |
16:30.52 | *** part/#asterisk userdefined (i=jr000430@shell1.phx.gblx.net) |
16:31.13 | [TK]D-Fender | jbalcomb : yes, only $10 cheaper, but includes PoE, lighted indicators. |
16:31.13 | *** join/#asterisk StevenL (n=steve@216.62.85.65) |
16:31.22 | _Thor | Hello everyone |
16:31.43 | jbalcomb | [TK]D-Fender That does seem like a good deal then. |
16:31.57 | masonf | how come there is no cdr_mysql in svn? |
16:32.21 | jbalcomb | masonf: thats covered on the wiki i believe |
16:32.30 | jbalcomb | masonf: licensing. |
16:33.47 | ManxPower | masonf, Digium has a binary only Asterisk product. Digium does not want two codebases and so cannot include MySQL stuff in Asterisk |
16:35.37 | masonf | makes sense.. thanks |
16:36.21 | Nugget | postgresql! :) |
16:36.31 | jbalcomb | Is the AGI for use inside asterisk and the API for use outside asterisk? |
16:36.37 | Nugget | better solution and free-er. |
16:37.13 | jbalcomb | Nah, I like L.A.M.P. rather than L.A.P.P. |
16:38.06 | sparkleytone | SAOP |
16:38.08 | sparkleytone | ftw |
16:38.14 | Nugget | I was just pointing out that using postgresql would remedy digium's licensing challenges with mysql and provide for a more robust solution. |
16:39.28 | masonf | voip-info says to get cdr_mysql from cvs.. sorry I have to ask another RTFM but where can I can cdr_mysql |
16:39.45 | wunderkin | addons? |
16:39.56 | ManxPower | masonf, all mysql stuff is in asterisk-addons |
16:40.10 | *** join/#asterisk copantl (n=galel@190.4.22.82) |
16:40.14 | jbalcomb | Is the AGI for use inside asterisk and the API for use outside asterisk? |
16:40.14 | copantl | hi guys |
16:40.34 | ManxPower | jbalcomb, You mean AMI not API |
16:40.42 | Katty | Nugget: prepare yourself. you're about to be hugged. |
16:40.47 | copantl | can asterisk carrier a dedicated data channel? |
16:41.02 | jbalcomb | ManxPower AMI is Asterisk Management Interface? |
16:41.03 | ManxPower | copantl, In theory. Define "dedicated data channel" |
16:41.09 | ManxPower | jbalcomb, correct |
16:41.13 | Nugget | hooray |
16:41.36 | copantl | hi ManxPower, can i have lised lines for data? |
16:41.37 | Katty | and now i'm gone (= |
16:41.41 | jbalcomb | ManxPower ok, gotcha. So then what ths diff. between the AGI and the AMI? |
16:41.53 | jbalcomb | ManxPower As far as what to use them for really |
16:41.56 | ManxPower | lised? |
16:42.26 | jbalcomb | copantl: there is some function/system out there for /sharing/ |
16:42.36 | ManxPower | jbalcomb, exactly what you said. AGI is designed for use inside the dialplan. AMI is designed for applications outside of Asterisk or the dialplan to control Asterisk |
16:43.02 | ManxPower | copantl, Zaptel allows you to DACS channel(s) to other channel(s) |
16:43.09 | jbalcomb | ManxPower: Very good, thank you. |
16:43.11 | copantl | ManxPower: if i have a multiplexer connected via e1 to my asterisk, can i dedicate some channels for data and others for voice? |
16:43.45 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
16:43.50 | ManxPower | copantl, yes, assuming your "dedicated some channels" just means "a digital version of cross connecting 1 channel to another" |
16:43.56 | copantl | wich protocols can use?, frame relay, HDLC? |
16:44.08 | ManxPower | copantl, DACS is protocol independent |
16:44.12 | copantl | ok |
16:44.19 | ManxPower | for example: |
16:44.28 | ManxPower | Telco -> T-1 -> Asterisk -> Channel Bank |
16:44.36 | ManxPower | well... |
16:44.41 | ManxPower | Telco -> T-1 -> Asterisk -> T-1 -> Channel Bank |
16:45.17 | ManxPower | We have channels 5-9 on the telco T-1 patched directly to channels 5-9 on the channel bank. Asterisk does not even have to be loaded, just zapte |
16:45.17 | ManxPower | l |
16:45.36 | ManxPower | this assumes your server has at least 2 digium T-1 ports on it |
16:46.08 | copantl | can i do this?: telco ->E1 -> asterisk -> channelbank -> DATA client |
16:46.30 | ManxPower | copantl, yes, that is really no different from the setup I just described. |
16:46.49 | ManxPower | assuming your Asterisk has at least 2 E-1 interfaces |
16:46.58 | ManxPower | copantl, what KIND of data? |
16:47.04 | ManxPower | If you mean fax then say that. |
16:47.17 | copantl | frame relay or ip |
16:47.25 | Nugget | 110 baud acoustic modem for a tandy model 100. |
16:47.44 | ManxPower | copantl, for DACS it's just bits. Zaptel does not know those higher level protoco.s |
16:48.52 | copantl | ManxPower: is a clear channel? |
16:49.04 | ManxPower | copantl, yes. |
16:57.19 | florz | when using the "domain" setting in sip.conf, how does it interact with [named] sections having "host" settings when determinint which context to drop the call into and which settings to use?! |
16:57.28 | nothinman | exit |
16:57.32 | nothinman | sorry ;] |
16:58.05 | florz | erm, more clearly: |
16:58.18 | florz | when using the "domain" setting in sip.conf, how does it interact with [named] sections (potentially having "host" settings) when determining which context to drop the call into and which settings to use?! |
16:59.00 | *** join/#asterisk japerry (n=japerry@216.231.51.208) |
16:59.13 | rpm | whats the ibm clustering/redundancy project for asterisk called? |
16:59.19 | japerry | g'morning Cunningpike =) |
16:59.34 | japerry | any ideas pop in the mind on the nice 'vancouver commute'? |
17:00.00 | *** join/#asterisk gennaro (n=root@ppp-62-10-136-185.dialup.tiscali.it) |
17:00.04 | gennaro | hi |
17:00.38 | CunningPike | Hey japerry - not really :( |
17:00.39 | gennaro | someone can help me with _9xxxxxx pattern ? |
17:00.47 | Juggie | ? |
17:00.50 | *** join/#asterisk bernardovieira (n=bvieira@c911935d.static.bhz.virtua.com.br) |
17:00.50 | Juggie | what do you need help with |
17:00.55 | gennaro | in italy there are numer like this 0573416604 or |
17:01.17 | japerry | Juggie: something wrong with a polycom 601 dropping calls outgoing randomly |
17:01.18 | gennaro | number more long |
17:01.22 | gennaro | in the same city |
17:01.33 | gennaro | how can i do? |
17:01.44 | CunningPike | japerry: Did you get anything helpful from your debugs? |
17:01.50 | Juggie | give me an example of a short and long number |
17:02.00 | japerry | CunningPike: nope |
17:02.02 | gennaro | 0573416604 |
17:02.09 | gennaro | 05734166041 |
17:02.13 | japerry | Cunningpike: but then again, she hasn't reported it dropping yet |
17:02.18 | Juggie | oh hmmm |
17:02.23 | Juggie | those are both valid? |
17:02.39 | gennaro | 0573416604 |
17:02.44 | CunningPike | japerry: OK - you have 'pri debug span x' set, yes? |
17:02.51 | jbalcomb | rpm: perhaps its 'System i' |
17:02.54 | CunningPike | japerry: And a CLI session runnng? |
17:02.57 | gennaro | 057325371 |
17:03.10 | gennaro | and others... |
17:03.15 | mut | ugh, i've been working at this company almost 3 years, and i still don't make what i asked for wages when i started, i wish the job market was better |
17:03.18 | gennaro | in other city... |
17:03.19 | japerry | Cunningpike: yup, I just called the front desk, told her to call me first before calling out |
17:03.30 | CunningPike | japerry: Then get her to call you as soon as she gets a dropped call and scroll back in the PRI debug |
17:03.37 | CunningPike | japerry: Perfect! |
17:03.51 | CunningPike | japerry: It's tedious, but really the only way........ |
17:04.10 | gennaro | juggie do u know how can i do? |
17:04.10 | *** join/#asterisk speedwagon (n=Ariel@70.46.87.158) |
17:04.28 | japerry | heh |
17:04.32 | Juggie | gennaro, _0XXXXXXXXXX. perhaps |
17:04.38 | japerry | pri debug span 1 gives an error |
17:04.41 | Juggie | you'll have a delay while asterisk waits for more digits |
17:04.46 | japerry | CunningPike: not a pri |
17:04.46 | Juggie | but it should be ok. |
17:05.11 | *** part/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
17:05.30 | CunningPike | japerry: But I thought you had a span configured in your zaptel.conf, no? |
17:05.57 | *** join/#asterisk creativx (n=creative@229.80-202-110.nextgentel.com) |
17:05.57 | japerry | yup, but its not a pri |
17:06.03 | CrashHD | I have nat=yes in my sip.conf but when I do sip show peers the Nat for all the peers shows "N"...what is going on? |
17:06.09 | gennaro | oh so if i do so |
17:06.35 | japerry | Cunningpike: well its not 'technically' a PRI, its 4 regular lines over a T1 carrier |
17:06.46 | nortex | japerry, Is it a channelized T-1 |
17:06.54 | gennaro | with an analog phone i cant use SEND key and pbx wait for nothing and call dont start |
17:06.59 | CunningPike | japerry: Using E&M, right? |
17:07.10 | japerry | Yes. but we are only using 4 channels |
17:07.19 | japerry | so nortex, yes |
17:08.04 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
17:08.06 | gennaro | can i use # as send key? |
17:08.18 | CunningPike | japerry: I think your config might need tweaking then - your zaptel.conf looks like it's set up for a PRI..... |
17:08.34 | Juggie | gennaro, have you checked to see if * can read the send key? |
17:08.52 | japerry | CunningPike: if it was a PRI though wouldn't it have a B channel, etc? |
17:09.04 | gennaro | how |
17:09.17 | japerry | CunningPike: I believe e&m=9-12 in the /etc/zaptel.conf file is what makes it not a PRI |
17:09.30 | Juggie | i assume you are using a analog phone connected to an ata which is registered to asterisk? |
17:09.49 | CunningPike | japerry: Can you pastebin your zaptel and zapata.conf again? |
17:09.53 | gennaro | no directly with DIGIUM FXS |
17:09.55 | japerry | yah, one sec |
17:09.59 | CunningPike | japerry: Or resend the link |
17:10.05 | Juggie | oh ok, either way your phone has dialtone correct? |
17:10.12 | gennaro | yes |
17:10.13 | japerry | Zapata http://pastebin.ca/73561 |
17:10.23 | Juggie | ok, well just remove whatever extension logic you have and do only this |
17:10.35 | Juggie | _X.,1,Noop(${EXTEN}) |
17:10.43 | Juggie | then pick up the phone dial a few numbers and press send |
17:10.46 | japerry | zaptel: http://pastebin.ca/74075 |
17:10.47 | Juggie | see what asterisk sees |
17:11.00 | gennaro | Noop ?!? |
17:11.09 | Juggie | noop is just a way to echo debug output |
17:11.12 | Juggie | you'll see :) |
17:11.22 | Juggie | we are just testing, this isnt going to connect a call. |
17:11.22 | gennaro | i use DIAL(ZAP/4/EXTEN |
17:11.26 | Juggie | i know that |
17:11.35 | Juggie | but i want to see what is being sent from the phone |
17:11.52 | Juggie | so do '_X.,1,Noop(${EXTEN})' pick up the phone, dial something, press the send button |
17:11.53 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
17:11.56 | Juggie | and then lets see what happens |
17:11.58 | CunningPike | japerry: OK - taking a look |
17:12.05 | Juggie | so, exten=>_X.,1,Noop(${EXTEN}) |
17:12.09 | Juggie | thats all you should have. |
17:12.11 | gennaro | i put down internet... |
17:12.16 | gennaro | tanx |
17:12.29 | Juggie | hah. |
17:12.31 | Juggie | awesome |
17:12.45 | Juggie | i was trying to see what dtmf the send key sent. |
17:12.52 | Juggie | which i suspect is a,b,c,d |
17:12.55 | Juggie | one or the other |
17:13.17 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
17:15.14 | rene- | hello, is feasible to setup a callcenter using desktops with softphones using bluetooth headsets? is RF interference a problem if we are talking large quantities (e.g. 100) ? |
17:15.47 | [TK]D-Fender | rene- : Few clients will be able to pick up based on the eadsets |
17:15.50 | Juggie | ummm.... this is support for asterisk :) |
17:16.31 | creativx | softphones is the shit |
17:16.48 | rene- | nobody i know wants to buy ip phones for call centers |
17:17.35 | creativx | i have yet to see a non-shitty softphone |
17:17.40 | Juggie | soft phones with windows roaming profiles would be better. |
17:17.52 | Juggie | user logs in, they get all their settings on whatever machine they log into and register. |
17:18.36 | rene- | [TK]D-Fender: what do you mean? |
17:19.18 | creativx | Juggie: but what softphone |
17:19.32 | Juggie | xten is fine. |
17:21.25 | creativx | i found it to be a bitch |
17:21.37 | creativx | too much eyecandy |
17:22.38 | nortex | creativx, But did it work? |
17:22.57 | creativx | i dont remember ;) |
17:23.11 | creativx | i bought ip10s phones instead and forgot about it |
17:26.08 | Heimidal | is there a way to add a prefix to the caller id (using SetCIDNum) for all inbound calls from numbers outside our office? |
17:26.13 | Qwell[] | sure |
17:26.23 | *** join/#asterisk jgoo (n=isec@ppp129-190.adsl.forthnet.gr) |
17:27.13 | CunningPike | japerry: Have you played around with rxwink and similar settings? |
17:27.21 | creativx | theres no way to answer a channel via the asterisk manager interface, is there? |
17:27.23 | MACscr | Qwell: is it possible to rewrite the Name in the caller id, but keep the number the same? |
17:27.27 | Qwell[] | MACscr: sure |
17:27.42 | CunningPike | MACscr: Set(CallerID(name)) |
17:28.45 | jgoo | CunningPike, as I newb, i am just curious, where would you put that Set(callerID(name)) ?? |
17:28.53 | Qwell[] | jgoo: dialplan |
17:29.03 | jgoo | really? oh ok, thought as much |
17:29.11 | jgoo | that is extensions.conf? or another file? |
17:29.14 | Qwell[] | yes |
17:29.25 | MACscr | The reason i ask is that im going to have a line for each company and would liek to have the company name appear when they call, but still the phone number of the client |
17:29.29 | jgoo | ok cool thanks |
17:29.31 | MACscr | er caller |
17:29.58 | jgoo | MACscr, PM me more info what you are doing, I find that interesting |
17:30.09 | jgoo | MAC, will this be DB driven_ |
17:30.10 | jgoo | ? |
17:30.13 | *** join/#asterisk caloi (n=caloi@nat-66-218-1-201.usadatanet.com) |
17:30.32 | MACscr | right now im only going two companies, so it wont be that fancy |
17:30.35 | smackus | quick question, what is the command for the cli to show what sip devices are configured? sip show channels is only for devices that are connected, right? |
17:30.39 | MACscr | but i was planning on having it interfacing with a crm software |
17:30.47 | MACscr | such as SugarCRM |
17:31.03 | smackus | I am troubleshooting permissions issues on the box again. |
17:31.03 | MACscr | im pretty sure Trixbox already does this |
17:31.13 | caloi | smackus - sip show (tab tab) will give you all the sip show options |
17:32.10 | salviadud | smackus, sip show peers |
17:32.23 | CunningPike | jgoo: In your dialplan: exten => 1234,n,Set(CallerID(name)=foo) |
17:32.38 | salviadud | smackus, sip show channels is for active sip channels |
17:32.51 | japerry | CunningPike: a litte, could that be causing the drop though? |
17:33.26 | CunningPike | japerry: Not sure - but what if asterisk was somehow detecting a hangup when there wasn't one? |
17:33.52 | *** join/#asterisk goldsmurf (n=rgoldber@64-13-22-231.dul.clearwire-dns.net) |
17:33.52 | MACscr | Cunning: could that be set per incoming DID instead of per ext? |
17:34.12 | CunningPike | MACscr: Could what be set? |
17:34.32 | MACscr | the callerID(name) |
17:34.55 | Heimidal | Qwell[]: how can I achieve the alteration of just caller id from numbers outside our office? |
17:34.55 | japerry | Cunningpike: hmm okay, I'll fiddle with it and see what outcoes |
17:35.27 | MACscr | Heimdal, thats what their talking about |
17:35.28 | japerry | CunningPike: so the other 'issue' is callerid which you probably see is statically set |
17:35.29 | CunningPike | japerry: OK - it'd be really nice to get some debug info....... |
17:35.30 | Qwell[] | Heimidal: see above |
17:35.52 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-145.dynamic.qsc.de) |
17:35.57 | CunningPike | japerry: Outgoing, right? |
17:36.34 | jgoo | CunningPike, just roughly how tricky is it to get the dialplan to interface with sugarcrm or another database_ I guess the dialplan can call scripts? |
17:36.47 | CunningPike | MACscr: I have a macro for incoming calls that sets the CID - what exactly are you trying to accomplish? |
17:36.47 | Heimidal | I guess I'm a bit confused.. how do you set it for just certain numbers (not based on the number that was called... the number the call is originating from) |
17:36.48 | jgoo | and evaluate expression inside it etc... |
17:36.58 | japerry | Cunningpike: heh Note: Caller ID can only be transmitted to the public phone network with supported hardware, such as a PRI. It is not possible to set external caller ID on analog lines. On supported systems, the phone company only receives the number, and supplies the name from their records. |
17:37.12 | *** join/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net) |
17:37.18 | japerry | japerry: so it seems that outgoing callerid has to be handled by verizon |
17:37.25 | CunningPike | japerry: Yes - are you getting incoming CID OK? |
17:37.31 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
17:37.39 | Persilon | How can I play a soundfile until some extension answers the call ? I've tried background, playback and musiconhold but none of them passes to n+1 while playing |
17:37.40 | CunningPike | japerry: The ability to set outgoing is determined by your telco |
17:37.45 | japerry | Cunningpike: negative :-( |
17:37.56 | MACscr | CunningPike: Basically im going to be running mutliple companies on asterisk, but the same staff and extensions answering the calls. I want the agents to be able answer the calls with the name of the correct company |
17:38.10 | MACscr | but i would like them to still be able to see the phone number that is calling |
17:38.14 | japerry | Cunningpike: asreceieved doesn't seem to work |
17:38.21 | CunningPike | japerry: Try adding a Wait(2) before you answer - CID takes 2 rings to appear |
17:38.22 | MACscr | especially with calls that are fwded to cell phones |
17:38.49 | Heimidal | :( |
17:39.18 | CunningPike | MACscr: So, in your incoming call context, use Set(CallerID(name)) based on the number called |
17:39.56 | CunningPike | MACscr: exten => whateveryourincomingnumberis,1,Set(CallerID(name)=foo) |
17:40.34 | CunningPike | japerry: Are you getting any kind of incoming CID at all? What are you seeing for incoming CID? |
17:41.20 | CunningPike | brb |
17:41.59 | japerry | kk |
17:43.30 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
17:44.17 | caloi | quick question (i hope) I've got an ingress SIP trunk that I route calls to my Asterisk box with, How can I route in the sip.conf/extensions.conf based on the called number. i.e. caller calls 3152222222 they go to A, they call 3153333333 they go to B. Both of these calls will come in the same SIP trunk from the same IP. I've tried the fromuser=X, but didn't have any luck.. |
17:44.21 | *** join/#asterisk goldsmurf (n=rgoldber@64-13-22-231.dul.clearwire-dns.net) |
17:50.37 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
17:51.29 | Hmmhesays | so whoes going to swing me some cash for cluecon |
17:52.08 | mogorman | hmm akward silence.... i guess thats a no |
17:52.30 | symlink | if you'll accept 1000% daily interest starting on the nanosecond it's loaned to you |
17:52.31 | symlink | sure |
17:53.29 | Katty | Hmmhesays: you can nap in my room if ya want ;) |
17:53.45 | Katty | Hmmhesays: if you bring any girls back, tho, you have to share. |
17:53.54 | symlink | !!! |
17:54.00 | mut | ewww katty is a cannibal? |
17:54.10 | Hmmhesays | lol |
17:54.22 | Hmmhesays | Fried or Baked? |
17:54.31 | Katty | Hmmhesays: annnd, you have to teach me how to jump in billiards. |
17:54.38 | mut | and she does drugs?! |
17:54.40 | Hmmhesays | symlink: do you know what a vagina is? |
17:54.42 | mut | NO WONDER! |
17:54.47 | Hmmhesays | because with a comment like that surely you've never seen one |
17:54.53 | Katty | Hmmhesays: i prefer baked...less fat haha |
17:55.05 | Katty | Hmmhesays: i appologize, that was low. |
17:55.07 | symlink | Hmmhesays: I was surprised with Katty's response |
17:55.30 | Katty | symlink: yeah well you don't know me that well either ;) |
17:55.48 | symlink | Katty != Katty! |
17:56.00 | japerry | Cunningpike: when I put up callerid=asrecieved it just says 'asterisk' |
17:56.50 | *** join/#asterisk bkw__ (n=brian@asterisk/friend-and-developer/bkw) |
17:57.21 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
17:57.27 | *** join/#asterisk catch23_ (n=catch23@hosta.sixcontinentshotels.com) |
17:57.56 | rene- | Katty's cute, cant say the same for the rubyonrails chick tho |
17:58.04 | catch23_ | anyone here ever run servers with non-ecc memory? I'm just wondering how likely single-bit errors would cause the kernel to crash... i've never experienced it before |
17:58.22 | MACscr | Katty? |
17:58.29 | catch23_ | I just figured there would be more server administrators in #asterisk than #ubuntu... |
17:59.09 | MACscr | lol, im blind, sry |
17:59.10 | MACscr | =P |
17:59.29 | Katty | MACscr: yes? |
17:59.36 | Katty | iDunno: you didn't hug it enough )= |
17:59.43 | Katty | Hmmhesays: you thinking about coming to cluecon? |
17:59.53 | Hmmhesays | PossiblI |
17:59.58 | MACscr | katty: someone was saying you were hot in comparison to the RoR chic |
18:00.00 | Katty | Hmmhesays: i could use an escort downtown. |
18:00.11 | MACscr | didnt see your name and was trying to figure out who they were talking about |
18:00.14 | Katty | Hmmhesays: can't have any of those weird guys mumbling to themselves while i walk by ;) |
18:00.38 | Hmmhesays | lol |
18:00.43 | Katty | MACscr: i'm kinda surprised rene- has seen a picture of me, really. |
18:00.43 | Hmmhesays | except me? |
18:00.49 | Katty | MACscr: i dunno who rene- is |
18:00.53 | Katty | Hmmhesays: yes dear, i know you're insane. |
18:01.02 | Katty | Hmmhesays: i've come to expect mumbling from you! |
18:01.04 | Hmmhesays | LOL |
18:01.18 | MACscr | lol |
18:01.57 | MACscr | lol, i didnt even know about cluecon |
18:02.09 | MACscr | its only about 2 and half hours from me |
18:02.42 | Katty | it's about 8 from me. |
18:02.46 | Hmmhesays | hop on your vespa and come on down |
18:02.49 | Katty | i'm sure someone will meet me at the amtrak station tho |
18:02.52 | MACscr | but im also not in the telecom arena anymore really |
18:03.05 | Katty | surely bkw will, if no one else. |
18:03.11 | MACscr | Katty, are you north? |
18:03.18 | rene- | Katty: you were @ astricon anaheim right? |
18:03.20 | Katty | MACscr: south west. |
18:03.23 | japerry | CunningPike: BINGO |
18:03.24 | Katty | rene-: no, no i wasn't. |
18:03.28 | japerry | Cunngingpike: Incoming call: Got SIP response 500 "Internal Server Error" back from 10.0.6.100 |
18:03.28 | japerry | <PROTECTED> |
18:03.36 | MACscr | ah, ok. Im in Peoria, IL |
18:03.37 | Katty | MACscr: i'm right around st. louis |
18:03.53 | rene- | oh, i have lived a live of deceit |
18:04.01 | MACscr | 8 hours to chicago from st. louis? |
18:04.08 | Katty | MACscr: i'm 2 hours south of st. louis ;) |
18:04.19 | MACscr | lol, ok |
18:04.22 | rene- | well then you were right i havent seen you ever :) |
18:04.40 | Katty | rene-: (= |
18:04.44 | japerry | Cunningpike: I think when she makes an outgoing call and someone calls her, it drops the call she made |
18:04.48 | Katty | MACscr: you're roughly 5 hours from me. |
18:05.34 | Katty | rene-: rubyonrails girl? i'd be interested in seeing that. post gifs. |
18:05.53 | Hmmhesays | self esteem is one of the greatest offspring songs EVER |
18:06.05 | Katty | Hmmhesays: i don't know about that. |
18:06.10 | Katty | Hmmhesays: you better send it to me so i can compare |
18:07.38 | Hmmhesays | sent |
18:08.05 | Katty | excellent. |
18:08.13 | Katty | i'm currently playing the hell out of lullaby by the cure |
18:08.17 | Katty | ever heard of it? |
18:08.21 | *** join/#asterisk phigwork (n=phigan@71-209-152-225.phnx.qwest.net) |
18:08.34 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
18:08.39 | phigwork | hello hello. |
18:08.42 | anthm | umm yeah they made it when they were elderly |
18:08.58 | phigwork | is there a string name for an incoming callerid number? |
18:09.10 | Katty | anthm: the cure? |
18:09.18 | phigwork | i'm trying to set the text field of an incoming line to something, but I still want to pass the telephone number |
18:09.22 | anthm | yah =D |
18:09.33 | Katty | not surprising...but i just heard of them about 2 or 3 months ago |
18:09.52 | anthm | oh |
18:09.58 | anthm | there's more |
18:10.21 | Katty | yeah, i have 6 albums now |
18:10.26 | Hmmhesays | Yeah i've heard it |
18:10.30 | Katty | lovecats is a favorite too |
18:10.37 | Katty | Hmmhesays, did you send it to my gmail? |
18:10.53 | Hmmhesays | the lady!@#!@#!@gmail yeah |
18:10.54 | anthm | in between days, head on the door |
18:10.59 | phigwork | and whats the difference between SetCIDName and SetCallerId? |
18:11.19 | anthm | play for today |
18:11.26 | anthm | hanging garden |
18:11.37 | sparkleytone | what is/are the proprietary codec(s) that i'm not supposed to use again? |
18:11.50 | Katty | Hmmhesays: k, got it now. tis been taking forever of late. |
18:12.15 | *** part/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
18:15.45 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
18:16.13 | *** part/#asterisk catch23_ (n=catch23@hosta.sixcontinentshotels.com) |
18:16.57 | CunningPike | japerry: OK - the 'asterisk' means that your CID is blank - which you probably knew already. If your telco is adamant that they are sending it, try the Wait(2) thing |
18:18.04 | *** join/#asterisk bkervaski (n=bkervask@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
18:18.08 | CunningPike | japerry: OK - that 500 error gives us something - the only time I have seen it here is when our hints (buddies) stop working. If you consistently get a 500 error when the call drops we'll have to see why that is |
18:18.31 | CunningPike | caloi: What does your context for incoming calls look like now? |
18:18.41 | bkervaski | Hi all. What does it take to keep asterisk 1.2 from using mpg123? I don't have mpg123 specified anywhere in my musiconhold.conf, but it still runs. I want to use the built in player with my mp3 files. Any help would be greatly appreciated. |
18:19.04 | creativx | bkervaski: look at the sample musiconhold.conf file |
18:19.13 | japerry | Cunningpike: well I thought we were getting somewhere, but alss nope. the 500 server error is independant of hangups |
18:19.18 | bkervaski | That's what I'm using. There's nothing about mpg123 in there?? |
18:19.18 | CunningPike | bkervaski: Have you configured native MOH properly |
18:19.29 | japerry | Cunningpike: also, incoming calls don't necissarly hang up her phone |
18:19.33 | bkervaski | @CunningPike: Probably not. Do you have to specify it? |
18:19.36 | CunningPike | japerry: That's what I suspected. Drats |
18:19.56 | bkervaski | Ahh.. got it.. thanks all, ;) |
18:19.59 | CunningPike | bkervaski: Oh, yes - look up 'native MOH' on the wiki - there is good into there |
18:20.11 | CunningPike | ~nativemoh |
18:20.20 | japerry | Cunningpike we tried two cell phones, she called one, and one called her.. didn't hang up or anything. I've tweaked rxgain a little too, but dunno if that makes any difference |
18:20.48 | japerry | Cunningpike: and talking about rxgain, I've seen a suggestion to lower it to get callerid to work.. all other ways seem to not work |
18:21.27 | CunningPike | ~moh |
18:21.29 | jbot | moh is probably Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf |
18:21.29 | bkervaski | I like mpg123, but from time to time it spikes out the CPU and won't let up until you kill it... |
18:21.59 | CunningPike | bkervaski: With the advent of native MOH, there is no longer any good reason to use mpg123 |
18:22.33 | CunningPike | japerry: Worth a shot - and there was nothing else in the CLI near the last hangup? |
18:23.03 | CunningPike | japerry: Can you do a 'sip debug peer' on her phone and see if that shows anything....... |
18:23.57 | bkervaski | So basically, just change the default mode=???? to mode=files to enable native moh? |
18:25.00 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
18:25.00 | *** join/#asterisk andrebarbosa (n=andrebar@62.48.215.74) |
18:25.01 | bkervaski | Does the native MOH keep the place in the file so when it starts next time it seems like it's been playing all along like mpg123? |
18:25.16 | Qwell[] | bkervaski: no, it streams it non-stop |
18:25.25 | japerry | CunningPike: I'll keep it running. she isn't calling out atm |
18:25.30 | andrebarbosa | anyone know how i call a dialplan function from an agi script? |
18:25.35 | Qwell[] | even when there are no calls on hold |
18:25.45 | Zodiacal | anyone know my analog phone (fxs) can't receive multiple calls? i enabled call waiting by pressing *71 and it said call waiting activated. anything else i have to do? |
18:25.56 | Qwell[] | It's your fault, you know? |
18:25.58 | andrebarbosa | i can call aplications using exec() ;) |
18:26.00 | symlink | yup |
18:26.06 | Qwell[] | This is seriously just out of hand |
18:26.26 | Qwell[] | I'm pretty sure Edison has been screwing up on my bill... |
18:26.43 | CunningPike | japerry: OK - hopefully we'll see something |
18:26.44 | Qwell[] | It's incredibly difficult to get a $300+ electric bill in a friggen apt |
18:27.09 | Qwell[] | That's more than double what it was 4-5 months ago |
18:27.20 | Qwell[] | nearly triple, in fact |
18:27.27 | salviadud | if i want to make a script that creates .call files using bash, how do i interact with asterisk? suppose the output was an error cause the call could not be completed? |
18:27.29 | symlink | Qwell[]: I snuck in |
18:28.09 | salviadud | if i did it with perl, would that be better? |
18:28.26 | rob0 | file ... directory ... symlink ... I am seeing a pattern here |
18:28.31 | xheliox | What's the trick to getting 'a' in current context to work with cmd VoiceMail? When I dial *, it just ignores it and doesn't send it to a,1 in the current context. :) |
18:28.33 | Qwell[] | rob0: No you aren't |
18:28.41 | symlink | rob0: I am not the person you are looking for |
18:28.45 | Zodiacal | qwell whats a normal electric bill in a house with central air in 90+ weather? just got an A/C :P |
18:28.51 | rob0 | yes you are |
18:29.07 | Qwell[] | Zodiacal: I don't know...but my usage hasn't changed from before, when it was only like $150 or less |
18:29.13 | CunningPike | Got a funny story from yesterday - our tax department recently moved to Asterisk and they have a roundrobin queue with agent penalties. Worked for a couple of weeks, and then our Engineering department began complaining that they were getting a slew of tax line calls. We couldn't figure out what the heck was going on, until I finally saw it in the CLI - one of the tax agents (with a high penalty, thank goodness) had forwarded th |
18:29.35 | Qwell[] | Zodiacal: That's in CA though, where electricity is like $9/kwh |
18:29.51 | salviadud | forwarded what? |
18:30.42 | creativx | we only got 80% of that story CunningPike :o |
18:30.52 | CunningPike | xheliox: Try operator=yes in voicemail.conf |
18:31.12 | CunningPike | creativx: Oh - sorry. How far did you get? :) |
18:31.26 | Zodiacal | qwell yeah socal |
18:31.41 | creativx | CunningPike: up to "forwarded th" |
18:31.42 | symlink | "had forward t" |
18:31.42 | creativx | ;) |
18:31.50 | xheliox | CunningPike: Hmm, ok.. |
18:31.53 | salviadud | yeah, man, you wrote it |
18:31.56 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
18:32.09 | CunningPike | had forwarded their phone to a phone in Engineering.......... |
18:32.19 | symlink | people are smrt |
18:32.29 | salviadud | you denied forwarding after that? |
18:32.37 | CunningPike | salviadud: Bingo! :D |
18:32.44 | anthm | Katty: http://www.cluecon.com/mp3/ made from videos of Live 8 in Paris last Summer |
18:32.48 | salviadud | hehe, nice one |
18:32.52 | X-Gen | salviadud, have u had a look at the management API to setup calls ? |
18:33.03 | salviadud | nortex, i have not |
18:33.08 | salviadud | i mean |
18:33.11 | CunningPike | salviadud: The Polycoms allow you to deny forwarding on a per registration basis which is great |
18:33.14 | salviadud | X-Gen, no i have not |
18:33.33 | X-Gen | salviadud, http://www.voip-info.org/wiki/view/Asterisk+manager+experience |
18:33.49 | salviadud | CunningPike, so you tweaked the phone instead of the box? |
18:33.55 | CunningPike | salviadud: Yes |
18:34.06 | salviadud | X-Gen, thanx man |
18:34.13 | CunningPike | salviadud: We use the UA forwarding, not the asterisk one |
18:34.37 | CunningPike | salviadud: Just happened to see the SIP 302 as it flashed past in the CLI |
18:34.56 | creativx | salviadud: i have |
18:34.56 | *** join/#asterisk NeonLevel (i=HydraIRC@200.52.142.184) |
18:36.30 | salviadud | creativx, would you say the call manager api is flexible? |
18:36.37 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) |
18:37.01 | creativx | salviadud: to some extent. how many clients are you planning to hammer the ami? |
18:37.08 | X-Gen | salviadud, it looks like its not suitable for multiple clients talking to it |
18:37.18 | creativx | thats why we use astmanproxy |
18:37.21 | *** join/#asterisk pnlarsson (n=niklas@c83-248-2-120.bredband.comhem.se) |
18:37.24 | creativx | then you can beat the shit out of it =) |
18:37.25 | japerry | CunningPike: okay got the log |
18:37.25 | X-Gen | u should write an in-betweener |
18:37.36 | X-Gen | ~astmanproxy |
18:37.40 | CunningPike | japerry: Great - can you pb it? |
18:37.40 | japerry | doesn't seem to show much to me, but getting it in pastebin now... |
18:37.47 | CunningPike | japerry: Great |
18:38.11 | X-Gen | ~waves |
18:38.13 | vader-- | hmmm |
18:38.25 | salviadud | hehe, it's more like this. i place a call to a number, then, i wait for it to get completed, after that, there should be a cycle that does +1 and i call the next number |
18:38.25 | creativx | salviadud: i use the ami for originating calls, CID and hangup |
18:38.32 | vader-- | i installed ntpupdate on my debian box and there is /etc/init.d/ntpupdate |
18:38.39 | vader-- | but for some reason it never updates automatically |
18:38.47 | vader-- | and the server's time is always off |
18:38.56 | japerry | Cunningpike: http://pastebin.ca/74149 |
18:38.56 | vader-- | i run ntpupdate and all is fine |
18:39.17 | X-Gen | creativx, how about a CSTA intergace ontop of that ? |
18:39.19 | salviadud | so i'm wondering if i should pull out some duct tape and perl it. or just use the manager api |
18:39.30 | creativx | X-Gen: yet another acronym.. whats csta J |
18:39.52 | creativx | salviadud: originate, then wait for hangup event, wait -> place new call, repeat? |
18:40.02 | salviadud | riiight! |
18:40.25 | salviadud | it's for a political campaign. haha |
18:40.30 | creativx | hehe |
18:40.32 | creativx | vicidial? |
18:40.53 | salviadud | i'm gonna dial a lot of freakin numbers |
18:40.56 | creativx | theres already some campaign-dialers out there |
18:41.12 | salviadud | yeah, i've heard |
18:41.21 | creativx | where you can have 50 agents just attending outbound calls |
18:41.21 | X-Gen | what about fax/answere machine detection ? |
18:41.22 | salviadud | and since it's not hard to do |
18:41.34 | creativx | nothing is hard with asterisk =) |
18:41.38 | creativx | you just gotta know how.. hehe |
18:42.05 | salviadud | fax/answering machine... well, thats a big sorry for the guy with the money |
18:42.54 | salviadud | and the funny thing is. most automated calls get hung up |
18:43.12 | creativx | well people in general hate talking to random people, dont they |
18:43.25 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
18:43.26 | salviadud | i like prank calling, but, that's something else |
18:43.30 | creativx | hehe |
18:43.40 | salviadud | i like calling people with a british accent |
18:43.47 | creativx | but are you planning to let a live person talk to some random number? |
18:43.53 | creativx | or play them some automated jumbomumbo |
18:44.14 | creativx | i dont know how these dial campaigns works.. except those who are out to ask questions/statistic analyzis |
18:44.18 | salviadud | just play automated recordings |
18:44.27 | salviadud | i can't even get dtmf signals to work |
18:44.32 | *** join/#asterisk jilott (i=jordan@24.79.192.187) |
18:44.33 | salviadud | damn voipdiscount! |
18:44.51 | creativx | hehe |
18:44.57 | salviadud | it's easy, you get a range of numbers |
18:45.01 | salviadud | you call 'em |
18:45.09 | salviadud | and hopefully they will interact with the IVR |
18:45.20 | salviadud | i can't get an ivr, cause i'm using SIP and not Zap |
18:45.33 | creativx | um |
18:45.36 | creativx | no |
18:45.40 | salviadud | my provider is not the kind of provider i can ask for assistance on that |
18:45.43 | CunningPike | japerry: I see a couple of SIP CANCEL messages in there - I'm just going to see if they are normal for this call progression |
18:45.45 | creativx | you have problem with transferring dtmf? |
18:45.45 | salviadud | well, its ugly sip |
18:45.47 | X-Gen | u cant use IVR on sip, thats new ? |
18:45.58 | salviadud | you can |
18:46.03 | salviadud | but, not with voipdiscount |
18:46.10 | *** join/#asterisk NotJohnDavid (i=dave@c-68-47-199-178.hsd1.tn.comcast.net) |
18:46.14 | bkervaski | Will "show channels" from the * cli show *ALL* channels? |
18:46.17 | salviadud | i don't wanna confuse anybody here... dtmf signalling should work on any codec |
18:46.17 | NotJohnDavid | anyone use an sipura hardware? |
18:46.28 | salviadud | i use sipura |
18:46.29 | creativx | then well if you need dtmf/ivr i would change service providers ;) |
18:46.31 | salviadud | sip 3000 |
18:46.52 | salviadud | yeah. well, that does not go along with my "suppa cheap mexican prices" |
18:47.02 | NotJohnDavid | salviadud: that's what I have. trying to make it ring thru from PSTN->FXS |
18:47.11 | NotJohnDavid | (without using asterisk) |
18:47.13 | jilott | good afternoon all. I have a question, perhaps someone could lend some advice. I have an IAX2 connection to my voip provider, when I place calls, they are cyrstal clear, beautiful. When I call in on my DID it's very poor quality, and there is a lot of static. Does anybody have any suggestions, or thoughts about this? |
18:47.15 | creativx | like selling you an internet connection but not letting you use the smtp protocol |
18:47.45 | salviadud | notjason, you gotta abilitate the pstn ring through line 1 |
18:47.50 | salviadud | oops |
18:47.51 | NotJohnDavid | salviadud: thought that if I put "*" in as ring1 caller under ring-thru distinctie caller it'd work but apparently not |
18:48.27 | NotJohnDavid | PSTN Ring Thru Line 1: is set to YES |
18:48.38 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
18:48.46 | salviadud | NotJohnDavid, how many rings before it rings it through? |
18:48.49 | japerry | CunningPike: thanks, yah I'm not sure if they are normal or not.. and I dunno how they'd be called |
18:48.56 | salviadud | there's an option for that too |
18:49.01 | NotJohnDavid | it doesn't ring through |
18:49.07 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
18:49.09 | CunningPike | japerry: Those CANCEL messages look suspect - I made a couple of calls and didn't get any CANCEL messages |
18:49.16 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
18:49.27 | salviadud | are both your channels registered? |
18:49.38 | justinu|laptop | cancel is what happens when you hang up before a call is answered |
18:49.46 | CunningPike | japerry: Call termination is indicated with a BYE message, not a CANCEL |
18:49.54 | salviadud | i've had some trouble with the spa 3000. if you don't have a channel registered, it kinda screws up |
18:50.05 | CunningPike | justinu|laptop: But japerry is getting them halfway through a call...... |
18:50.08 | salviadud | and if you do happen to have the sipura connected to asterisk |
18:50.19 | NotJohnDavid | oh I guess that makes sense that you'd have to register it. I don't want to make outbound calls on the PSTN, just accepting incoming |
18:50.28 | justinu|laptop | CunningPike: he's getting CANCELs after 200OK? |
18:50.46 | salviadud | NotJohnDavid, you can get the manual at their site |
18:50.50 | rpm | how do i convert gsm sounds to g729 compatible? |
18:50.51 | CunningPike | justinu|laptop: http://pastebin.ca/74149 |
18:51.03 | NotJohnDavid | salviadud: i don't have a proxy to register the pstn to? |
18:51.27 | salviadud | NotJohnDavid, if you don't register the pstn to something... it kinda freaks |
18:51.32 | CunningPike | japerry, justinu|laptop: There's a '487 Request Terminated' in there, too |
18:51.35 | NotJohnDavid | oh i think i may have found some settings *Tries* |
18:51.43 | salviadud | NotJohnDavid, is your line1 connected to asterisk? |
18:52.12 | NotJohnDavid | salviadud: no |
18:52.47 | *** join/#asterisk lars-ut-away (n=lars-ut@70.103.228.158) |
18:52.50 | NotJohnDavid | salviadud: all I want is for PSTN when ringing to pass through to the FXS on the spa3000 |
18:53.02 | justinu|laptop | CunningPike: i don't see that call ever being answered |
18:53.04 | NotJohnDavid | outbound VoIP and inbound VoIP works just fine |
18:53.06 | justinu|laptop | so the CANCEL looks valid |
18:53.50 | japerry | Cunningpike, justinu|laptop: but the call does go through.... hmm |
18:54.07 | justinu|laptop | japerry: for some reason, nothing is generating an answer... (200 OK) |
18:54.40 | CunningPike | justinu|laptop: He's using E&M - it looks like there is something missing in the supervision |
18:54.48 | salviadud | NotJohnDavid, this is what you can do, you tell the sipura to transfer all incoming calls from the pstn to an extension in your dialplan, which will then dial your FXS port |
18:55.08 | japerry | Cunning, so perhaps the wink settings in zapata aren't totally correct |
18:55.17 | NotJohnDavid | salviadud: can I do that without asterisk? |
18:55.29 | justinu|laptop | CunningPike: ok, in that case, the far end is either not setting AB bits to 1 on answer |
18:55.33 | X-Gen | creativx, are there any apps that run ontop of astmanproxy ? |
18:55.37 | justinu|laptop | or asterisk isn't seeing the AB bits go high. |
18:55.57 | justinu|laptop | but yes, the problem is that for some reason, answer supervision isn't working. |
18:56.21 | salviadud | NotJohnDavid, i'm trying to find it on voip-info, but... i can't, and i gotta go. yes, you can do it without asterisk. but for some weird reason it's not working for you... try registering the pstn line to asterisk, but... don't give it any permissions, i suspect that that's the problem |
18:56.26 | creativx | X-Gen: no |
18:56.39 | creativx | X-Gen: i run astmanproxy along with asterisk, and my clients connect to the astmanproxy and do their manager api there |
18:56.41 | *** join/#asterisk Zhadnost (n=tom@cpc1-sout6-0-0-cust691.sot3.cable.ntl.com) |
18:56.44 | salviadud | cya later guys! |
18:56.47 | CunningPike | japerry, justinu|laptop: Could be worth messing with wink settings......... |
18:57.02 | CunningPike | justinu|laptop: It's em_w, apparently.... |
18:57.26 | justinu|laptop | i've never seen e&m wink work right on asterisk |
18:57.29 | japerry | cunningpike, justinu|laptop: wink is 300, prewink is 20 |
18:57.34 | japerry | lol |
18:57.39 | justinu|laptop | some people say it does |
18:57.40 | CunningPike | japerry: This would probably explain the CID issues, also |
18:57.50 | *** join/#asterisk Zhadnost (n=tom@host-84-9-159-76.bulldogdsl.com) |
18:57.51 | CunningPike | japerry: Can your telco provide partial PRIs? |
18:58.02 | CunningPike | japerry: We have a 3-channel PRI here |
18:58.07 | japerry | Cunningpike: probably. I'm wondering if thats what we should do |
18:58.13 | justinu|laptop | ask your telco to switch to immediate, as a last resort |
18:58.15 | japerry | Cunningpike, how much might I ask does that run? |
18:58.21 | justinu|laptop | but go PRI if at all possible |
18:58.36 | CunningPike | japerry: It's quite reasonable - let me check |
18:59.30 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
18:59.32 | CunningPike | japerry: Well, hey, at least we got hints working! :D |
19:00.08 | japerry | Cunningpike: tis true! hehehe |
19:00.45 | japerry | the date/time is still off, but thats because ntp isn't working through the firewall-but I'm starting to think we should go with another provisioning plan |
19:01.58 | jilott | good afternoon all. I have a question, perhaps someone could lend some advice. I have an IAX2 connection to my voip provider, when I place calls, they are cyrstal clear, beautiful. When I call in on my DID it's very poor quality, and there is a lot of static. Does anybody have any suggestions, or thoughts about this? Maybe just suggest where to start looking? |
19:02.40 | creativx | jilott: codec conversion? |
19:02.43 | Zhadnost | are the incoming and outgoing calls using the same codec? |
19:02.56 | jilott | I assume so. Is there a way to tell? |
19:02.57 | justinu|laptop | the problem is likely on the ITSP side |
19:03.08 | creativx | sounds more like something i'd ask the service provider about |
19:03.12 | CunningPike | japerry: CAD$140/mo for 3B+D |
19:03.53 | jilott | cool, I've found that the problem exists for alaw,ulaw, and g729 |
19:04.00 | Zhadnost | look in your asterisk console (with verbosity set high) or iax2 show channels when in a call. |
19:04.13 | creativx | what codec is the calls delivered to you in? |
19:04.26 | Zhadnost | ? ? ? |
19:04.33 | jilott | thanks for that. I'm new to asterisk. I thought it was ULAW that it was delivered in. |
19:04.57 | CunningPike | japerry: For provisioning, we are using FTP. It's better than TFTP because a) you get logs b) the phone can upload exceptions and directories and c) the phones can detect config file changes |
19:05.29 | japerry | CunningPike: really for that price? damn. |
19:05.47 | CunningPike | japerry: High? |
19:06.08 | japerry | CunningPike: oh no, thats very nice |
19:06.19 | japerry | Cunningpike: we're paying something like $275/month-ish |
19:06.43 | CunningPike | japerry: Ah - you'll be much better off with a partial PRI then - and you can add extra channels really easily |
19:06.48 | Zhadnost | I have 3 POTS lines here, 2 from the same provider and 1 from a different provider, with the latter one, If I receive a fax call, I can pick it up using a modem, but not on the line card (TDM400) through iaxmodem, which works on all the other lines. |
19:07.01 | Zhadnost | does anyone know a good strategy to tweaking the settings for the line? |
19:07.11 | japerry | Cunningpike: I assume if you want, you can expand your channels as you grow? |
19:07.22 | *** join/#asterisk roche (n=roche@200.122.154.250) |
19:07.28 | CunningPike | japerry: Yes - we have a 13-channel in production that used to be a 3 |
19:07.47 | CunningPike | japerry: There's a max of 23 ;) |
19:07.57 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
19:08.18 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
19:09.20 | rpm | is it normal for festival to have a zombie process running along with the parent? |
19:10.52 | roche | Hello People, I am using ViciDial and we are having some strange problems with iax, we are paying an aix line with 50 channels and seems like all channels are activated and are not hang up |
19:13.12 | Hmmhesays | make[2]: Entering directory `/usr/src/hmm0/buildroot/build_mipsel/asterisk-1.0.9/channels' |
19:13.13 | Hmmhesays | ./gentone busy 480 620 |
19:13.13 | Hmmhesays | ./gentone: ./gentone: cannot execute binary file |
19:13.13 | Hmmhesays | make[2]: *** [busy.h] Error 126 |
19:13.16 | Hmmhesays | what.. the crap |
19:13.23 | *** join/#asterisk asterisknewbiezz (n=chatzill@rrcs-67-52-187-18.west.biz.rr.com) |
19:13.43 | Hmmhesays | why are we trying to execute gentone...hmmmmmmmmmm? |
19:13.46 | asterisknewbiezz | anyone know how to fax in asterisk? |
19:13.52 | Hmmhesays | alksdfja;sljkf |
19:13.53 | rob0 | hmm, I say! |
19:13.55 | Hmmhesays | Thank you that is all |
19:13.58 | Zhadnost | with hylafax? |
19:14.41 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
19:15.00 | asterisknewbiezz | does that work with voip? |
19:15.20 | Hmmhesays | anyone know what ./gentone is supposed to do? |
19:15.28 | Hmmhesays | cause it sure as hell is not going to work when i'm cross compiling |
19:15.39 | justinu|laptop | it makes sound files |
19:15.47 | funxion | is there anyway to run a command in the dial plan on a call that is currently in use |
19:15.49 | creativx | generate tone |
19:15.50 | creativx | hehe |
19:15.57 | *** join/#asterisk jorgeikeda (n=freebot@201.154.10.103) |
19:16.02 | rob0 | Gentone was the predecessor of Gentwo. |
19:16.07 | Hmmhesays | well duh |
19:16.21 | Hmmhesays | rob0 was that a serious remark? |
19:16.26 | justinu|laptop | Hmmhesays: why it's in the makefile, you'd have to ask digium |
19:16.35 | justinu|laptop | i'd just remove it |
19:16.41 | Hmmhesays | yeah i'm going to |
19:16.58 | *** join/#asterisk watchy (n=watchy@office2.gwhsi.com) |
19:17.02 | watchy | aynone here run an isp? |
19:17.07 | Zhadnost | asterisknewbiezz> hylafax? yeah. Look up iaxmodem on voip-info.org, it lets an iax channel act as a virtual modem. |
19:17.24 | jorgeikeda | hi |
19:17.44 | CunningPike | funxion: Say what? |
19:17.53 | funxion | lol |
19:17.54 | rob0 | I confess, it was a lousy attempt at a joke. |
19:18.01 | jorgeikeda | anyone knows how to make the chan_unicall.so work? I´m trying to make it work with asterisk-1.2.5 |
19:18.40 | funxion | once I place a dial command is there anyway to update a table in mysql before the call hangs up? |
19:19.44 | CunningPike | funxion: Not before the call hangs up I don't think so, unless you do some funky agi thing |
19:19.55 | funxion | tahts what I thought |
19:20.01 | Hmmhesays | if I'm cross compiling can I pull *.h files from another platform? |
19:20.07 | funxion | but was hoping someone in here might be able to say different |
19:20.20 | CunningPike | funxion: You can do stuff after hangup..... |
19:20.24 | funxion | I know |
19:20.27 | justinu|laptop | app_dial... asterisk's achilles heel |
19:20.47 | funxion | lol |
19:21.09 | Hmmhesays | yeah you need a really really big microscope |
19:21.11 | justinu|laptop | Hmmhesays: i don't see why not |
19:21.15 | watchy | no one here run an isp? |
19:21.26 | funxion | its like when osama bin laden got pantsed on south park |
19:21.38 | rpm | watchy: an isp or itsp? |
19:21.43 | funxion | if anyone has seen that episode |
19:21.50 | watchy | isp. im curious if i should subnet my clients |
19:22.01 | watchy | and i would imagine someone here actually runs an isp |
19:22.08 | justinu|laptop | bridging is lame |
19:23.17 | funxion | is there no way to run two commands on one priority of a context? |
19:23.26 | rpm | subnet your clients? you mean like give them a /30 subnet and over-populate your routing table? |
19:23.48 | watchy | rpm: yea, so i should just leave em on 255.255.255.0? |
19:24.03 | watchy | well 255.255.254.0 i mean |
19:24.12 | watchy | i got a /23 and a /24 |
19:24.50 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
19:25.34 | rpm | yes. just have network, most people have basic firewalls to prevent eachother from conencting to eachother |
19:26.03 | rpm | "just have one network" |
19:27.15 | watchy | well im having issues with broadcast issues i think |
19:27.28 | justinu|laptop | see... bridging sucks |
19:32.56 | mountainm2k | anybody know of a good Polycom distributor? |
19:33.22 | mountainm2k | I've just been told that Atacomm, where I've gotten a few phones to test with, is _not_ authorized |
19:33.24 | justinu|laptop | what do you need? rock bottom price? |
19:33.31 | justinu|laptop | atacomm also rapes you on shipping |
19:33.38 | mountainm2k | Well, that'd be good... |
19:33.54 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.94.154.226) |
19:33.58 | *** join/#asterisk mfedyk (n=mfedyk@adsl-63-194-240-129.dsl.lsan03.pacbell.net) |
19:33.59 | mountainm2k | I really don't care if they know how to support Asterisk, because I think I'm going to buy the Digium support |
19:34.12 | justinu|laptop | i bought about 25 phones from the voipconnection guys, and their price isn't the lowest, but they really came thru and worked with me on the price |
19:34.16 | mountainm2k | (company, you know -- some crap about mission critical, yadda) |
19:34.33 | mountainm2k | are they authorized distributor? |
19:35.12 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
19:35.12 | *** mode/#asterisk [+o denon] by ChanServ |
19:35.25 | justinu|laptop | i believe so |
19:37.35 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
19:38.20 | [TK]D-Fender | mountainm2k : Costs a bit more, but VoIPSupply |
19:38.49 | mountainm2k | <PROTECTED> |
19:38.52 | [TK]D-Fender | mountainm2k : Whats your troule with them? |
19:38.56 | [TK]D-Fender | mountainm2k : Apparently |
19:38.58 | justinu|laptop | i've been dicked by voipsupply too many times |
19:39.15 | justinu|laptop | saying they have shit in stock, when its not |
19:39.17 | TripleFFFF | weird.. a rep at voipsupply is anmed justin |
19:39.26 | justinu|laptop | sending me bad phones, taking over a month to replace them |
19:39.37 | TripleFFFF | justinu|laptop wich phones ? |
19:39.44 | justinu|laptop | cisco 7960 |
19:39.47 | TripleFFFF | anyone here have the pap2t nam admin pdf ? |
19:39.55 | TripleFFFF | voipcupply didnt send it.. lol |
19:39.58 | TripleFFFF | k JunK-Y |
19:40.02 | TripleFFFF | k justinu |
19:40.02 | justinu|laptop | they actually sent me a bad 7960 and a bad power brick |
19:40.15 | justinu|laptop | either way, the experience was unpleasant... |
19:40.15 | [TK]D-Fender | justinu : Taken me often nearly that long WITH my auth'd Polycom reseller |
19:40.15 | TripleFFFF | yeah on phone you need to consider 10% fail |
19:40.22 | mountainm2k | hah, Atacomm on the phone... |
19:40.42 | TripleFFFF | they are eorgansiing due to grotwh and seperating resel from end user |
19:40.55 | TripleFFFF | so they will be better.. myself got 30/60.. 30 pleasent 60 unpl |
19:40.57 | luke-jr_ | Any suggestions for a SellVoIP-like ITSP? |
19:41.07 | TripleFFFF | luke-jr_ whant u need ? |
19:41.10 | TripleFFFF | what you need ? |
19:41.15 | TripleFFFF | theclubvoip.com |
19:41.17 | TripleFFFF | ;) |
19:41.22 | justinu|laptop | so i threw voipconnection a bone, and they came thru... very good communication, prompt shipping, etc. |
19:41.34 | luke-jr_ | TripleFFFF: cheap DIDs and decent service? |
19:41.40 | TripleFFFF | ok how cheap dids |
19:41.41 | TripleFFFF | and where |
19:41.44 | TripleFFFF | location |
19:41.53 | TripleFFFF | where are orig/term going to be |
19:41.53 | luke-jr_ | SellVoIP's were $1/mo |
19:42.12 | justinu|laptop | so why not go with them? |
19:42.12 | TripleFFFF | 1$ a month for how much.. and. cheaper you pay less support/service you get you know.. |
19:42.20 | luke-jr_ | justinu: they don't really exist anymore |
19:42.21 | TripleFFFF | i can get them at 0.10 cent |
19:42.23 | justinu|laptop | oh |
19:42.29 | TripleFFFF | but i need 5000 |
19:42.36 | justinu|laptop | we pay 50c/mo for our DIDs |
19:42.44 | luke-jr_ | TripleFFFF: heh, I don't have that many users |
19:42.48 | TripleFFFF | thats the point |
19:42.56 | TripleFFFF | usa luke-jr_ ? |
19:42.59 | luke-jr_ | yeah |
19:43.04 | TripleFFFF | k |
19:43.56 | luke-jr_ | oh, and no flash UIs =p |
19:44.13 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
19:44.35 | luke-jr_ | (that was brought on by theclubvoip.com BTW) |
19:45.26 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
19:45.26 | *** mode/#asterisk [+o russellb] by ChanServ |
19:46.16 | *** join/#asterisk SparFux (n=player@e182028106.adsl.alicedsl.de) |
19:46.17 | Hmmhesays | bah why doesn't gcc give the full path it is looking for the headers in |
19:47.36 | SparFux | I use a bri line. When doing wait(9) before Answer, the line doesn't get picked up, but rings endlessly. With wait(5) it works, but I want to do pickup later on. What could be the reason? |
19:48.20 | X-Gen | someone reboot sourceforge please |
19:49.02 | TripleFFFF | lol |
19:50.08 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
19:50.09 | CunningPike | X-Gen: Doine |
19:50.13 | CunningPike | :) |
19:50.32 | *** join/#asterisk skac (i=ash@kakistocracy.co.uk) |
19:51.08 | CunningPike | :) |
19:51.50 | *** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net) |
19:53.01 | *** join/#asterisk southtel (n=slester@c-67-191-211-17.hsd1.ga.comcast.net) |
19:53.33 | southtel | Is there a cli command to kill a given channel? |
19:53.51 | TripleFFFF | soft hangup |
19:54.08 | southtel | Beautiful! Thanks. |
19:54.17 | skac | anyone got documents on how to set up vonage and asterisk? |
19:54.27 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
19:54.33 | skac | vonage is doing my head in, so i am going to go on to a new provider soon. |
19:54.38 | Hmmhesays | give me 50 bucks and i'll do it |
19:54.44 | skac | no thanks. |
19:54.53 | Hmmhesays | I use them all the time with no problems |
19:55.20 | skac | they are okey. |
19:55.36 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
19:55.44 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
19:55.48 | skac | Hmmhesays: got any documentation on how you did it? |
19:55.58 | justinu|laptop | they work like any other sip provider |
19:56.04 | skac | i see. |
19:56.08 | Hmmhesays | everything you need is on the wiki |
19:56.25 | justinu|laptop | afaik, they won't give out the sip credentials unless you buy a softphone account |
19:56.26 | *** join/#asterisk ATravelingGeek (n=atg@pool-162-83-107-154.culp.east.verizon.net) |
19:56.26 | Hmmhesays | i need a fscking cigarette |
19:56.34 | stack_ | I'm getting "PROGRESS with cause code 31 received" on the console. I looked up the code and it says, "normal, unspecified. This cause is used to report a normal even only when no other cause in the normal class applies"... what's that mean? |
19:56.36 | skac | justinu|laptop: damn |
19:56.42 | Hmmhesays | or a business account |
19:56.45 | skac | yeah |
19:56.48 | justinu|laptop | skac: if you threaten to cancel, they might change their tune |
19:57.01 | skac | *nod* |
19:57.10 | skac | are the details not guessable? |
19:57.13 | justinu|laptop | no |
19:57.17 | skac | okey. |
19:57.24 | justinu|laptop | well, yes |
19:57.33 | justinu|laptop | but it would take hundres of years, unless you're the NSA |
19:57.42 | skac | ah haha. |
19:57.57 | Hmmhesays | my password has an underscore in it |
19:58.07 | justinu|laptop | stack: progress means you've got inband tones available, generally |
19:58.07 | Hmmhesays | so good luck with that |
19:58.11 | CrashHD | what is a good way to have internal caller id be extensions on internal but to use a DID number for external calls (Have a pri hooked up) |
19:58.13 | CrashHD | ? |
19:58.26 | stack_ | justinu|laptop: and what does that mean? ;p |
19:58.38 | justinu|laptop | it means the network is telling you something on the voice channel |
19:58.48 | justinu|laptop | like you call can't be completed |
19:58.52 | smackus | ok, I am having trouble with one phone. When dialing it, it goes straight to voicemail. It can dial out, and it is not set to do not disturb. What could the issue be? |
19:59.06 | smackus | the exten => is identical to other working phones. |
19:59.08 | smackus | using macro |
19:59.11 | justinu|laptop | smackus: isounds like the phone isn't registered |
19:59.17 | skac | anyone know a good provider in Canada? |
19:59.25 | smackus | ok. |
19:59.30 | smackus | so sip.conf? |
19:59.31 | stack_ | justinu|laptop: ok, but the message for code 31 doesn't make sense to me... any idea? |
19:59.51 | justinu|laptop | stack: no, if you give me some context, i might be able to explain it better |
19:59.59 | justinu|laptop | smackus: sip show peers |
20:00.20 | CunningPike | CrashHD: Set(CallerID(num)=<insertDIDprefixhere${CallerID(num)}) |
20:00.52 | CunningPike | smackus: sip show peers |
20:00.53 | stack_ | justinu|laptop... someone dialed a number and I got that on the console... you get a fast busy when it makes the attempt |
20:01.13 | CunningPike | skac: Provider of what? |
20:01.19 | justinu|laptop | stack: they've probably called a number that's Non-ISDN equipment |
20:01.28 | skac | CunningPike: VoIP BPX? |
20:01.29 | CrashHD | CunningPike: so there is no general option I can set? I have to set it in the dialplan? |
20:01.39 | skac | i think.. i am very new to VoIP |
20:01.40 | justinu|laptop | stack: turn on pri debug and call it again |
20:01.41 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
20:01.55 | CunningPike | CrashHD: Yes |
20:02.04 | justinu|laptop | stack and skac |
20:02.05 | CrashHD | CunningPike: ok thank you |
20:02.09 | justinu|laptop | who's trying to confuse today? |
20:02.12 | CunningPike | skac: Where are you? |
20:02.22 | CunningPike | There is only one CunningPike |
20:02.30 | justinu|laptop | thankfully |
20:02.33 | justinu|laptop | :) |
20:03.08 | justinu|laptop | my wife made me watch independance day last night.... what a shitty movie |
20:04.08 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
20:04.13 | CunningPike | justinu|laptop: Hey! :P |
20:04.22 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
20:04.46 | CunningPike | biab - jog time |
20:05.10 | justinu|laptop | go file! |
20:05.29 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
20:06.07 | smackus | wouldnt sip show user show me if the phone was registered? |
20:06.55 | smackus | my bad... you guys already answered that one |
20:06.59 | smackus | I missed it |
20:08.22 | smackus | ok, so if my extension I am having issues shows up in sip show peers, is it then registered? |
20:08.42 | SparFux | How come my call doesn't get picked up anymore when waiting more than Wait(7) ??? |
20:09.22 | *** join/#asterisk Wazb^ (n=wazb@199.243.74.220) |
20:09.26 | Wazb^ | Hi all |
20:09.46 | svenbart | anyone tried sipX? |
20:10.07 | Wazb^ | any idea from where i can get G729 codec (non-commercial) |
20:10.49 | justinu|laptop | smackus: if it shows an IP address, yes |
20:11.30 | smackus | yes it does... what else could be the issue? |
20:12.02 | justinu|laptop | is qualify on? |
20:12.11 | smackus | it plays his voicemail recordings correctly. you can leave them. he can log in and listen to them. |
20:12.22 | smackus | justinu|laptop: you asking me? |
20:12.25 | justinu|laptop | yes |
20:12.34 | smackus | where is that located? |
20:12.40 | justinu|laptop | sip.conf |
20:12.50 | smackus | I dont have that on any of my phones. |
20:12.55 | justinu|laptop | if it's on, you should see "OK (52ms)" in the status column |
20:13.00 | smackus | even the ones that are working correctly |
20:13.05 | smackus | should i add it? |
20:13.10 | smackus | qualify=on? |
20:14.02 | justinu|laptop | give it a shot |
20:14.15 | vader-- | can an fxs channel be used as a fxo channel? |
20:14.31 | justinu|laptop | no |
20:15.02 | smackus | did not change it, still says unmonitored. |
20:15.14 | Wazb^ | any idea from where i can get G729 codec (non-commercial) ? |
20:15.14 | justinu|laptop | sip reload? |
20:15.23 | justinu|laptop | Wazb^: there's no such thing |
20:15.26 | justinu|laptop | you hae to buy it from digium |
20:16.00 | smackus | yes |
20:16.19 | justinu|laptop | smackus, something's not right then... you must have added the qualify in the wrong place |
20:16.27 | smackus | could be, where should it be? |
20:16.37 | justinu|laptop | in the section for that phone |
20:16.57 | smackus | yeah, i did qualify=on |
20:17.19 | *** part/#asterisk Wazb^ (n=wazb@199.243.74.220) |
20:18.20 | *** join/#asterisk Johnnie (i=odysseus@pdpc/supporter/active/Johnnie) |
20:19.39 | vader-- | anyone know what this means: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
20:19.43 | *** join/#asterisk DrkShdw (n=DrkShdw@fl-209-26-20-205.sta.embarqhsd.net) |
20:19.46 | vader-- | and should i worry about it? |
20:19.57 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
20:19.58 | X-Gen | vader-- yup, be afraid |
20:20.02 | vader-- | really? |
20:20.18 | X-Gen | it means the PC cant handle the IRQ fast enough |
20:20.38 | X-Gen | make sure card is not sharing an irq with something else |
20:20.39 | vader-- | hmmm |
20:20.39 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
20:21.43 | CrashHD | music on hold seems to work fine but music on call park does not work...which config am I missing? |
20:22.35 | *** part/#asterisk mfedyk (n=mfedyk@adsl-63-194-240-129.dsl.lsan03.pacbell.net) |
20:22.58 | vader-- | whats the linux command to see what irq are being used by what cards? |
20:23.07 | DrkShdw | lspci |
20:23.12 | justinu|laptop | cat /proc/interrupts |
20:23.14 | X-Gen | cat /proc/interrupts |
20:23.18 | X-Gen | *echo* |
20:23.20 | X-Gen | :P |
20:23.41 | X-Gen | vader--, make sure your disks are using DMA |
20:23.59 | vader-- | <PROTECTED> |
20:24.00 | vader-- | <PROTECTED> |
20:24.50 | X-Gen | anyone else care to comment ? |
20:24.59 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
20:26.43 | justinu|laptop | you have a T1 card, and a tdm2400 card in there? |
20:26.56 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-145.dynamic.qsc.de) |
20:28.09 | _alex_mx_ | vader--, 4 procs, 2 dual cores, or 2 with HT? |
20:30.06 | vader-- | ya |
20:30.13 | vader-- | dual core |
20:30.19 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
20:30.19 | vader-- | i believe |
20:30.53 | vader-- | justinu i am providing analog for a couple lines |
20:30.56 | vader-- | fax machines and stuff |
20:31.12 | _alex_mx_ | vader--, are you dropping calls? How often do you see the message? |
20:34.37 | websae | anyone going to ClueCon in here...? |
20:34.42 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
20:39.06 | *** join/#asterisk bmg505 (n=leon@c1-111-6.rndf.isadsl.co.za) |
20:39.23 | caloi | anyone feel like helping me through configuring ingress SIP channels? |
20:40.09 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
20:40.32 | Dr-Linux | hi all |
20:41.29 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
20:42.10 | Dr-Linux | anybody knows about Cisco Voip conference phone? |
20:42.39 | iq | hi Dr-Linux |
20:42.50 | *** join/#asterisk bkw__ (n=brian@asterisk/friend-and-developer/bkw) |
20:42.54 | Dr-Linux | iq: ehh! how are you dude? |
20:43.02 | iq | Dr-Linux: me good - how r u |
20:43.59 | Dr-Linux | iq: i'm ok, today we recieved a Cisco VOIP conference phone. i didn't see it yet, but tomorrow i will have to configure it |
20:44.25 | Dr-Linux | so i was just about to ask it's little info maybe someone know about it |
20:45.19 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
20:47.01 | *** join/#asterisk clive- (n=pirch@dsl-145-33-168.telkomadsl.co.za) |
20:48.50 | iq | Dr-Linux: model number? |
20:49.27 | Dr-Linux | iq: that i don't know yet sorry :) |
20:49.29 | iq | Dr-Linux: I don't think it will be that different than a regular VoIP phone. These phones do have some nice Speaker system, etc. |
20:49.47 | iq | Dr-Linux: array - koi baat nahi yaar. kal tak wait ker lo |
20:50.24 | Dr-Linux | iq: i just want to confirm that if this cisco conference needs same SIP firmwares as already i'm using for our couple of cisco 7960's |
20:50.44 | *** part/#asterisk _alex_mx_ (n=_alex_mx@200.94.154.226) |
20:51.20 | Dr-Linux | iq: bas as you know bro , mojhey iss filed say shok hai, phone kal daikhna hai, excited abhi say hoon, |
20:51.21 | Dr-Linux | :) |
20:51.39 | iq | Dr-Linux: sorry - I've no idea :( ...it will be hard to tell without the model number. My experience is that these things work out of the box - then you can upgrade firmware of available |
20:52.01 | iq | Dr-Linux: hota hai yaar... sab ke saath hota hai :) |
20:52.16 | Damin | Dr-Linux: WTF? |
20:52.37 | *** part/#asterisk Vorondil (n=cwaldeck@mail.yhamerica.com) |
20:52.40 | Dr-Linux | Damin: STFU |
20:52.49 | Dr-Linux | iq: hehe yeah |
20:52.56 | Dr-Linux | Damin: sorry! |
20:54.09 | iq | Dr-Linux: do you know anyone who gives DID in Lahore? |
20:54.31 | Dr-Linux | iq: US DID? |
20:54.43 | Dr-Linux | iq: i don't think if there is any |
20:54.43 | iq | Dr-Linux: no |
20:54.51 | Dr-Linux | iq: paki? |
20:54.59 | iq | Dr-Linux: Lahore |
20:55.08 | brijn | India |
20:55.27 | iq | brijn: do you know anyone offers DID in India? |
20:56.00 | brijn | Nope, sorry |
20:56.06 | brijn | Google is your friend |
20:56.15 | iq | brijn: I like Yahoo more |
21:00.41 | clive- | this cvs thing is confusing me:)...just checked-out a shit load of junk .:) |
21:01.16 | CrashHD | how can I get music on hold to play while the call is parked? |
21:01.20 | clive- | I mean svn |
21:01.47 | nortex | clive-, What did you try and chckout? |
21:02.15 | *** join/#asterisk findlay (n=justin@67.137.24.114) |
21:03.20 | *** join/#asterisk okdo (n=goldenol@65.171.196.18) |
21:03.22 | okdo | hi |
21:03.30 | findlay | hello |
21:03.31 | *** join/#asterisk skraelings001 (n=skraelin@190.40.39.154) |
21:03.43 | clive- | nortex, I did a "svn checkout http://svn.digium.com/svn/astcc |
21:03.44 | clive- | " , I guess I should have named the "trunk" as well |
21:03.45 | okdo | anyone have any recommendations for T1 cards when splitting voice and data channels? |
21:03.52 | okdo | like use the sangoma instead of the digium, etc. |
21:04.47 | skraelings001 | hi folks |
21:05.43 | skraelings001 | does direct pickup application wok with any chan? |
21:05.49 | symlink | yes |
21:06.37 | skraelings001 | it does work with sip and iax channels only for me |
21:07.03 | symlink | if you're doing by extension, a CDR record has to exist... if you're doing via variable, then the variable has to exist |
21:07.29 | *** part/#asterisk nortex (n=nortex@64.136.65.142) |
21:07.44 | skraelings001 | symlink: can it be a mysql cdr? |
21:08.24 | symlink | a CDR has to exist on the channel |
21:11.41 | *** join/#asterisk tech9iner (n=hacim@unaffiliated/tech9iner) |
21:11.56 | tech9iner | g'day all.. |
21:12.18 | tech9iner | can ye mates clarify something i seem ill equipped to discern on me own please?.. |
21:12.49 | skraelings001 | symlink: ok, i'll check this out. thanks |
21:13.31 | tech9iner | is asterisk overkill for a pc answering machine app on suse 10.0 using simple old ''' 02:0a.0 Communication controller: Agere Systems 56k WinModem (rev 01) ''' ?.. |
21:14.03 | jbalcomb | tech9iner: Asterisk@home is prolly a simpler choice |
21:14.06 | tech9iner | googled n read n read every bit o documentation i can find seeking this answer.. |
21:14.46 | tech9iner | ahhh.. sounds like it jbalcomb !! never heard of it.. lol.. prolly RIGHT on *'s homepage huh.. lmao.. |
21:15.01 | jbalcomb | How can I use the AMI to get the IPs and Extensions of all my phones? |
21:15.03 | tech9iner | oi.. tx mate.. ill dig that up.. |
21:15.09 | jbalcomb | g'luck |
21:15.16 | tech9iner | tx mate.. |
21:15.26 | jbalcomb | good on ya mate |
21:17.07 | tech9iner | lmao.. handup for ye help jbalcomb lol http://www.koalanet.com.au/australian-slang.html wink.wink............ |
21:17.19 | tech9iner | cheerio then.. |
21:17.20 | *** part/#asterisk tech9iner (n=hacim@unaffiliated/tech9iner) |
21:18.31 | jbalcomb | How can I use the AMI to get the IPs and Extensions of all my phones? |
21:21.03 | creativx | Action: command ? :p |
21:21.22 | X-Rob | sip show peers |
21:23.29 | *** join/#asterisk Carp1 (i=Carp1@ip-204-97-151-80.modem.logical.net) |
21:23.35 | Carp1 | +---- Asterisk Installation Complete -------+ |
21:23.35 | Carp1 | <PROTECTED> |
21:23.36 | Carp1 | <PROTECTED> |
21:23.36 | Carp1 | <PROTECTED> |
21:23.36 | Carp1 | <PROTECTED> |
21:23.38 | *** join/#asterisk [hC] (n=hardcore@66.119.176.4) |
21:23.44 | Carp1 | [root@pbx asterisk]# asterisk -vvvgc |
21:23.45 | Carp1 | bash: asterisk: command not found |
21:23.46 | Carp1 | :( |
21:23.53 | Carp1 | Do I need to type a path? |
21:24.33 | NotJohnDavid | /usr/sbin/asterisk |
21:24.44 | NotJohnDavid | that's not in your path apparently |
21:24.55 | Carp1 | yup |
21:24.59 | Carp1 | it starts when I type that |
21:25.37 | Carp1 | How do I make it so thats the path when I type just asterisk |
21:26.59 | NotJohnDavid | add /usr/sbin to your path |
21:27.27 | Carp1 | I dont know linux all that well |
21:28.04 | findlay | Carp1: I suggest you find a good command line tutorial |
21:28.14 | *** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66) |
21:28.51 | esculapio__ | Hola quien me puede ayudar con unos reporte en asterisk via web con una database |
21:30.13 | esculapio__ | Hello who can help with me it reports in asterisk in the Web with a database |
21:33.11 | *** part/#asterisk terrapen (n=cjs@166.70.183.108) |
21:35.48 | NotJohnDavid | why does PSTN pass thru not work on a SPA-3000 when I have Line 1 registered to a VoIP line. it's supposed to automatically pass thru using 127.0.0.1 |
21:35.57 | NotJohnDavid | I want to hurt someone |
21:38.30 | *** join/#asterisk vo (n=saigon@pdpc/supporter/basic/vo) |
21:38.48 | CrashHD | can anyone tell me how to get music on hold for call parking? it doesn't seem tow ork |
21:39.42 | *** part/#asterisk vo (n=saigon@pdpc/supporter/basic/vo) |
21:42.22 | FuriousGeorge | CrashHD: then you MoH isnt working at all |
21:42.24 | FuriousGeorge | is it? |
21:42.47 | FuriousGeorge | i can change the ringer via setting my sip header right? |
21:43.32 | CrashHD | FuriousGeorge: works fine during transfer and when call is placed on hold |
21:43.43 | FuriousGeorge | CrashHD: thats kinda odd |
21:43.48 | CrashHD | I thought so as well |
21:43.55 | FuriousGeorge | what version of *? |
21:44.00 | CrashHD | 1.2.9.1 |
21:44.05 | FuriousGeorge | im at a loss |
21:44.42 | CrashHD | I have [default] setup |
21:44.43 | CrashHD | works ok |
21:44.52 | CrashHD | is there a special option that needs to be set in features or something |
21:45.04 | CrashHD | I am using mp3's |
21:45.07 | CrashHD | with format_mp3 |
21:45.19 | CrashHD | I do see the moh start and stop 3 or 4 times in the CLI |
21:45.24 | FuriousGeorge | if it works on hold it should work for parking as far as i know |
21:45.51 | X-Rob | freepbx uses format_mp3 now, and moh in parking definately works |
21:46.30 | FuriousGeorge | anyone using snoms? isnt it possible to change the ring tone via the sip headder? |
21:46.50 | X-Rob | FuriousGeorge, yeah. you can set alert info to be a URL |
21:46.53 | X-Rob | it's on the snom wiki |
21:47.10 | esculapio__ | Hello who can help with me it reports in asterisk in the Web with a database |
21:47.22 | esculapio__ | Hola quien me puede ayudar con unos reporte en asterisk via web con una database |
21:47.27 | jhb | hi *, I would like to find out if a certain user (1234@sipgate.de) is available, maybe using something external for agi. Any hints? |
21:47.42 | jbalcomb | no hablo espanol |
21:47.50 | *** join/#asterisk Money5ack (i=moneysac@wer.will.spontanficken.de) |
21:48.05 | jbalcomb | jhb: the AMI examples have something that may be helpful |
21:48.20 | FuriousGeorge | X-Rob: can i just change the ringer to another one of the default ringers? |
21:48.42 | jbalcomb | esculapio__: are you using a reporting package or are you custom coding something? |
21:49.02 | X-Rob | FuriousGeorge, I think so. Check the snom wiki for howtos |
21:49.21 | X-Rob | Set(SIP_HEADER(Alert-Info: ...))something like that |
21:49.29 | jbalcomb | FuriousGeorge: I believe there are instructions on how to do that with Polycoms. I would think something similar is available for the SNOMs |
21:49.34 | CrashHD | http://pastebin.ca/74303 is the moh problem |
21:50.45 | jbalcomb | X-Rob: I see that Command: SIPpeers has the IP and Extension. Any thoughts on perl or php code to pull them out efficiently? |
21:51.03 | jhb | jbalcomb: I am looking at http://www.voip-info.org/wiki/view/Asterisk+manager+Examples - did you mean that? |
21:51.36 | jbalcomb | jhb: yes'm |
21:51.39 | jhb | jbalcomb: or asked the other way around - I don't have access to the foreigns ami |
21:51.54 | jhb | jbalcomb: thx btw |
21:52.00 | jbalcomb | jhb: ah, now that's rather different |
21:53.02 | jbalcomb | jhb: How about just trying to make a socket connection to that address? |
21:53.38 | jbalcomb | jhb: perl, python, or php probably has some module for SIP support that would let you interact enough to get a status |
21:54.17 | jhb | jbalcomb: I would not know the clients address - just their registration number at provider. So basically I am asking the proxy/registrar/whatever <- me newbie |
21:54.49 | jhb | jbalcomb: I could use twisted or so. Any hints on how to ask for a status using SIP? |
21:55.21 | jbalcomb | jhb: no, I haven't gotten that far yet. I might be able to answer that in a month or two as my project progresses |
21:55.42 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
21:56.32 | jhb | jbalcomb: thx. I will keep on searching. Maybe I find it, and can share with you |
21:56.41 | ariel_ | I have a quick Polycom question. How can I get the 2nd call inbound the phone to ring instead of flashing the light? |
21:57.35 | jbalcomb | Anyone want to be awesome and help me parse this output (http://pastebin.ca/74308) to grab the IP and extension? |
21:57.37 | [TK]D-Fender | ariel_ : it only goes through the handset unles you're on speaker |
21:57.41 | jbalcomb | You can be listed as a contributor to ZIPP (Z IP Phone Provisioner). =) |
21:57.43 | De_Mon | I've got some users whos ISP is blocking voip traffic, how can I get around it? |
21:58.01 | [TK]D-Fender | ariel_ : But you can change the CW "beep" sound in sip.cfg for a "ringing" wav if you like. |
21:58.06 | jbalcomb | De_Mon change the port and/or use a proxy |
21:58.17 | ariel_ | ok |
21:58.18 | [TK]D-Fender | jbalcomb : Change the name.. people will ask what it's worth ;) |
21:58.38 | ariel_ | but that is only in the head set |
21:59.05 | jbalcomb | [TK]D-Fender Its more a measure of what I know about developing software but I'm open to suggestions on the name. =) |
21:59.13 | jbalcomb | jhb: sounds good. |
22:00.11 | [TK]D-Fender | ariel_ : Yeah, thats just the way it is... then again... what phone really rings while you're on it? |
22:00.23 | jhb | jbalcomb: you would like to have a regex or what are you searching for? |
22:00.28 | ariel_ | nortels do |
22:00.57 | [TK]D-Fender | ariel_ : Norhell.. *shudder* |
22:01.09 | jbalcomb | jhb: if thats the right way to do it that would be fine. i just need to make sure i keep the IP and exten together so i can load them into MySQL |
22:01.18 | [TK]D-Fender | ariel_ : I wonder if you CAN force it somehow... nothing that stood out to me... |
22:01.18 | ariel_ | yes I know but it's what I am replacing |
22:01.35 | [TK]D-Fender | ariel_ : Congrats.... I did mine last Sep.... |
22:01.52 | ariel_ | I have switched about 10 setups already. |
22:02.04 | ariel_ | and you have different questions as you go through the change |
22:02.08 | jhb | jbalcomb: The ipaddress lines start with IPaddress, how do I (human) find the extension? |
22:02.34 | ariel_ | they sure love the paging/intercom which I am going to see if I can get working on the polycom soon. |
22:02.39 | *** part/#asterisk smackus (n=smackus@63.149.122.94) |
22:02.51 | *** join/#asterisk Manipura (n=chatzill@S01060011954c9c46.cg.shawcable.net) |
22:03.50 | jbalcomb | jhb: it's ObjectName in the same section |
22:04.26 | Manipura | Would asterisk run any better on a dual core processor? |
22:04.30 | jbalcomb | jhb: I do wish it had the User Name in there as well |
22:04.40 | *** join/#asterisk nortex (n=nortex@64.136.65.142) |
22:04.48 | jbalcomb | Manipura: it is recommended by digium. |
22:05.04 | Manipura | Dual Cores are recommended? |
22:05.09 | Manipura | Awesome.... |
22:05.10 | jbalcomb | Manipura: They also recommend against hyperthreading because of the interrupts it generates. |
22:05.23 | Manipura | Thank you..... |
22:05.26 | jbalcomb | Manipura: I have Dual Dual-Core 2.8 Ghz in mine |
22:05.44 | Manipura | I'm looking for a blade server |
22:06.31 | jbalcomb | Dell has nice ones, as does HP |
22:06.32 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
22:06.50 | jbalcomb | I'm out. G'Night yall. |
22:07.05 | CunningPike | Night, jb |
22:07.14 | Katty | later guys |
22:07.16 | *** part/#asterisk Katty (n=aisaacs@64.82.232.54) |
22:08.11 | *** join/#asterisk rayvd (i=rayvd@arthur.bludgeon.org) |
22:08.23 | rayvd | anyone remember what size fans were on the old oem Duron 700's? |
22:08.24 | rayvd | 45mm? |
22:09.32 | NotJohnDavid | that sounds about right |
22:11.45 | *** join/#asterisk saftsack (n=saftsack@p54A7DF23.dip.t-dialin.net) |
22:12.07 | rayvd | i think so too :) |
22:12.15 | Dr-Linux | anybody knows about Cisco Voip conference phone? |
22:12.41 | Dr-Linux | does it come with skinny protocol? |
22:13.26 | Qwell[] | Dr-Linux: yes |
22:13.30 | Qwell[] | 7935/7936 |
22:14.35 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
22:15.08 | Dr-Linux | Qwell[]: so just i'll need same way to load SIP firmware on it as i do on my cisco 7940/60 ? |
22:15.26 | pdtmobile | does anybody know if there is a reason that queues don't have but one general option? |
22:15.49 | CunningPike | pdtmobile: ?? |
22:15.51 | pdtmobile | it seems like they would work like the majority of the applications in asterisk and have general options for pretty much every option you can set in a section |
22:16.39 | Manipura | would having a faster hard drive ie, 10,000 or 15,000 RPM make voip any better? |
22:17.24 | CunningPike | Manipura: Unlikely - other than voicemail and moh, there's very little disk activity involved |
22:17.40 | Manipura | Thats what I thought... Thank you.... |
22:17.59 | pdtmobile | CunningPike: like for voicemail, sip, iax etc... you set things in general and override them in the various sections below if you need a different setting. |
22:18.15 | CunningPike | Manipura: There are 3 things that make it "better": Processor, processor and processor ;) |
22:18.22 | FuriousGeorge | so people that interface a SER sip gateway with asterisk... how does that work? the clients register with SER, which is registered with asterisk? |
22:18.35 | Manipura | CunningPike, Ram doesn't do much? |
22:18.36 | CunningPike | pdtmobile: What are you trying to do? |
22:18.40 | pdtmobile | queues only have one option thats global persistentmembers, everything else must be set for every queue or you use the in the source code defaults |
22:18.56 | pdtmobile | use defaults ;) |
22:19.08 | pdtmobile | globals/general settings whatever you want to call them |
22:19.09 | *** join/#asterisk Delta239 (n=delta_of@201.218.116.114) |
22:19.09 | Dr-Linux | Qwell[]: .... |
22:19.20 | CunningPike | Manipura: I don't think so....... I think that processor and bandwidth are the real constraints |
22:19.36 | pdtmobile | but looking through the code, if it sees a general section, it handles persistentmembers and quits |
22:19.38 | FuriousGeorge | ~ser |
22:19.39 | jbot | ser is probably Sip Express Router - see http://www.iptel.org/ser/ |
22:20.04 | CunningPike | pdtmobile: I see |
22:20.09 | pdtmobile | but other applications handle oodles |
22:20.10 | vader-- | how well do sip phones work over vpn? |
22:20.12 | Manipura | CunningPike, awesome, bandwidth is not a problem for me anymore ;) |
22:20.17 | pdtmobile | just wondering if that was intentional or lazy |
22:20.21 | vader-- | like as far as QOS goes |
22:20.36 | razu | is there any way to modify playback volume in asterisk with extension Playback() |
22:20.37 | razu | ? |
22:21.41 | Delta239 | hello.. im having some problems trying to register my broadvoice account |
22:21.43 | CunningPike | vader--: I haven't tried SIP - IAX works well |
22:22.05 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
22:22.09 | vader-- | i have a cisco 3000 vpn concentrator at our office |
22:22.10 | Qwell[] | Dr-Linux: I don't know if the 793x can use sip |
22:22.12 | pdtmobile | i can understand the lazy, I was just wondering if there was a reason before I "fixed" it |
22:22.19 | CunningPike | vader--: So do we :) |
22:22.30 | vader-- | and im trying to figure out a way that people could take phones and possible a vpn hardware device and connect to our asterisk server |
22:22.32 | vader-- | from home |
22:22.47 | vader-- | maybe some sort of linksys home device |
22:22.47 | Manipura | So is there a way to store Voicemail & such on a seperate server? And just run voip on one server with a small hard drive, then have a storage server to handle VM? |
22:22.50 | vader-- | that just does VPN shit |
22:23.00 | Dr-Linux | Qwell[]: hhm... |
22:23.03 | CunningPike | vader--: Actually, ours is a 3500 |
22:23.09 | vader-- | you got the bigger one |
22:23.11 | vader-- | :) |
22:23.18 | vader-- | you shwartz is bigger than mine |
22:23.20 | vader-- | your |
22:23.36 | CunningPike | Manipura: Two ways - nfs to another server, or have another asterisk server for voicemail. We do nfs |
22:23.45 | CunningPike | vader--: Heh |
22:24.15 | vader-- | i was pissed when we got ours and found out it doesn't support vlan's |
22:24.33 | Manipura | CunningPike, whats nfs stand for? |
22:24.37 | CunningPike | vader--: I'm not sure if ours does |
22:24.39 | vader-- | network file system |
22:24.45 | CunningPike | ~nfs |
22:24.46 | jbot | Try using -o nolock,soft (soft will help stop the error cascades, especially over wireless) |
22:24.55 | pdtmobile | vader--: snom phones if i remember correctly have vpn capability built in |
22:25.35 | vader-- | i have cisco 7940G |
22:26.34 | pdtmobile | well looks like i was wrong must have been somebody else |
22:26.49 | pdtmobile | but I looked at a phone at one point that had that built in so you wouldn't have to carry around two devices |
22:27.05 | Qwell[] | don't all the ciscos do vlan? |
22:27.16 | pdtmobile | and on the second ethernet port it will tunnel whatever you hook into it |
22:28.26 | *** join/#asterisk japerry (n=japerry@216.231.51.208) |
22:28.27 | jhb | jbalcomb: http://pastebin.ca/74346 |
22:28.40 | jhb | good night * |
22:28.46 | japerry | CunningPike: so I talked to verizon, they won't do a PRI for under 800 |
22:29.09 | vader-- | hmm i don't think opening asterisk directly to the internet is a good idea either |
22:29.14 | pdtmobile | hell I was way off |
22:29.16 | pdtmobile | it's avaya |
22:29.30 | CunningPike | japerry: That's for a full PRI, I sincerely hope |
22:29.35 | japerry | Cunningpike: and you can't channelize it I guess.. so I'm wondering if its worth it to call repair and see if it can be changed from e&m wink to groundstart or loopstart |
22:29.35 | pdtmobile | http://news.thomasnet.com/fullstory/476738 |
22:29.42 | vader-- | we pay 385 for our 23 channel pri |
22:29.46 | japerry | CunningPike: yup. full |
22:29.52 | japerry | vader--: lucky :-P |
22:29.52 | pdtmobile | vader--: thats a good price, where you at? |
22:29.55 | vader-- | paetech |
22:29.58 | CunningPike | vader--: Not to everyone, but to specific IPs might be OK |
22:30.12 | vader-- | we are in usa/de |
22:30.15 | vader-- | delaware |
22:30.25 | CunningPike | japerry: We get our partial PRI from Allstream |
22:30.30 | vader-- | we hired a company to go out to all the different providers and get us pricing |
22:30.44 | vader-- | and we picked which one we wante |
22:30.45 | vader-- | d |
22:31.11 | japerry | Cunningpike: now only if it was 100miles south of the border |
22:31.45 | pdtmobile | i can't remember what we pay, we only have 13-23 on ours and I am pretty sure we pay more than that |
22:31.59 | CunningPike | japerry: ? |
22:32.36 | japerry | Cunningpike: I'm not sure if they have T1 service out here? |
22:33.00 | CunningPike | japerry: Allstream - I don't believe so...... |
22:33.13 | FuriousGeorge | is there a way to set cadence in the dialplan for zap channels |
22:33.30 | japerry | Cunningpike: anywho, so do you think it'd make any difference if we switched it to groundstart or loopstart? |
22:33.38 | CunningPike | FuriousGeorge: Don't think so - why would you want to set it per call? |
22:34.00 | CunningPike | japerry: It can't hurt..... also confirm your wink settings with them |
22:34.51 | kpettit | I'm in macro hell |
22:35.05 | japerry | hehe, I wonder if its featd possibly |
22:35.31 | CunningPike | kpettit: Share your pain |
22:36.12 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
22:36.17 | kpettit | can i not set extensions in a macro? I can s,1, stuff ok but if I do exten=>1,1,.... I get that "extension 1 is not in the XXX context" xxx being the context where I first called the macro |
22:36.21 | *** join/#asterisk metfan (n=metfan@dsl-201-129-222-101.prod-infinitum.com.mx) |
22:37.16 | metfan | hi, I need help with a TDM400, somebody helpe!!! |
22:37.32 | CunningPike | kpettit: Macros don't have extensions the way that dialplan entries do. You have already done your pattern matching in your exten => when you called Macro |
22:37.42 | Strom_C | metfan: just ask your question |
22:37.48 | Strom_C | metfan: someone will answer |
22:37.58 | CunningPike | kpettit: So s is a placeholder for the extension |
22:38.17 | kpettit | bugger. |
22:38.23 | CunningPike | kpettit: If you need to refer to the extension as a variable, use ${MACRO_EXTEN} |
22:38.54 | kpettit | I'm passing all the variables I need to the macro, and I've even tried calling another context from within the macro but I can't seem to pass the variables i need |
22:39.09 | CunningPike | kpettit: Pastebin |
22:40.10 | metfan | Ok, I have a TDM400 with my Asterisk PBX, everything works ok, I can make out calls, and recieve another ones.. but the problem is when the other party from the PSTN hangsup, the call does not disconnect, the telco does not sends the busy tones, any advise?? |
22:41.15 | Dr-Linux | anybody knows if Cisco 7935/7936 supports SIP ? |
22:41.53 | kpettit | CunningPike, pastebin.com is crawling... |
22:42.00 | kpettit | any other good pastebin's? |
22:42.07 | De_Mon | pastebin.ca |
22:42.35 | De_Mon | ??paste |
22:42.39 | kpettit | thanks |
22:42.42 | De_Mon | wrong channel |
22:42.45 | kpettit | CunningPike, http://pastebin.ca/74355 |
22:43.02 | Strom_C | metfan: your telco needs to provision far-end disconnect supervision on your analog lines |
22:43.06 | kpettit | the [people] context is where I have all my extensions and where I call the macro |
22:44.19 | kpettit | CunningPike, with what I have there option 2 works, but the voicemail options don't |
22:44.38 | CunningPike | kpettit: OK - looking now.... |
22:44.45 | kpettit | I was origionally trying to put all of that in the macro, but it didn't like my exten's 1 and 2 in the macro. |
22:44.46 | kpettit | thanks |
22:45.06 | metfan | Strom_C: Do I need to ask my telco for that??? what if my telco does not support it? |
22:45.19 | Strom_C | metfan: your telco should support it |
22:45.53 | Strom_C | metfan: are you in north america? |
22:46.07 | metfan | Strom_C: yes, in Mexico |
22:47.39 | Dr-Linux | anybody knows if Cisco 7935/7936 supports SIP ? |
22:47.44 | metfan | Strom_C: that service is provisioned per line??? I mean, is it a special feature to ask to the telcos?? |
22:49.13 | kpettit | CunningPike, I've tried both {MACRO_EXTEN} and {EXTEN} with the same results |
22:50.02 | CunningPike | kpettit: Ya - neither will work. ${MACRO-EXTEN} will have died because you're no longer in a macro, and ${EXTEN} will be 1 or 2 |
22:50.43 | kpettit | any suggestions? |
22:51.24 | Strom_C | metfan: it's per-line yes |
22:51.33 | kpettit | I was thinking of doing just a context rather than a macro but then I can't pass all the var's |
22:51.33 | Strom_C | usually the telco should automatically provision it |
22:51.42 | Strom_C | i only know of u.s. and canada though |
22:51.44 | CunningPike | kpettit: Use Set() to set a variable in to ${MACRO_EXTEN} in your macro (or to ${EXTEN} in your original context) and refer to that |
22:52.32 | CunningPike | kpettit: Set(foo=${MACRO_EXTEN}) |
22:53.05 | kpettit | then I just call ${foo} ?? |
22:53.06 | CunningPike | kpettit: Then, exten => 1,1,VoiceMail(${foo}|u) |
22:53.18 | kpettit | ah cool, let me try that. thanks |
22:53.53 | nortex | Is it possible to build conference rooms dynamicly based on a extension that transfered the call to the macro/meetme application? |
22:54.28 | CunningPike | nortex: I believe so, but I have no direct experience - I think there's something in the wiki about it |
22:54.33 | CunningPike | ~thewiki |
22:54.34 | jbot | thewiki is, like, at http://www.voip-info.org/wiki-Asterisk |
22:55.05 | CunningPike | nortex: Not very helpful, I'm afraid |
22:55.22 | kpettit | CunningPike, ohhh kick ass, that worked great. |
22:55.29 | metfan | anybody knows if far-end disconnect supervision is a special service on most telcos or is a default service? |
22:55.31 | kpettit | CunningPike, so that's basically like setting up a global |
22:55.55 | CunningPike | kpettit: Not quite - a global would be same for all calls - the scope of that variable is per call |
22:56.13 | kpettit | got ya. that's pretty sweet |
22:56.21 | CunningPike | kpettit: Glad it worked out |
22:56.34 | kpettit | I'm going to go nuts with that, there are a couple places that will be really usefull for me |
22:56.47 | CunningPike | kpettit: Hee hee - go nuts ;) |
22:56.50 | kpettit | Hey maybe you can help wiht a really odd questions I've been stumped on all day |
22:56.59 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
22:56.59 | kpettit | I know I can do this in the source code, but I'd like to find a better way. |
22:57.00 | nortex | CunningPike, No problem, I had read through that, but it really dosen't do dynamic the way I thought it would. |
22:57.02 | CunningPike | kpettit: I can try..... |
22:57.04 | shmaltz | anybody here using an MSI AMD motherboard? |
22:57.40 | kpettit | CunningPike, with call parking. Three is a timeout, after the timeout it goes back to the sip presence that made the parked call, but there is no timeout on how long it rings that sip presence |
22:57.56 | kpettit | I think it goes about 60 seconds and hang's up. I'd like to be able to set that timeout. |
22:58.17 | CunningPike | kpettit: Let me check our setup....... |
22:59.09 | kpettit | when you park a call it creates a [park-dial] on the fly. which is cool, I'm able to set a priority 2 for for those returning parked calls, but it's the timeout I can't fix |
23:00.22 | kpettit | My overall goal would be to not change C code, and have the parked call that timed out go to my [ringall] type context so somebody gets that missed parked call. |
23:00.34 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
23:00.57 | kpettit | I can do that with a 2 priority but i have to wait for that 60 seconds for that sip call back to fail |
23:01.44 | CunningPike | kpettit: OK - I have to wait for 120 seconds here :D |
23:02.25 | shmaltz | how do I tell lilo to pass noapci to the kernel? |
23:02.27 | kpettit | you can set the timout in features.conf for how long the parked call will wait, then whhen it rings back it just does Dial(Sip/XXX) with no time set |
23:03.08 | kpettit | I think it would be usefull overall to set a default timeout for a generic Dial() statement but nothing in sip.conf or extensions.conf I've tried has worked |
23:04.03 | Dr-Linux | anybody knows if Cisco 7935/7936 supports SIP ? |
23:04.41 | CunningPike | kpettit: OK - in our setup (using Polycoms) the call goes back to where it came from - i.e. it resumes at the same point in the dialplan where it was when it was parked - so it rings extension again as if the caller had dialed the number again. So the normal Dial() takes over |
23:04.58 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
23:05.00 | CunningPike | kpettit: Not sure if that makes any sense at all lol |
23:06.23 | kpettit | I can see it doing the Dial, there is just no timeout set. |
23:06.44 | kpettit | According to what I found earlier that's the way it was in the source code. But that was what i was told, I'm not enough of a C guy to know for sure |
23:07.04 | CunningPike | kpettit: But is there a timeout on the Dial statement for that extension normally? |
23:07.46 | kpettit | you mean 700? |
23:07.57 | kpettit | or the one that got the call before they parked it |
23:08.14 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
23:08.39 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
23:08.44 | mds2 | does anyone know how to shorten the period of time Cisco 79xx phones spend "Configuring VLAN" when they boot? |
23:08.51 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
23:08.56 | CunningPike | kpettit: The one that got the call and is now having the call returned to it |
23:09.17 | kpettit | yes |
23:09.22 | kpettit | there is atimeout of 30 seconds |
23:09.34 | kpettit | that's what I have to my phone (polycom as well btw) |
23:09.35 | *** join/#asterisk ctaloi (n=Chris@cpe-24-58-22-17.twcny.res.rr.com) |
23:10.11 | kpettit | actually it's 20 seconds |
23:10.27 | Beighto | Asterisk on a cluster, is it possible? |
23:10.57 | CunningPike | kpettit: And it's executing that Dial() statement as far as you can tell? |
23:10.59 | kpettit | CunningPike, but I know it's ringing back at least 60 seoncds, so I'm possitve it's not picking up from where it origional dialed the extension |
23:11.06 | CunningPike | kpettit: Ah OK |
23:11.10 | kpettit | <PROTECTED> |
23:11.15 | kpettit | that's my excat dial statement |
23:11.40 | CunningPike | kpettit: What does your park extension look like? |
23:11.52 | kpettit | I just use the default from features.conf |
23:12.08 | kpettit | I park to 700, it gives me back 701-705 to where it parks it. |
23:12.16 | ctaloi | im having some trouble routing ingress SIP calls, i've got the friend defined in sip.conf and am passing it to my extensions.conf, but I get a 404 back from Asterisk when I place the call from PSTN->AST, anyone feel like helping?? |
23:12.50 | CunningPike | kpettit: We have this in our extensions.conf: |
23:12.54 | CunningPike | exten => callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1) |
23:13.39 | CunningPike | kpettit: The internal,${DIALEDPEERNUMBER},1 bit determines where the call goes on timeout, so you can make that whatever you want |
23:13.41 | kpettit | wow that's fancy |
23:14.02 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
23:14.20 | CunningPike | kpettit: It was the only way we could get the Polycom Park key to work - how are you doing it? |
23:14.27 | kpettit | where do you get "callpark" and "ParkAndAnnounce" from. I haven't used those before |
23:14.35 | kpettit | I just transfer to 700 |
23:14.59 | kpettit | it tells me where it's parked. Then they page or whatever to call holler for who they want |
23:15.23 | CunningPike | kpettit: callpark is specific to Polycoms - it's what they dial when you press the Park key. ParkAndAnnouce is an asterisk application |
23:15.48 | kpettit | what model do you have? I don't have a Park key on the 501 |
23:16.16 | CunningPike | kpettit: It's a softkey - usually on More during a call |
23:16.16 | kpettit | we're 90% polycom 501 and the rest 601's and 301's |
23:16.23 | CunningPike | kpettit: Same here :D |
23:16.32 | kpettit | its' not a default key is it |
23:16.39 | CunningPike | kpettit: Actually - we have no 301's |
23:16.46 | kpettit | they suck |
23:16.54 | CunningPike | kpettit: It should be - unless you have the feature turned off.... |
23:17.03 | kpettit | how do you get that soft key? I don't have that on the default polycom image |
23:17.16 | CunningPike | kpettit: Even during a call? |
23:18.07 | CunningPike | kpettit: During a call, press 'More' - it should be there |
23:18.13 | kpettit | yeah I'm on a call now and IU haved hold, endcall, transfer, confrence |
23:18.22 | ctaloi | can I send all my ingress SIP calls to one context in extensions.conf and route the calls based on the call-ed number? |
23:18.36 | kpettit | I don't have a more button. Just those 4 soft keys |
23:18.52 | CunningPike | kpettit: OK - you need to enable the feature - hang on a sec |
23:19.16 | kpettit | oh hell yeah, that'll be cool |
23:20.17 | kpettit | I figured out how to remove some keys, DND, call forwarding and that sort of hting. but that's as fancy as I've got |
23:20.29 | CunningPike | kpettit: Look in your sip.cfg for <feature and then for a feature with a name of "call-park" |
23:20.42 | brijn | exit |
23:20.59 | kpettit | ok got it |
23:21.06 | CunningPike | kpettit: Set feature.x.enabled (can't remember what the x is) ="1" |
23:21.10 | kpettit | feature.11.name="call-park" feature.11.enabled="0" |
23:21.16 | kpettit | just change that to a 1 I guess |
23:21.20 | CunningPike | You got it! |
23:21.31 | CunningPike | kpettit: Then reboot your phone |
23:22.26 | kpettit | CunningPike, are you running 1.6.6? |
23:22.32 | kpettit | that's what we've been using latgely |
23:22.33 | CunningPike | kpettit: Yes |
23:22.46 | CunningPike | kpettit: 1.6.6 with 3.1.3 bootrom |
23:22.53 | kpettit | same here. |
23:23.41 | kpettit | I'm still anxious to figure out what your doing what that park line you gave me above, tath's really cool |
23:24.36 | kpettit | ok I have the "park" option now. |
23:24.44 | kpettit | donse't really do anything in it's current state though |
23:25.01 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
23:25.12 | CunningPike | kpettit: Basically, the Polycom will call an extension called callpark. The ParkAndAnnounce app is documented here: http://www.voip-info.org/wiki-Asterisk+cmd+ParkAndAnnounce |
23:25.45 | kpettit | ah ok. |
23:26.29 | CunningPike | kpettit: The important thing for your purposes is return_context - you can send the returning call anywhere you like |
23:27.38 | kpettit | <PROTECTED> |
23:27.38 | kpettit | what's the differnce? |
23:28.43 | kpettit | oh this is a trip, man this is alot of fun toys to play with |
23:30.42 | kpettit | CunningPike, I think this is eactly what i need though, thanks a ton. |
23:30.57 | kpettit | CunningPike, don't happen to be in the Houston area do ya? |
23:30.59 | CunningPike | kpettit: The first one is who to call to tell where the call is parked. The second is where the parked call goes on a timeout |
23:31.08 | CunningPike | kpettit: No - Vancouver, BC |
23:31.12 | kpettit | bugger |
23:31.18 | CunningPike | kpettit: But we have the internet........ ;) |
23:31.21 | kpettit | Trying to hire some more people that do asterisk here |
23:31.37 | CunningPike | kpettit: ssh is your friend ;) |
23:32.03 | kpettit | yeah i wish. Most of the stuff we need help for is new installs, and that needs somebody onsite. |
23:32.24 | CunningPike | kpettit: Ah |
23:37.14 | *** part/#asterisk Beighto (n=chatzill@64.160.113.130) |
23:37.16 | CunningPike | Dr-Linux: What's your question? |
23:38.12 | Dr-Linux | CunningPike: |
23:38.15 | Dr-Linux | anybody knows if Cisco 7935/7936 supports SIP ? |
23:38.35 | Dr-Linux | it's conference VOIP phone |
23:39.33 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
23:39.33 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
23:41.37 | CunningPike | Dr-Linux: Looks like it's SCCP only, according to voipsupply |
23:42.04 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
23:42.13 | terrapen | best music-on-hold evar: |
23:42.18 | Dr-Linux | CunningPike: so can i use it with Asterisk? |
23:42.22 | kpettit | Cisco's suck for SIP. I hate them sooooo bad |
23:42.31 | terrapen | "Yakety Sax" (the Benny Hill theme music) |
23:42.46 | CunningPike | Dr-Linux: Only if you run chan_scp - do you feel lucky? ;) |
23:43.00 | kpettit | I've got a Cisco 7940 that briked becuase a sip upgrad didn't work |
23:43.02 | CunningPike | Dr-Linux: Why not get a SoundPoint IP4000? |
23:43.23 | CunningPike | kpettit: I broke a Blackberry in similar fashion...... |
23:43.24 | terrapen | kpettit, switch to polycom...best decision i made |
23:43.35 | Dr-Linux | CunningPike: bcoz just we recieved shipment from USA and got this phone |
23:43.43 | Qwell[] | Dr-Linux: No, it doesn't... |
23:43.50 | kpettit | I'm really happy with the Polycom's |
23:44.01 | Qwell[] | http://www.cisco.com/cgi-bin/tablebuild.pl/ip-7900ser |
23:44.07 | CunningPike | Dr-Linux: Guess you're stuck running SCCP then |
23:44.21 | Dr-Linux | i have polycom too but that's not VOIP phone |
23:44.33 | Dr-Linux | CunningPike: i don't know how to setup SCCP |
23:44.39 | Qwell[] | Dr-Linux: send it here...I'll get it working with chan_skinny |
23:44.41 | Qwell[] | :p |
23:44.47 | CunningPike | Dr-Linux: Cut your losses and get an ATA for your Polycom |
23:45.12 | Qwell[] | kpettit: send me the b0rked 7940 too :P |
23:45.38 | Dr-Linux | CunningPike: i have ATA as well |
23:45.48 | CunningPike | Dr-Linux: You're all set |
23:46.09 | CunningPike | Qwell[]: Want my fscked Blackberry too? |
23:46.11 | Dr-Linux | but this new cisco conference phone is not set :( |
23:46.13 | Qwell[] | CunningPike: sure! |
23:46.54 | drray | Qwell, you sell phones? |
23:46.58 | Qwell[] | drray: no |
23:47.07 | Qwell[] | I want to make them work, heh |
23:47.36 | drray | qwell, I am looking for a phone with a cleaner transfer function than a cisco 7960, I have users that are too stupid to transfer correctly with it |
23:48.06 | CunningPike | drray: No offense, but your users must be pretty stupid |
23:48.11 | drray | yes |
23:48.13 | drray | I get that |
23:48.18 | CunningPike | drray: Have you tried Polycom? |
23:48.20 | drray | it is only one site |
23:48.26 | drray | no.. but I will |
23:49.14 | CunningPike | drray: We haven't had too many problems with our users (except the firefighters, but they're a special case) - parking has been a challenge, but transfers are fine |
23:49.25 | drray | 601? |
23:49.44 | CunningPike | If you lock a firefighter in a bare room with two steel ball bearings, he lose one and break the other |
23:49.44 | drray | you have to scroll to a second screen on the cisco |
23:49.55 | CunningPike | drray: Really? wow |
23:50.01 | CunningPike | drray: That bitrs |
23:50.07 | CunningPike | s/bitrs/bites/ |
23:50.15 | drray | or I don't have it configured right |
23:50.28 | kpettit | CunningPike, I'm trying to make parlk work correctly. I already have pagin setup to work on the phones. I do *XXXX to page a individual phone. SO with parked calls I'm tring to do ... |
23:50.37 | drray | and I don't need park |
23:50.40 | kpettit | <PROTECTED> |
23:50.47 | kpettit | which dosen't seem to do thr trick |
23:51.51 | CunningPike | kpettit: Your 'internal,*2032,2' needs to be a Dial() style string - SIP/foo |
23:52.06 | kpettit | ok |
23:54.15 | *** join/#asterisk iq|mobile (n=iq@unaffiliated/iq) |
23:55.02 | kpettit | how are you pagin then? |
23:55.14 | kpettit | I'm using Page(Sip/XXXX) type of thing |
23:55.26 | kpettit | not sure how I can do that in the park |
23:56.59 | *** join/#asterisk anthm (n=anthm@h460856b1.area4.spcsdns.net) |
23:57.00 | *** mode/#asterisk [+o anthm] by ChanServ |
23:57.09 | kpettit | I do exten => _*XXXX,1,Set(_ALERT_INFO="Ring Answer") exten => _*XXXX,2,Page(SIP/${EXTEN:1}) to page. Trying to figure out how I can still page a announce from the callpark |
23:57.48 | CunningPike | kpettit: We're not doing paging - we just call the parker back |
23:58.04 | kpettit | ah. I'm wanting to page just for the announce. |
23:58.11 | kpettit | the call back does eactly whaqt I want though. |
23:58.38 | CunningPike | kpettit: Cool |
23:59.09 | kpettit | and I've set a couple short cuts so I can page through all the polycom phones |
23:59.26 | kpettit | so in my mind it would kick as to park a call and have it announces on all the polycom speakers. |
23:59.38 | CunningPike | kpettit: Neat - we haven't played with paging yet |
23:59.52 | kpettit | it's pretty cool, saved us alot in dumb paging equipment |