irclog2html for #asterisk on 20060628

00:00.10japerryboth of those error messages come with varying gibberish in debug every second or so
00:00.21directorythey aren't errors
00:00.26directorythey are debug messages, vastly different
00:00.38japerrydirectory: err debug message. sorry =)
00:01.09mrdigitalhey CunningPike: can i pm u
00:01.13CunningPikejaperry: Hmm - I'm beginning to wonder about network latency or some other issue
00:01.34CunningPikemrdigital: Spit it out right here - we're all friends here ;)
00:02.00mrdigitalwhats the difference between SPA 3000 and 3102
00:02.08rob0102!
00:02.08Druken102
00:02.10Druken:)
00:02.17japerryCunningPike: 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
00:02.19CunningPikeSpot the comedians
00:02.50Drukena 4 channel t1?
00:02.50CunningPikemrdigital: It is supposed to be a newer replacement for the 3000 - not sure what the difference is
00:03.03mrdigitalwill the 3000 work just fine?
00:03.10Drukenmine works...
00:03.29CunningPikejaperry: What switches are you using?
00:03.31japerrythe T1 is called 'flexgrow'
00:03.44CunningPikejaperry: Just tossing out ideas here, really
00:03.48Drukenmines called fantasy
00:03.52japerryCunningpike its a D-Link soho 24port 10/100 switch
00:03.53mrdigitalDruken: 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
00:04.16Drukenit can work that way...
00:04.19japerrythe flexgrow however doesn't act like a PRI
00:04.34japerryit acts more like 4 PSTN lines bundled onto a T1 line
00:04.47mrdigitali have 1 pstn. and 1 analog phone i want to use with asterisk
00:04.56japerryso channels 9-12 are being used and configured as fxo
00:05.14Drukenmrdigital: then you want the 3000 or 3102
00:05.39CunningPikejaperry: So, what do your zaptel and zapata files look like?
00:05.42CunningPike~pb
00:05.43jbotrumour 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/
00:05.43mrdigitaland thats it besides the comp?
00:06.15Drukenmrdigital: yep
00:06.17*** part/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com)
00:06.43mrdigitalcan i mod the box to connect to multiple analogs?
00:07.00Drukenmeaning?
00:07.08CunningPike:S
00:07.48mrdigital<pstn line> ------ <spa3000> ----- <rj11/cat3cable> ------ <box 1> ------- <box 2> ------ <box 3>
00:07.57mrdigitalbox 1 - 3 = normal phone outlet
00:08.19Drukenuhmm.. yeah... just plug in as many phones as you want...
00:08.26CunningPikemrdigital: If you want them all to be on the one line, I guess - not sure what REN the SPA is able to provide
00:08.46japerryheh
00:08.46mrdigitalren?
00:08.49japerrypastebin not working
00:08.51Drukeni know my rt31p2 rings like 6 phones in my house
00:09.03CunningPikejaperry: try pastebin.ca
00:09.11CunningPike~ren
00:09.19h3xthats coz current day phones have a ren of like 0.000001
00:09.43Drukenexactly
00:09.44Druken:)
00:09.58Drukenunless you got an old rotary phone... it's no big deal
00:10.06h3xa crystal radio prob uses more power
00:10.06h3xheh
00:10.10mrdigitalso i should be ok?
00:10.18Drukenyep
00:10.25mrdigitaltheres 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)
00:12.25phigworkgiesen: Sorry, are you still there?
00:12.28Drukenthis is such a fucked up tv show....
00:12.29japerryCunningpike: its e&m
00:12.42CunningPikeHmmm
00:12.56Drukenanyone ever seen "big love" ?
00:13.33phigworkgiesen: If I do exten => _9.,1,Dial(Zap/1,${EXTEN:1}), it just gives me a dialtone when I dial 9+anything (which I can then dial out from, but that's silly)
00:14.01*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
00:14.04Drukenuhmm... change that , to a /
00:14.07CunningPikejaperry: I gotta run for home - I'll be back on later. I'd focus on the 'pri debug' side of things and see what comes up, seeing as it's external outgoing calls only
00:14.20CunningPikejbot ren is Ringer Equivalence Number - a telephone line can normally supply upto 4 REN, where a standard telephone/answering machine etc would equal 1 REN
00:14.22jbotCunningPike: okay
00:14.25japerryCunningpike: okay, me too  http://pastebin.ca/73561
00:14.51h3xi would say an old phone with a bell ringer in it
00:14.54japerrycunningpike: thats the pastbin for zapata.. you'll notice I statically assigned the callerid because the other method was reulting in no numbers
00:15.13CunningPikejaperry: OK
00:15.29CunningPikeI'll do some thinking on the ride home :)
00:16.33Drukenphigwork: you make that change?
00:17.16phigworkDruken: which one?
00:17.16japerrycool, thanks =)
00:17.30Druken[20:14] <Druken> uhmm... change that , to a /
00:18.03phigworksorry missed that :)
00:18.44Druken:)
00:19.14*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
00:19.34phigworkDruken: i don't think it's dialing all the numbers
00:19.47phigworksitting at silence atm.. then three tones, we're sorry, your call did not go through
00:19.53Drukenwell, not with your old way... no :)
00:20.09phigworkexten => _9.,1,Dial(Zap/1/${EXTEN:1})
00:20.47Drukenshow me the cli output
00:20.56Drukencause that would be working
00:20.57phigwork<PROTECTED>
00:21.14phigworki got the "remember, you must now dial 1 + the area code, or 0 + the area code"
00:21.20phigwork<PROTECTED>
00:21.36Drukenit did dial the 1.. hehe
00:21.44phigworkmaybe the card is dialing too fast
00:21.54Drukencould be...
00:21.56phigworknot enough time between pick up of line and dialing
00:22.16phigworkis there a pause?
00:22.23phigworkpause character, that is
00:22.30Drukennope
00:22.38phigwork:/
00:22.46phigworkatdt,
00:22.46Drukenbrowse in zapata.conf
00:22.47phigwork:)
00:23.20Drukennot dialing a modem....
00:23.21Drukenhehe
00:23.45phigworkyeh, 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)
00:24.51chumper2342How do I create custom intro voice menus? Like "Thanks for calling Company name"
00:24.54phigworkah well. I guess I'll just let it dial the 9
00:25.38phigworkchumper: 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.
00:26.18chumper2342i know its possible, cause the company I work for has a custom intro
00:26.51chumper2342i think it might cost money
00:26.59phigworkDrukon: how do you add strings together? commas?
00:27.18Drukenadd string together?
00:27.25phigworkconcatinate
00:27.31InfraRedchumper2342: save your own .wav file
00:27.34InfraRedthen read the wiki
00:27.36Drukenlike math? or just stringstring ?
00:27.40phigworkstringstring
00:27.47Drukenexactly like that
00:27.51phigworkcool
00:27.56Drukenexactly like that ${exten}
00:27.59Drukenbah!
00:28.07Druken${exten}${exten}
00:28.10phigworkthanks
00:28.42benjamin7062Chumper... you are looking for something like this:
00:28.44benjamin7062exten => s,n,Playback(tt-somethingwrong&tt-weasels)
00:29.12chumper2342yep
00:29.43phigworkoh ya, i've been wondering looking through examples and stuff, what is N?
00:29.44benjamin7062Open 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
00:29.45benjamin7062poof
00:29.49phigworkoh number, duh
00:29.55phigworkand X is any character?
00:29.56benjamin7062n is the 'next' priority
00:30.00phigworkoh
00:30.01phigworkword
00:30.01benjamin7062instead of numbering the priorities
00:30.05DrukenNEXT! :)
00:30.49chumper2342how can I get it to sound like the lady?
00:30.58phigworki'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)
00:31.00benjamin7062You can also do n(name) and call a priority by 'name' with a Goto Application
00:31.06phigworkis there a better way of setting this up?
00:31.16*** join/#asterisk coppice (n=chatzill@223.193.17.210.dyn.pacific.net.hk)
00:31.19benjamin7062phigwork, not really
00:31.25phigworkcool
00:31.28Drukenchumper2342: get allison to record it for you
00:31.41benjamin7062phigwork, don't forget international.. =)
00:31.48benjamin7062_X011XXXX etc
00:31.54benjamin7062err.. that was wrong
00:32.10phigworki'm not payin no long distance :)
00:32.10benjamin7062_9011XXX
00:32.14benjamin7062whatever.. you get the point
00:32.48benjamin7062=)
00:33.07*** join/#asterisk droops (n=root@adsl-065-005-212-128.sip.jan.bellsouth.net)
00:34.51coppicewow. 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)
00:35.19phigworki wonder why you can't call 800 numbers through fwd anymore
00:37.29benjamin7062coppice, That's because Intel wants to see a P4 in every phone... instead of cancer... you then have to worry about skin burns.
00:38.07benjamin7062coppice, They are trying to get people to stop making calls while driving
00:38.30Drukenain't gonna happen
00:38.49benjamin7062Beh... you can wear batteries around your waist... it'll happen
00:38.54benjamin7062=)
00:39.26Drukenain't gonna happen people stop using the phone while driving :)
00:39.44coppiceintel'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.
00:40.04coppicewill Dialogic be next?
00:41.18coppicethey can't get 802.11 working well, either, but they still aim to take the world by storm with WiMax :-)
00:42.29anonymouz666my friend won here... "intel hero" with wimax projects
00:42.52benjamin7062Wireless net everywhere... Wi-whatever... I don't care... but I want Wi(something) everywhere in a major city... That will simply be HAWT
00:42.59benjamin7062something needs to do this.. NOW.. and for free
00:43.17benjamin7062coppice, that's a good business model... lets start a company
00:43.32benjamin7062If we could go back to 1997 I bet we could get a BUTT load of VC
00:44.18coppiceif 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
00:44.47Drukenbenjamin7062: uhmm... yeah... toronto hydro has ya beat :)
00:44.48anonymouz666hah
00:46.58*** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com)
00:47.47rob0LOL @ http://www.voip-info.org/wiki/view/Cheapest+ATAs+and+Service ... "VoIPSUPPLY PLEASE QUIT DELETING OUR LINKS"
00:47.58anonymouz666the ISP's here wanna drop all the VoIP traffic. No more VoIP with domestic ADSL modems
00:50.55benjamin7062anonymouz666, I would hate to be with an ISP that does that.
00:51.27benjamin7062rob0, that's dirty...
00:52.17anonymouz666benjamin7062: No choice. The ISP's are the Telcos.
00:52.35rob0Yeah, if that's true, VoIPSupply ought to be kicked off the wiki.
00:53.10rob0If not, Digiumcards.com ought to be kicked off. :)
00:54.06rob0You give a greedy and stupid merchant free advertising ... they do greedy and stupid things.
00:55.03Drukeni guess i was never a greedy and stupid merchant....
00:56.21chumper2342anybody 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
00:56.46chumper2342i set shorewall rules ACCEPT all all udp 10000:33000
00:56.47X-Rob_chumper2342, give your asterisk box a non-natted IP address.
00:56.57chumper2342it has a public ip
00:57.05*** join/#asterisk Freman (n=twitsrus@jaguar.wbs.net.au)
00:57.07chumper2342it was working yesterday i believe
00:57.20[TK]D-Fenderchumper2342 : You need 5060, 10000-20000, and a bunch of settings in [general] in sip.conf
00:57.20X-Rob_well, reboot stuff until it starts working again then.
00:57.33[TK]D-Fenderchumper2342 : externip, localnet, and nat=yes
00:57.34chumper2342sure
00:57.48X-Rob_anyone had any experience with the netcomm v85 phones?
00:58.19Fremanany 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-FenderFreman : 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.
01:01.22FremanI'd rather not duplicate everything for garry tho
01:01.24chumper2342actually, i can hear sound for half a sec when i pickup my cell (using my softphone to call cell)
01:01.41Fremanlike... 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)
01:09.05phigworkman, I'm still wishing for a skype gateway, or at least some way to log in to skype
01:10.00Carp1Which ports do I need open?  I am using NuFone connecting with IAX and Using SIP for IP Phones
01:13.09rob0Hey! 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)
01:15.02rob0I have 2000/tcp, and {2727,4520,5060,4569}/udp bound. I know what some of those are, but not all.
01:15.38rob0http://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)
01:16.26rob0What 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)
01:22.50BrijnGood evening
01:23.08Greek-Boyoops
01:23.09Greek-Boysorry
01:23.11Greek-Boymistake
01:23.55tengulre11Good morning!!
01:24.33Greek-BoyI can't record as described in http://www.voip-info.org/wiki/view/Monitor+stereo-example
01:24.34Greek-Boy:(
01:24.38Greek-Boycan someone help me?
01:25.33[TK]D-FenderGreek-Boy : Pastebin all the related bits of your config.
01:25.35[TK]D-Fender~pb
01:25.37jbotfrom 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-FenderGreek-Boy : and CLI output...
01:29.18rob02000/tcp appears to be SCCP
01:31.19*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
01:32.20Greek-Boyhttp://pastebin.com/734096
01:32.24Greek-Boyno CLI output
01:34.18chumper2342can 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)
01:36.13rob0Looks like I might close SCCP with a "noload => chan_skinny.so" in modules.conf ?
01:36.18chumper2342when I use my cell and call in, i can hear audio both ways
01:37.23rob04520/udp appears to be DUNDi, which I probably want to keep, IIUC what it does.
01:42.45[TK]D-FenderGreek-Boy : You goofed pretty big... you have to do monitor BEFORE you dial...
01:43.38Greek-Boylol
01:44.05*** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net)
01:44.08rob0Looks 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?
01:44.20CrashHDhow can I pass a var from one a sub macro to the parent macro?
01:44.24Carp1I just intalled asterisk and it used to config it for me
01:44.31CrashHDs/one/
01:44.38Carp1Just type 'asterisk ' and it starts
01:44.42[TK]D-Fenderrob0 : nope... I killed Dundi, MGCP, and SCCP myself..
01:44.45Carp1whats its path/
01:44.52rob0[TK]D-Fender: thanks
01:44.54*** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net)
01:45.04chumper2342anybody seen this in cli when making call? Forcing Marker bit, because SSRC has changed
01:47.52CrashHDhow 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)
01:49.45chumper2342got it, forgot externip and localnet in sip.conf
01:51.22CrashHDI 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-FenderCrashHD : Use AstDB.
01:52.16CrashHDbut those vars are persistent correct?
01:52.26[TK]D-FenderCrashHD : and pass the UNIQUEID as part of the keyvalue
01:52.26CrashHDwould that not conflict with multiple calls
01:52.28CrashHD?
01:52.36[TK]D-FenderCrashHD : AstDB = consistant.
01:53.07CrashHDI 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-FenderCrashHD : Yup.
01:53.52CrashHDno easy way to just pass a return value eh?
01:53.55CrashHDlol
01:54.21[TK]D-FenderFirst 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-FenderCrashHD : Imperfect solutions for an imperfect world.
01:55.06CrashHD*nods*
01:55.18CrashHDI might as well just use an AGI at that point
01:55.44CrashHDthanks for the heads up
01:58.53[TK]D-Fendernp
01:59.20[TK]D-FenderThere are just so many places that this is the only really viable mechanism for passing values...
01:59.21Greek-Boy[TK]D-Fender is it correct to use $SOXMIX -v 1 $LEFT-tmp.wav -v 1 $RIGHT-tmp.wav -v 1 $OUT.mp3?
01:59.33chumper2342actually setting the externip and localnet only fixed 1 call, it worked once but still doesn't
01:59.38[TK]D-FenderGreek-Boy : Beats me..... I suck at Linux :)
01:59.38Greek-Boydoesn't seem to work for me. the output mp3 comes out empty :(
02:01.27Carp1How to I start asterisk?
02:01.27Carp1?
02:02.33Carp1./path/to/asterisk -vvvc
02:02.34Carp1?
02:02.34Greek-BoyCarp1 /usr/sbin/asterisk -g
02:02.35ManxPowerCarp1, "service asterisk start" or "asterisk" or "safe_asterisk"
02:02.40Greek-Boyread the wiki
02:03.10*** join/#asterisk {DDP}Natas (n=natas212@69-165-73-122.sbtnvt.adelphia.net)
02:03.22Carp1no such file or directory
02:03.23Carp1hmm
02:03.27{DDP}Natashi everyone
02:03.32Carp1I installed using SVN
02:04.14ManxPowerAnd?
02:04.47Carp1[root@pbx asterisk]# service asterisk start
02:04.47Carp1bash: service: command not found
02:04.48Carp1[root@pbx asterisk]# /usr/sbin/asterisk -g
02:04.48Carp1bash: /usr/sbin/asterisk: No such file or directory
02:04.52{DDP}Natasdoes anyone have any experience with a2billing?
02:04.59Qwelldid you install it?
02:05.12anonymouz666what means bridge two channels together? ZAP 1 and SIP-channel for example?
02:05.14Carp1I followed hte download instructions
02:05.19anonymouz666I did'n't understand
02:05.24QwellCarp1: which are?
02:05.24Carp1and at the end of the install it showed a big Asterisk in ASCII
02:05.43QwellYou didn't follow the instructions
02:05.53Carp1http://www.asterisk.org/download
02:06.00Carp1I followed the SVN instructions
02:06.21Qwelland what about the crap that it said on your screen during the install?
02:06.32Carp1It was going to fast
02:06.38Qwellspecifically the part where it says "make must be restarted" at the VERY end
02:06.49Carp1Yeah, I seen that
02:06.52Qwelland?
02:07.17*** join/#asterisk skraelings001 (n=skraelin@190.40.104.78)
02:07.24Carp1I ignored it because when I installed asterisk like 5 years ago I remembered a whole bunch of errors and everything worked fine
02:07.32Carp1Can you tell me whatr that means?
02:07.36Qwellit means...
02:07.40QwellMAKE MUST BE RESTARTED
02:08.05Carp1I don't know how to do that, is what I meant.  Sorry
02:08.14Qwellyou type make install...
02:08.18Carp1right
02:08.47Carp1I did that
02:08.58Carp1I did "make clean; make instal"
02:09.03Carp1install*
02:09.15anonymouz666Qwell did you already use an app called bridge?
02:09.25Carp1Does that mean to make install again?
02:09.25QwellIf you can't follow instructions...don't use trunk
02:09.30QwellCarp1: YES
02:10.00Carp1How 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.
02:10.57QwellLike I said..  If you can't follow instructions...don't use trunk
02:11.28directoryQwell: but I want to!
02:11.37Qwelldirectory: no trunk for you either
02:11.46Qwelldirectory: You must maintain 1.0.x now
02:11.47Carp1That still doesnt help me get asterisk started
02:11.47directorydarn
02:11.58{DDP}Natasif 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)
02:13.51userdefinedanyone running * behind an ipcop firewall?
02:14.57userdefinedi'm having a heck of a time getting internal->dmz to asterisk working, just curious if/how others have done it
02:15.23userdefined(tried asking on #ipcop, they're apparently all sleeping over there ;-)
02:16.04rob0I don't use ipcop, but am pretty handy with iptables.
02:17.02skraelings001good evening
02:18.40userdefinedrob0: 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 =/
02:19.02X-Rob_heh
02:19.13X-Rob_iptables -P FORWARD ACCPT
02:19.17X-Rob_iptables -F FORWARD
02:19.22X-Rob_repeat for input and output
02:19.24X-Rob_then start again
02:19.33X-Rob_and, spell ACCEPT properly, too 8)
02:19.36rob0xyzzy!
02:20.09userdefinedheh
02:20.37*** join/#asterisk onweald_tim (n=onweald_@c-67-173-213-205.hsd1.tx.comcast.net)
02:21.12rob0userdefined: what specifically is your * trying to do that the firewall is blocking?
02:21.30rob0userdefined: 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)
02:21.58userdefinedasterisk to/from the net is working
02:22.10userdefinedproblem is from the 'green' interface to the 'orange' is failing
02:22.21userdefined(green == trusted, orange == dmz in ipcop parlance fwiw)
02:22.40rob0SIP phones? *What* is it doing?
02:23.04userdefinedrob0: timeout on register
02:23.14rob0SIP?
02:23.22userdefinedaye, sorry
02:23.55rob0Usually the RELATED,ESTABLISHED rule should cover this. I presume ipcop uses that.
02:24.38userdefinedit does
02:24.52userdefinedand as i understand it, by default green->orange should be permitted
02:24.53rob0go ahead and pastebin your iptables-save
02:25.01userdefinedk. one sec
02:25.14rob0make it "iptables-save -c"
02:25.32rob0(doesn't matter much)
02:27.10userdefinediptables -n -L work ?
02:27.17userdefinednot sure where ipcop keeps the save
02:27.22*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
02:27.38rob0well it's not as good, but I'll look
02:27.48userdefinedthat's here-> http://pastebin.ca/73631
02:28.51rob0oops, no -v on that
02:28.56userdefined(fwiw, iptables-save isn't present on ipcop, not just being ornery =0)
02:28.57rob0so it's useless
02:29.01userdefinedah. one sec, i'll fix
02:29.38onweald_timHi all.  I am trying to get SIP working through a vonage router (motorola VT2442) and I think  that vonage uses sip too.
02:29.41onweald_timSo is this an impossible senario?
02:30.18onweald_timEssentially, two SIP connections can't NAT if you want connections in both directions.
02:30.23onweald_timThat is my theory...
02:30.24rob0userdefined: I take it that the * server is in the DMZ and the SIP clients are in the LAN?
02:30.56rob0SIP in green, * in orange?
02:31.15onweald_timLooks like this: Grandstream gxp2000 -> Vonage router -> Cable modem -> internet -> cablemodem -> general router -> asterisk server
02:31.51userdefinedrob0: correct
02:31.53onweald_timMy config, not userdefined in case there is any confusion... :-)
02:32.55userdefinedrob0: http://pastebin.ca/73633  <-- iptables -Z;iptables -v -n -L
02:34.58rob0so which ethX is which?
02:35.08userdefinedeth0 == green, eth2 == orange
02:35.46userdefinedit 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)
02:36.43rob0is 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)
02:38.57rob0192.168.3.3 is the * IP ... maybe it's a NAT problem
02:39.37*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)
02:39.48rob0Add rules to Chain PORTFWACCESS to ACCEPT for -i eth0
02:40.14*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
02:41.21rob0hmmm, looks like line 33 is a blanket ACCEPT for eth0
02:44.19rob0You 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?)
02:46.00userdefinedi'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)
02:47.03rob0a route, as in "ip route add ..." or "route add ...": ip(8) or route(8).
02:47.43rob0anyway, I'm done for ... afk.
02:47.47userdefinedah. as opposed to a route as in "a firewall rule" ... which isn't a route at all =P
02:47.51rob0right
02:48.27rob0if you're still having the trouble tomorrow I'll be around.
02:48.46userdefinedcool. i'll keep poking at it, thanks for taking a look
03:08.28benjamin7062If 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:
03:08.31benjamin7062app_dial.c:713 wait_for_answer: Unable to forward voice
03:11.09Brijnbenjamin7062: where you the one setting sup the Prove of Concept before monday?
03:11.34BrijnProof :)
03:12.03benjamin7062Yes, and it worked.
03:12.14benjamin7062Now I jsut compiled on a new machine using 2.16.x kernel and udev
03:12.28benjamin7062and, well, dif. installation issues
03:13.09benjamin7062The PoC went very well... we invested in the phones today so now I have to get this sucker live.
03:13.14benjamin7062I have... well, 3 days
03:13.15benjamin7062yippee
03:14.54BrijnHahaha
03:15.09Brijndid 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)
03:16.55Brijnbenjamin7062: http://lists.digium.com/pipermail/asterisk-users/2003-August/011207.html
03:17.37BrijnNever used a PRI, but can you enable debug on PRI messages
03:19.56benjamin7062Brijn, Yeah... tried all the 'normal' stuff
03:19.59benjamin7062D channels are right
03:20.07benjamin7062heck, I'm using the same config from the other machine
03:20.25BrijnAdn that box is OK? same driver version, same *?
03:21.01benjamin7062yup
03:21.04benjamin7062=(
03:21.19benjamin7062I had seen that email thread
03:21.25benjamin7062checked google and wiki first.
03:21.26benjamin7062no love
03:21.39benjamin7062thought 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)
03:22.19pdavidevening all
03:22.41pdavidanyone 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)
03:25.29jsaundersAnyone 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.
03:25.55Brijnbenjamin7062: does the other box have the same PRI card, identical?
03:26.44benjamin7062Brijn, yeah -- pulled the card from the other system
03:27.26BrijnBox-A, Pri-A = problem, Box-B, Pri-B = OK, Box-A Pri-B = ?
03:27.30BrijnWhere PRI is the card
03:29.09Brijn~fxs
03:29.14jbotit 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
03:30.11jsaundersHmm, 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'.
03:30.33jsaundersThis 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)
03:31.03seb-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)
03:31.58jsaundersseb: What voip phone you talkin' about?
03:32.54pdavidno 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)
03:33.50directoryspackle: it's a spackle!
03:35.42*** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net)
03:39.18benjamin7062sigh
03:39.22benjamin7062well, I got it working
03:39.40benjamin7062But it works only if I disable the EC on the Sangoma
03:39.43pdavidmaybe someone could walk through a quick setup of the spa3000 with me?
03:39.47benjamin7062which is the entire reason we bought the Sangoma
03:39.49benjamin7062bleh
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)
03:41.32morcegaoCan 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)
03:42.30jaybhey guys... im having problems with the blindtx bug..
03:42.41jaybcan somone suggest how to turn off blind transfer all together ?
03:44.22jaybthis is the bug im refering to : v
03:44.23jaybhttp://bugs.digium.com/view.php?id=7289
03:45.51[TK]D-Fenderbenjamin7062 : whats the problem with it?
03:45.52*** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
03:46.44morcegaoany knows register string using outbound proxy ?
03:47.32benjamin7062[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)
03:48.14benjamin7062debug on the span's shows a lot I should say... just nothing out of the ordinary
03:49.03benjamin7062Followed several versions of directions from sangoma and the wiki... thus far no luck
03:52.10spacklebenjamin7062 - 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)
03:52.49benjamin7062spackle, I disabled per instructions in the wanpipe* configs... and left on in * as sangoma states this is required.
03:53.11spacklenew card or current firmware?
03:53.19benjamin7062brand new
03:53.34benjamin7062Yeah... me too
03:53.38*** part/#asterisk skraelings001 (n=skraelin@190.40.104.78)
03:54.23*** join/#asterisk cypromis (n=michal@voiceworks.pl)
03:54.58morcegaoany knows how to crete register string using outbound proxy ?
03:55.05benjamin7062I 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..  =)
03:55.10benjamin7062I just did and rebooted
03:55.11benjamin7062who knows
03:55.59benjamin7062nope
03:56.00benjamin7062no love
03:56.18spackleare you connected to a PRI or a channel bank?
03:56.24benjamin7062PRI
03:56.52benjamin7062Wonder if the latest sangoma beta is compatible with * 1.2?  Maybe it only works with 1.0?
03:57.40benjamin7062I 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
03:57.44spackleit only has to work with zaptel
03:58.02benjamin7062spackle, err, good point
03:58.28benjamin7062Perhaps this is a plot by digium to push their cards o.O
03:58.37directoryha
03:58.42spacklenah.
03:58.52spacklego back over your settings
03:59.24benjamin7062Perhaps I'll find a line # if (sangoma) { loop forever and make noises };
03:59.42directorywouldn't that be funny
03:59.54benjamin7062I'd laugh... even if it was a comment
04:00.30benjamin7062thing is... the card works fine as long as I turn off EC
04:00.33benjamin7062;-)
04:00.51*** join/#asterisk netoguy (n=skelley@ppp-70-129-186-62.dsl.spfdmo.swbell.net)
04:00.51benjamin7062so 'something' in the zap driver needs to wake up and listen to the EC channel or something
04:01.49benjamin7062I would guess it would be something in the zaptel.conf but .. I see no relavence...
04:01.51benjamin7062bum sticks
04:01.55jaybany ideas guys on how to fix my blind transfer problem ??
04:02.47Greek-Boywhy can't I have macros in my dialplan for non patterns?
04:04.54onweald_timAny suggestion on how to trace sip through two routers +  two cable modems to determine which is causing the problem?  The firewalls look right.
04:05.17benjamin7062onweald - tcpdump?
04:05.56onweald_timbenjamin7062: That would trace it on one side or the other but I don't know which router/modem is rejecting the packets.
04:06.12onweald_timWhat I really need is a tracert for SIP ports
04:06.39onweald_timI know there is some utility that will do it but I'll be damned if I can't find it.
04:06.55onweald_timUsing debian and I have searched apt up and down.
04:06.59benjamin7062do you see the traffic on both ends.. or neither?
04:07.25onweald_timIt looks like the control ports are working but I get audio in one direction only.
04:07.39benjamin7062is one behind a nat?
04:08.06onweald_timSporadic audio.  I think one of the routers is flaky.
04:08.09onweald_timBoth are behind nat
04:08.28onweald_timI think the other router is flaky but can't prove it to the person on the other end.
04:08.35benjamin7062hmm... i assume you did a * forward on both firewalls to the asterisk box just to test?
04:08.58onweald_timWe went so far as to set both up on the DMZ.  No good.
04:08.59benjamin7062* = wildcard... forward all packets incoming to * box
04:09.08benjamin7062hrrm
04:09.15onweald_timYeah.  Very frustrating.
04:09.20benjamin7062dmz could still = firewall
04:09.26benjamin7062did you check the firewall on the * boxes?
04:09.37onweald_timImportant info: My side is a vonage router.
04:09.42benjamin7062(just running through things in my head)
04:10.03onweald_timChecked the firewall on the asterisk box.  iptables -L showed everything clear.
04:10.13onweald_timjust during testing :-)
04:10.38benjamin7062jayb, have you tried changing blind transfers to a completely dif feature... just to see if that removes it?
04:10.54benjamin7062what kind of routers/firewalls?
04:11.21jaybI put blind transfer as **
04:11.35benjamin7062jayb, did you do a reload?  (just asking)
04:11.36jaybbut 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)
04:11.46benjamin7062oh
04:11.47benjamin7062right
04:11.47jaybdid reload res_features
04:11.48onweald_timGrandstream gxp2000 -> Vonage Motorola VT2442 -> Cable modem -> internet -> cablemodem -> general router w/SIP support -> asterisk server
04:11.49benjamin7062umm
04:11.56onweald_timNot sure exactly what his router is.
04:12.21benjamin7062jayb, can you change blind to ## or ###?  Dunno if that works.. never tried.  ;-)
04:12.29jaybbenjamin - do you know anyway to turn blind transfer off totaly ??
04:12.31spacklebenjamin7062 - did you see the stuff in sangoma wiki about hardware echo can?  http://sangoma.editme.com/wanpipe-asterisk-configure
04:12.45benjamin7062jayb, not without disabling in the code... i'm sure there is a better way
04:12.45jaybbenjamin = 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)
04:12.59jaybbut it still means DTMF # is not sent out to any endpoints..
04:13.05benjamin7062spackle, yeah.  Unfortunately, I did..  but I'll go read for giggles to make sure
04:13.07jaybthe blind transfer functionality STOPS it
04:13.33benjamin7062jayb, good point.  Wonder what happens if you leave it blank?
04:13.40jayband it does the same thing no matter if its "#", "*", "##", "**"
04:13.47[TK]D-Fenderbenjamin7062 : 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-Fenderbenjamin7062 : I've dealt with this twice before
04:13.58jaybyea I might try that later tonight..
04:14.08jaybI have lots of calls in the system now, so dont want to go TO experimental..
04:14.09benjamin7062[TK]D-Fender, hawt... let me check
04:14.28jaybim actualy looking at jumping in there and recompiling res_features without any blind transfer support
04:14.32benjamin7062jayb, if you type asterisk -vx'stop now' that will fix your call problem...
04:14.33benjamin7062heheh
04:14.37jaybif worst comes to worst..  but I Dont want to do that if I can avoid it
04:14.55benjamin7062s/v/r/
04:14.59[TK]D-Fenderjayb : Running Zap analog phones?
04:15.07jaybnope..
04:15.13jaybonly 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-Fenderjayb : then why are you using features.conf for blind transfers at all?
04:16.05benjamin7062[TK]D-Fender, so, umm, where is wan_ec?  /dev  /proc?  locate/find no match
04:16.18[TK]D-Fenderbenjamin7062 : /etc/wanpipe
04:16.34onweald_timAdios.  Gotta go get some sleep.
04:16.40benjamin7062maybe that's my problem... dir doesn't exist.
04:16.47benjamin7062onweald_tim, good luck!
04:16.48[TK]D-Fenderbenjamin7062 : Tha may be OK
04:17.03spacklebenjamin7062 - do you have the udev devices set up with correct permissions?
04:17.07[TK]D-Fenderbenjamin7062 : it moved after a certain revision of wanpipe.
04:17.14[TK]D-Fenderbenjamin7062 : cruise for it.
04:17.19jaybD-Fender -- well how do I turn it off though !?
04:17.20[TK]D-Fenderunder /proc I think
04:17.27onweald_timThanks benjamin7062.  Appreciate your brain cycles.  Post here if you think of anything else.  :-)
04:17.35jaybit was commented out in features.conf... but that defaults to # for blind transfer
04:17.38benjamin7062onweald_tim, will do!
04:17.39[TK]D-Fenderjayb : don't use "tT" when you Dial
04:17.51benjamin7062spackle, hrrmm???  Maybe that's my problem
04:18.05benjamin7062OMG... Why didn't I think of that
04:18.09jaybive also removed Tt from the dial statement.
04:18.13benjamin7062@ jayb
04:18.27n3glvhe guys, why can't * handle the fourth column of DTMF?
04:19.11n3glvis there a conf somewhere to jiggle to allow A B C D touchtones?
04:19.24benjamin7062spackle, if * is running as root and /dev/wp* is owned by root... I assume permissions are okay?
04:19.44benjamin7062same with /dev/zap/*
04:19.46spackleyep.
04:19.56spackleprolly
04:20.52n3glvis x86 around out there?
04:23.01benjamin7062I might have my problem
04:23.11spackledo tell
04:23.20benjamin7062Setup 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-FenderWhat version are you running?
04:23.37benjamin7062if wan_ec is the root.. it doesn't exist in source anymore  I'd have to read more if this is relavant
04:23.48benjamin7062wanpipe-beta4-2.3.4.tgz
04:26.15benjamin7062The 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)
04:26.45n3glvI would like to do some specialized controll stuff with the 4th col of DTMF
04:27.39[TK]D-Fenderbenjamin7062: Sounds like a botched install all right
04:28.42benjamin7062Maybe 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)
04:29.40P-NuTHi all.
04:30.11spacklebenjamin7062 - prolly won't work with the echo can then.
04:31.01benjamin7062Well, those poop faces should include the EC stuff in their source!  o.O
04:31.13P-NuTSay, 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?
04:31.22directory[TK]D-Fender: !!!
04:31.31jaybguys one other question...
04:31.39CunningPikeP-NuT: You don't even really need asterisk-addons or asterisk-sounds
04:31.39jaybasterisk & Vmware...
04:31.49P-NuTI don't?
04:31.55jaybis anyone doing this on any system that has a moderate amount of calls... ??
04:31.58CunningPikeP-NuT: They're optional extras
04:32.04P-NuTwhat are they?
04:32.18CunningPikeP-NuT: Are you planning to use MeetMe or MOH?
04:32.27P-NuTMOH yes..
04:32.42benjamin7062jayb, -- I've talked to 'many' who do when evaluating the system
04:32.47P-NuTDamn, I can't connect to svn.digium.com
04:33.07CunningPikeP-NuT: You'll need ztdummy then for timing, so you'd better install zaptek
04:33.15P-NuTzaptek?
04:33.18CunningPikes/zaptek/zaptel/
04:33.32P-NuToh ok then.
04:33.38jaybhehe .
04:33.52jaybyea I realise Ill need ZTDummy, but even with that.. Ive heard of problems with timing in vmware.
04:33.58n3glvon some SMP kernels there is an issue with zaptel / ztdummy
04:34.14jaybyea ok... anything to identify these kernels ??
04:34.15n3glvtrixbox for one ships with a bad smp setu
04:34.16n3glvsetup
04:34.32jaybcoz Id like to run a BIG (Quad or something) server
04:34.36benjamin7062sigh... all their damn code points to wan_ec -- they 'forgot' to include it with the source
04:34.37n3glvonly matters for dual cpu or dual core
04:34.38benjamin7062argg
04:34.44jaybwith VMWare ESX, and a few ASterisk instances.. and Id like it rock solid..
04:35.13n3glvvmware running under unix?
04:35.17jaybif its not possible, then so be it.. just wanted to hear if anyone has actualy done it
04:35.26n3glvit can be done
04:35.31n3glvam sure someone is doing it
04:35.32jaybyea vmware running in unix.. (ESX is built on top of redhat enterprise)
04:35.44[TK]D-Fenderdirectory : I DON'T WANT TO MEET YOU MOM!
04:36.03jaybid just like to hear from that "Somebody" who is doing it.
04:36.08directory[TK]D-Fender: hahahahaha
04:36.11jaybto talk about potential pitfalls, and what to look out for :)
04:36.12directory[TK]D-Fender: so THAT'S what you're doing
04:36.16n3glvthere 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)
04:36.24n3glvlot of university systems etc
04:36.43benjamin7062Sigh
04:36.46benjamin7062I hate moments like this
04:36.52benjamin7062Where I realize that I'm an idiot
04:36.59jayb:P
04:37.14n3glvok jay, the one I know of is from trixbox, I have that issue personally
04:37.22n3glvbut if you install from source etc
04:37.25P-NuTDoes anyone work for digium here?
04:37.29n3glvthen there seems to be no problem there
04:37.31jaybyea I do build asterisk from source..
04:37.31benjamin7062The 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.
04:37.35directoryP-NuT: what'cha need?
04:37.48benjamin7062<-- steps on his own toe
04:37.53benjamin7062<-- HARD
04:37.55jaybn3glv - but do you know what kernel problem is with trixbox ??
04:38.13P-NuTI was going to say that the SVN repositories were offline, but they've come good now.
04:38.20jaybn3glv -- 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 :)
04:38.23n3glvno, have not gotten that far into it, just know that the zaptel stuff is broken
04:38.30jayboh ok.
04:38.32n3glvI can get u a kernel id
04:38.38Juggiejayb, distro?
04:38.47jaybIm still open to suggestions.
04:38.52Juggiewhat distro are you using?
04:38.54jaybwas thinking CentOS..
04:38.59Juggiegood idea
04:39.00Juggieexcept
04:39.06Juggie??centos
04:39.07*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:39.11Juggie?? centosbug
04:39.12benjamin7062n3glv, I've just compiled 3 times using the 2.4 and 2.6 kernels just fine... with the debian base installs.. FYI
04:39.16benjamin7062Did one today
04:39.20Juggiegrrr....
04:39.21benjamin7062In case that helps
04:39.24Juggiejbot, centosbug
04:39.26jboti 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"
04:39.31n3glv2.6.9-34.ELsmp
04:39.34Juggiedont forget that :)
04:39.46jaybid 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)
04:39.57Juggieyou assumed correctally.
04:40.02Juggieyou can use the centos vanilla kernel
04:40.06Juggieno need to recompile
04:40.14Juggiejust be sure to install kernel-devel
04:40.14Juggieand your good to go
04:40.20jaybok..
04:40.20Juggieand of course
04:40.23Juggiejbot, centosbug
04:40.25jbotcentosbug 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"
04:40.29Juggieas i said.
04:40.31Juggiedont forget that
04:40.33Juggieor zaptel will bitch
04:40.35jaybhaha ok
04:40.59jaybok great...
04:41.01benjamin7062So, jayb -- What are you going to run before you compile?
04:41.03benjamin7062hint
04:41.05benjamin7062jbot, centosbug
04:41.06jbotcentosbug 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"
04:41.13jaybjbot, centosbug :)
04:41.16benjamin7062ahole!  Don't ignore me!
04:41.23benjamin7062heh
04:41.25jaybhehe .
04:41.27benjamin7062I got ignored by a computer
04:41.32benjamin7062~jbot
04:41.34jbotsomebody said jbot was only marginally useful at best,  He got a C- on his Turing Test, or a complete idiot, or a dolt
04:41.36benjamin7062~fart
04:41.37jbotACTION farts, releasing large quantities of methane and sulfur dioxide. "Evacuate the channel! GO! *gag* SAVE YOURSELVES *cough* MOVE *choke* MOVE!"
04:41.41jaybok so I can do a clean centos 4.2 install... run that command
04:41.48jaybthen build / install asterisk  and it should be good
04:41.52Juggieuhhuh
04:41.54jaybto run in a VMWare guest OS ??
04:42.00Juggiehah
04:42.00Juggieno
04:42.05P-NuTjbot: So, do I need to setup ztdummy, or is it just there automatically for the timing of MOH?
04:42.07jbotP-NuT: what are you talking about?
04:42.16Juggiewhy would you run * in vmware
04:42.21Juggiei guess it would be ok without any hardware
04:42.24Juggiebut it wont work with hardware
04:42.25P-NuTjbot wrote: jbotCunningPike meant: P-NuT: You'll need ztdummy then for timing, so you'd better install zaptel
04:42.33jaybthats what my question was relating to (above :P)
04:42.45directoryjbot: botsnack
04:42.45jbot:), directory
04:42.47jaybyea no hardware... just sip & IAX2
04:43.03P-NuTjbot: that's what I'm talkking about
04:43.29benjamin7062jbot -- you smell funny
04:43.39n3glvanyone got a url on that kernel fix for centos? is it in the wiki?
04:44.05jaybjbot, centosbug :)
04:44.10jaybis that what your after ? :)
04:44.14jaybjbot, centosbug
04:44.15jboti 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"
04:44.50benjamin7062jbot 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
04:47.29Juggiesure sounds like a bug
04:47.35Juggiesee 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 :(
04:48.24Juggiewhat trunk version works?
04:49.14_alex_mx_ouch, how to tell :)
04:49.18P-NuTCunningPike: What does asterisk addons actually give you?
04:49.35P-NuTCunningPike: and does it require mysql to run
04:49.35Juggiealex, 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)
04:55.01spacklebenjamin7062 - get it yet?
04:55.07benjamin7062Sadly... yes
04:55.13spackleeh?
04:55.38benjamin7062There is a seperate packages that adds to the source of the wanpipe driver
04:55.47benjamin7062which contained wan_ec
04:56.03benjamin7062The worst part is... it's in BIG BOLD LETTERS on their site
04:56.04spackleyeah, I was asking if you got that and installed it yet?
04:56.20spacklehave it working?  that sort of thing
04:56.24benjamin7062One might think I could read... but perhaps elementary school is where I should start before installing a * box
04:57.01benjamin7062Yeah, I recompiled, unloaded drivers... downed the iface, did the oposite.. ztcfg
04:57.08benjamin7062and I have crystal clear audio to my cell
04:57.10benjamin7062using EC
04:57.16spacklecool
04:57.34directorybenjamin7062: 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
04:57.44benjamin7062directory, not true
04:57.50benjamin7062directory, I read
04:57.56benjamin7062directory, if there are big pictures
04:57.56*** part/#asterisk netoguy (n=skelley@ppp-70-129-186-62.dsl.spfdmo.swbell.net)
04:58.04directorybenjamin7062: with pink arrows?
04:58.12benjamin7062directory, and voice!
04:58.20directorygood idea
04:58.22benjamin7062directory, and water colors involved
04:58.55benjamin7062honestly, i skimmed over that several times but it 'looks' at a skim like it's pointing to the same driver
04:59.07benjamin7062I thought it was saying something needed to be 'enabled'... which I thought I was doing
04:59.09benjamin7062beh
04:59.32directoryit's all good now
05:00.07benjamin7062I'm talkin' on it now and it's workin' good
05:04.00CrashHDwhat is the 1.2 way of pulling a db variable?
05:04.16CrashHD${DB(family/whatever)} ?
05:08.40*** join/#asterisk DjTremors (n=newjacks@stargate.citadelcomputer.com.au)
05:09.19DjTremorshey 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-Fenderbenjamin7062 : sO ALL IS GOOD NOW?
05:13.02[TK]D-FenderCrashHD : yup
05:13.03*** join/#asterisk cryptnix (n=andrew@64.25.198.123)
05:13.25*** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com)
05:13.32cryptnixAnyone 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-Fendercryptnix : They have a very comprehensive sample on their site...
05:14.01maskedyo
05:14.20maskedcan the spa9000 have multiple connections to a ITSP?
05:15.17cryptnixah, well basically i'm trying to connect to a broadsoft enabled server with no luck
05:15.58cryptnixasterisk -> sip account for my lines
05:16.09cryptnixpretty much mocking sipura settings ... with no luck
05:16.21cryptnixyet 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)
05:21.00benjamin7062[TK]D-Fender, yes, now I have to fix my music on hold
05:21.03benjamin7062sounds like a robot
05:21.10benjamin7062=)
05:21.17benjamin7062but 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)
05:21.42FuriousGeorgeif I had 200 sip clients i would probably need SER as a gateway, righ?
05:21.48FuriousGeorge*right
05:22.10Juggienot necessairly.
05:22.31Juggiei would ping jerjer and ask, he doesnt use ser, only static peers
05:22.32Juggieno realtime
05:22.48DjTremorscan anyone help with voicemail? I'm getting 'WARNING[27435] app.c: No audio available on SIP/225-9032??'
05:23.04DjTremorsi end up with a blank voicemail file.
05:23.04FuriousGeorgeehh, he's usually cranky at this time
05:23.21FuriousGeorgeon an unrelated note, where does one go to buy 200 PCs
05:23.51drrayFuriousGeorge - dell?
05:23.58FuriousGeorgedrray: makes sense
05:24.08FuriousGeorgedrray: unfortunately
05:24.10spackleHP, Lenovo?
05:24.13drraywith 200 you could probably send them an image
05:24.25benjamin7062FuriousGeorge, NewEgg =)
05:26.06drrayI built, imaged, and tested 75 beige boxes one weekend
05:26.14drrayhaving done it, I'd say make dell do it
05:26.37tclarkcurious if anyone has any wifi handsets in production with symbol ap 100 access point in a large warehouse environment ?
05:28.04tclarkor any other wifi config in a large 250K sq ft home depot type big box envronment ?
05:28.07FuriousGeorgebenjamin7062: i dont think even they carry that kind of stock that i wouldnt limit myself
05:28.55benjamin7062FuriousGeorge, I think you're right... =)
05:29.03FuriousGeorgedell it is i guess
05:29.18spackleugh
05:29.19benjamin7062Besides, if someone told me to 'build' 200 machines... I'd quit
05:29.34FuriousGeorgebenjamin7062: yeah, i didnt even think about that aspect
05:29.45benjamin7062lol
05:29.48drrayyou hire a $9 hr screw driver monkey
05:29.51spacklewhat is the DOA ratio on Dell?
05:30.08drrayyou hire him for a week
05:30.11benjamin7062spackle, they are actually pretty good.. but in that volume I've had some bad ones
05:30.25benjamin7062Cool thing about dell... you call, they overnight... done
05:30.28drraythe pita is not building the boxes, it's deploying them
05:30.42FuriousGeorgeso whos in NJ
05:30.53benjamin7062drray, you can pay dell for that too! =)
05:30.58drrayyouget one from dell, and trick it out the way you like, then you send them an ikmage
05:31.07spackleit's summer, hire a college student
05:31.11drraythen every one you get after comes with it
05:31.27drrayhell, 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)
05:32.05benjamin7062drray, those guys can give you excuses.  (I have to do my other job) when they realize IT sux
05:32.28drraya monkey with a screwdriver can do that job
05:32.43FuriousGeorgeknife fighting cable monkey
05:32.51FuriousGeorge~furiousgeorge
05:32.53jbotfrom memory, furiousgeorge is a knife-fighting monkey last seen with The Man with the Yellow Bat
05:33.11justinuwhere is yellow bat man, anyways?
05:33.14FuriousGeorgejbot: no,  furiousgeorge is a knife-fighting (cable) monkey last seen with The Man with the Yellow Bat
05:33.18jbotFuriousGeorge: okay
05:33.18benjamin7062FuriousGeorge, I think you've found your man then... GO into the bathroom... look above the sink
05:33.30FuriousGeorgelol
05:33.38justinu~seen r-evolution
05:33.51jboti haven't seen 'r-evolution', justinu
05:33.51P-NuThas anyone compiled mpg123 on ubuntu dapper?
05:33.52P-NuTIt's not working for em..
05:33.59justinu~seen r_evolution
05:34.06jbotr_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 ;)'.
05:34.06drrayI think monkey with a machine gun would make a fine tv show
05:34.15FuriousGeorgeshoot anything with a monkey...
05:34.22justinusheehs, i really hope that's not what happened to him
05:34.33FuriousGeorgei'd make a show called "the monkies"...   unless someone's done it
05:34.40spacklehehe
05:34.49spacklehay hay we're the monkeys
05:34.54drrayeven if they had, you could just spell it different
05:35.02benjamin7062justinu, think positive.. maybe he's kinky
05:35.06FuriousGeorgesettle down there daydream believer :)
05:35.26justinubenjamin7062: lol, problem is i know him well enough to be concerned
05:35.30benjamin7062Stop!  In the name of love!
05:35.56benjamin7062justinu, Yuk
05:36.02FuriousGeorgemy tunk, my trnk, my lovely asterisk trunk (check-it-out)
05:36.12FuriousGeorgethat never gets old
05:36.15justinueither way, he'll be ok... i just hope it doesn't take years to find out
05:36.49benjamin7062lets talk about pbx baby.. lets talk about you and me... lets talk about all the good things and the.....
05:37.06justinubenj, your robotic voice problem
05:37.08spackleugh
05:37.13FuriousGeorgei treat it very nice-y, config my SIP device-y
05:37.22justinucheck the RTP packetization value of your SIP devices
05:37.24justinushould be 20ms
05:37.44justinuanything else will make weird shit happen, like what you describe
05:37.52FuriousGeorgewachagonna do with all that zap, all that zap up in your trunk
05:37.55benjamin7062my lighter quit working... Now I can't sit here and smoke... I don't think I can stay at work any longer
05:37.57FuriousGeorgei'll stop
05:38.05justinubenjamin7062: you can smoke at work?
05:38.10benjamin7062justinu, yes
05:38.17FuriousGeorgei work from home too :)
05:38.21justinui can too, but only because I work from home
05:38.29benjamin7062no, I work from an office.. we just have smoking
05:38.34justinuweird, where's that?
05:38.38drray1987
05:38.41FuriousGeorgemexico?
05:38.42benjamin7062Photodex
05:38.48benjamin7062Austin, TX
05:38.51FuriousGeorgephotomex?
05:38.55justinuwow... texas, land of the free
05:38.58drrayyou can smoke in austin?
05:39.02drrayI grew up there,
05:39.02benjamin7062no
05:39.06benjamin7062but in Westlake
05:39.07benjamin7062you can
05:39.10benjamin7062=)
05:39.14justinui hear austin is like, the only nice city in tx
05:39.18benjamin7062our company moved because Austin changed there law last year
05:39.19spackleanybody use fxotune?
05:39.26drrayaustin is like the spec on top of the big pile of shit
05:39.31justinuheh
05:39.33drrayit's the nicest part
05:39.35justinuso i hear
05:39.45benjamin7062justinu, after katrina... it is... They shipped Lousianna to TX, and well, there was a different kind of people there.
05:40.02benjamin7062Most didn't come to Austin, thank god
05:40.04benjamin7062but some did
05:40.17drrayMy mom is lamenting here refugees
05:40.19cryptnixhmm, seeing 6 TB of ram and 119 processors ... then another shipment of the same the next day is quite invigorating
05:41.18benjamin7062drray, 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
05:41.21benjamin7062what more could you want
05:41.23benjamin7062=)
05:41.28justinubenjamin7062: your mistake was letting your stog go out before lighting another off i
05:41.29justinuit
05:41.41spackletwo guns?
05:41.46drrayI spent 26 years inAustin
05:41.50benjamin7062justinu, You're right about that.. but I didn't know I was going to lose a  lighter
05:41.52drrayI go back every spring
05:41.57justinui think they should stop executing people
05:42.07justinumy biggest beef with texas
05:42.18benjamin7062if someone killed your mother for fun.. you'd think otherwise
05:42.23justinuperhaps
05:42.25drrayno
05:42.35drraybecause it would not bring her back
05:42.36justinui can't say, because it hasn't happened
05:43.12benjamin7062At least here if you 'catch' them.. you can shoot them and it's legal
05:43.29justinuno issues with that
05:43.43drrayI believe you can do that anywhere
05:43.57benjamin7062Not 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.
05:44.04justinuit doesn't
05:44.06justinuthat's the problem
05:44.30benjamin7062It would if we executed in 2 months... EVERYONE that deserved it.. but we don't... there are hardly executions
05:44.40benjamin7062It's not a 'fear' factor at all
05:44.41justinuanyways, this is a slippery slope that we shouldn't get into
05:45.00drraywell, I vote we don't do it at all
05:45.12justinuagreed, and that's all i have to say about that :)
05:45.32benjamin7062true... 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)
05:46.02benjamin7062As long as you guys aren't white... stupid honkeys
05:46.04benjamin7062err wait
05:46.05benjamin7062I'm white
05:46.06benjamin7062damn
05:46.22benjamin7062Guess I'm not racist after all
05:46.36drraymisanthropy is not racism
05:46.50spacklefxotune?  anyone?
05:48.34benjamin7062hell yes... boss had lighter fluid and I had a spare zippo around here
05:48.35benjamin7062hawt
05:48.45RoyK[se]morning.....
05:48.48justinunight guys
05:48.57benjamin7062night
05:53.30benjamin7062hey.. 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?
05:53.32benjamin7062is that true?
05:54.38drrayyou are speaking of direct references?
05:55.42benjamin7062Well, 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?
05:55.44benjamin7062right?
05:55.46benjamin7062or wrong?
05:56.22benjamin7062err dialout 'includes' don't apply
06:12.45SwK[top] include=> next   [next] include => third [third] exten => foo
06:12.55SwKfoo is accessable from [top]
06:15.51benjamin7062SwK, 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)
06:16.06benjamin7062with a lower priority on match... but it 'is' accessible
06:16.08benjamin7062kewl
06:24.54*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
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)
06:29.23*** join/#asterisk littlejohn (n=little@host20-61.pool8711.interbusiness.it)
06:29.37*** part/#asterisk littlejohn (n=little@host20-61.pool8711.interbusiness.it)
06:30.29P-NuTHey all, moh-native for music on hold.
06:30.39P-NuTwhat are the files?
06:30.41P-NuTmp3's?
06:30.43P-NuTpcm?
06:38.14*** join/#asterisk n3glv (n=Omega__@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net)
06:38.22kmilitzerP-NuT: As far as I know it's mp3. At least that's the format that is coming with the asterisk-sources ...
06:39.10n3glvrehi all
06:40.30n3glvu taling abt MOH?
06:40.36n3glvtalking
06:42.56n3glvhey, where is controlled the console behavior? the actvity that scrolls up the screen on the console screen when not logged in?
06:43.02jaybanyone know any wholesale australian providers with great mobile rates ?
06:43.26n3glvtheres a few aussie sites listed in asteriskguru's list jayb
06:43.56jaybis that http://www.asteriskguru.com/ ??
06:44.04n3glvyes sir
06:44.17n3glvdidn't pay them much attn, since I am in usa
06:44.27jaybhehe no probs :)
06:44.27jaybthanks
06:44.38n3glvjust found out that my provider only allows two pstn sessions at a time
06:44.40n3glv:-(
06:44.44*** join/#asterisk zepmantra (i=mantra@203.215.100.96)
06:44.46n3glvbummer
06:45.01n3glvany combo of two, out or in etc.
06:45.04benjamin7062PRI it...
06:45.07benjamin7062=)
06:45.13n3glvyeah, for home use...
06:45.26benjamin7062oh
06:45.26n3glvwould prob be ok with 4
06:45.27benjamin7062ouch
06:45.30n3glv2 sucks
06:45.37benjamin7062Get 2 dif. accounts
06:45.39benjamin7062=)
06:45.43n3glvyeah, looking at that
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06:45.57n3glvif I buy a DID somewhere, that go's ip to my box yes?
06:46.06benjamin7062yup
06:46.17n3glvand then depending on their system can do x number of inboubnds?
06:46.18benjamin7062if you buy it from a sip provider that is
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06:46.32benjamin7062yes, that depends on their system
06:46.53n3glvlike I said, could use like 2 more leggs in
06:46.58benjamin7062technically, they can feed you and number of PSTN channels that they want.
06:47.08n3glvany number
06:47.14benjamin7062if yours allows 2... 1 more would work
06:47.22n3glvwell, mine has fail over
06:47.24*** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)
06:47.26benjamin7062just get another account with same provider.
06:47.34n3glvso if I put the 2nd DID as my fail over.....
06:47.44Tilii 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
06:48.06n3glvif I get a 2nd DID from same provider, it depends on them if it's separate from the 2 count?
06:48.26benjamin7062n3glv, yes... the provider controlls what you can/can't do.. definately not *
06:48.33n3glvyeah
06:48.39benjamin7062* will support whatever they will feed you
06:48.42n3glvcause axvoice we did like 11 legs, testing
06:48.49n3glvthis one (viatalk) does 2
06:49.22n3glvso, I think I need a prepay that allows free incoming
06:49.28n3glvany suggestions?
06:49.56n3glvI can set my failover to the prepay incoming
06:50.03benjamin7062have you tried http://www.voipdiscount.com/en/index.html
06:50.07*** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)
06:51.12n3glvjust went there, are they a listing service?
06:51.16n3glvlike pricewatch?
06:51.37benjamin7062no
06:51.40benjamin7062they provide service
06:51.41n3glvol
06:51.44benjamin7062but I just checked...
06:51.44n3glvooops ok
06:51.48benjamin7062may not be USA
06:51.53benjamin7062or maybe that is just there free service
06:52.05benjamin7062I'm pretty sure that is one of the ones I checked and they did USA but I could be crazy
06:52.40n3glvyes usa is there
06:52.45*** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169)
06:53.15muppetmasterSo, 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.
06:53.16n3glvsince 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)
06:53.31muppetmasterBut nothing seems to compile...  Is this just the state of things, or is there something 'special' I should be taking into consideration?
06:54.10n3glvso, u guys think someone could load * onto say a HP-Ipac running linux?
06:54.23n3glvwould be a palm top pbx
06:54.46benjamin7062n3glv, might be hard to get 'device' management working
06:54.46n3glvuse some wifi phones and make a hell of a demo box for sales
06:55.00n3glvjust wifi sip phones
06:55.11benjamin7062ie... don't know how linux runs on a HP Ipac
06:55.26n3glvneither do I but it would be wild
06:55.31muppetmastern3glv We are doing exactly that, with Asterisk inside of a VMWare session creating an ad-hoc WiFi network and then using WiFi SIP Phones.
06:55.34muppetmasterWorks great
06:55.44n3glvjack into a clients lan with a cable and hand em a couple wifi handsets
06:55.59Zeeekmuppetmaster how was LOndon ast?
06:56.02n3glvmuppet:cool
06:56.03benjamin7062The VMWARE thing works 'excellent'
06:56.18n3glvvmwarea on a M$ box?
06:56.23muppetmasterZeeek Was great, except that yesterday the room was an absolute icebox, think everyone will get sick.
06:56.23jaybbenjamin -- what vmware thing ??
06:56.36jaybben - im interested in * in VMWare also... please share
06:56.52Zeeekmuppetmaster it's probably the only time they'll have seen "weather"
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06:56.59benjamin7062jayb, Running * in vmware works just fine so long as you get your trunking for PSTN via sip provider
06:57.25muppetmasterbenjamin7062 Using bridged networking I have had no problem getting SIP trunks to work with a PSTN interconnect provider
06:57.27n3glvsince there is a vmware for M$ I always ask
06:57.36benjamin7062I ran 10 phones for demoing using VMWare
06:57.59benjamin7062muppetmaster, well, my point being that you can't use 'hardware' for PSTN...ie, trunk cards through vmware
06:58.03muppetmasterSo, anyone have any ideas on the state of trunk?  Zaptel, Asterisk & Asterisk-addons all crap when compiling.
06:58.33muppetmasterbenjamin7062 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.
06:58.35n3glvI am not a coder, does anyone know a wiki or url to upgrade my centos SMP kern? (trixbox zaptel broken)
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06:59.34benjamin7062~centos
06:59.41jbotcentos is probably better than Fedora Core except for that silly bug, see ~centosbug for details
06:59.53benjamin7062~centosbot
06:59.57benjamin7062~centosbug
06:59.58jbotextra, 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"
06:59.58benjamin7062damnit
07:00.01n3glvI just threw hardware at it to get some juice out of it on single kern
07:00.17n3glvI did that paste and it ignored me
07:00.23benjamin7062as root?
07:00.27benjamin7062hmm?
07:00.33n3glvthen it puked on the fact that the dir name is diff on the kern
07:00.43n3glvseems to be the issue
07:01.15benjamin7062well, if you have a custom location for some of the above referenced stuff... you may need to modify as necessary
07:01.17n3glvthat actually may be the problem loading the module
07:01.25n3glvI tihink it's looking in wrong dir for it
07:02.25benjamin7062do some of the escape commands and see what they do on your system.. ie `uname -r`
07:02.29benjamin7062uname - m
07:02.38benjamin7062etc
07:02.43benjamin7062make sure the kernels exist where they are suppose to be
07:02.45n3glvmy kern dir is called 2.6.9-34.EL-smp-i686
07:03.07benjamin7062what does uname -r  and uname -m return
07:03.07n3glvand I think it's trying to load from 2.6.9-34.ELsmp
07:03.18benjamin7062ahh
07:03.28n3glvso, the .mo is not found
07:03.46n3glvon startup (I think that is what I am reading)
07:04.07n3glvshould I see the module file in the kern dir?
07:04.20benjamin7062do vi /path/to/spinlock.h   then type :.,$ s/rw_lock/rwlock/i
07:05.01x86re
07:05.02n3glvcan we see if the module exsists but is in wrong dir?
07:05.28tzafrirbenjamin: why not give a sed -i / ex command and be done with it?
07:06.14benjamin7062tzafrir, either way
07:06.46tzafrirvi /path/to/spinlock.h +'%s/rw_lock/rwlock/' +x
07:07.12tzafrir(Though IIRC original vi did not support multiple + commands)
07:07.22benjamin7062tzafrir, I personally rather be in the file to see what changes and backout if necessary
07:07.38tzafrirn3glv, ls is known to be handy
07:07.45benjamin7062lol
07:07.45tzafrirfind/locate also
07:08.01benjamin7062and updatedb <-- or your boxens version
07:08.10*** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net)
07:08.22tzafrirupdatedb should be run nightly
07:08.43tzafrirand locate's db is hopefully a good approximation
07:09.20benjamin7062that 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
07:10.36benjamin7062muppetmaster, 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
07:11.37stoffellIf 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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07:14.05littleballhello, i don't understand how "hint" priority works. who can help?
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07:17.56dangerareamorning
07:18.23dangerareai'm still having issues with ztdummy and my usb-uhci modules
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07:34.02yacyachey
07:34.05yacyacanyone around ?
07:35.27Zeeekmillions of us!
07:35.35yacyachahaha
07:35.47Zeeekwell, actually 241 lurkers
07:35.50x86TRILLIONS HAHAHAHA ROFLMAO KTHX OMG
07:35.57yacyacZeeek... dude... which distro is the best for running asterisk
07:36.01yacyachahahaha
07:36.06yacyachey x86
07:36.09x86heya
07:36.11yacyacany suggestion guys
07:36.20littleballhello, i don't understand how "hint" priority works. who can help?
07:36.27maskedLFS
07:36.28x86yacyac: any distro will run asterisk fairly well, at least on x86 or x86-64 hardware
07:36.47x86yacyac: stay away from the BSD's and yellow dog linux, etc
07:37.44yacyachahaha
07:37.45yacyacok
07:37.51yacyaci was thinking about freebsd
07:37.52yacyaclol
07:38.03yacyaci guess i will need to kick it out of my mind then
07:38.04yacyaclol
07:38.09yacyacwhich one do you guys use
07:38.33[hC]I use debian.
07:38.49stoffelldebian also.. 2 votes.. ;)
07:39.00x86i use gentoo and ubuntu
07:39.03[hC]but, i compile asterisk from source..
07:39.08[hC]i dont use debian packages.
07:39.09yacyachow about arch
07:39.14x86yeah, you gotta compile asterisk from source :)
07:39.24[hC]plain x86
07:39.27x86yacyac: stick with x86 if you can, x86-64 if you must
07:39.37yacyaci have amd 64 machin
07:40.01yacyacso which is good 64bit edition of linux ?
07:40.15maskedgentoo?
07:40.33maskedor ubuntu?
07:40.38yacyaci dont like gentoo and ubuntu
07:40.48maskedneither really
07:40.50Zeeekslackware
07:40.50maskeduse LFS
07:40.57maskedyeah or slack!
07:40.57yacyacarchlinux ?
07:41.02yacyacbrb.... lunch time
07:41.02yacyaclol
07:41.42[hC]any of you guys play with hudlite?
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07:54.35X-Rob_you stole the match!
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08:05.37littleballhello, i don't understand how "hint" priority works. who can help?
08:05.48nettieHi, 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)
08:16.58FreezeShello
08:17.20FreezeSdo you know how can I make MoH to start from the beginning for every call ?
08:19.19nettieFreezeS
08:19.19nettieexten => 7601,1,Answer
08:19.19nettieexten => 7601,2,SetMusicOnHold(default)
08:19.20nettieexten => 7601,3,WaitMusicOnHold(3000)
08:20.12benjamin7062Wouldn't you just do MusicOnHold instead of making 'em wait 3000 seconds?
08:20.20benjamin7062Ouch
08:21.40drrayor just play an audio file
08:21.57FreezeSsorry for being so imprecise. How can I do it for a queue ?
08:23.55nettiebenjamin7062 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)
08:27.02Zeeekcase dismissed!
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08:28.18yacyacZeeek dude
08:28.19yacyaci am back
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08:31.49Zeeekhey now
08:32.07Zeeekwhat was the question? (I have a life as well and it interrupted my IRC career)
08:32.23giesenman
08:32.36giesenI just did some of the dirtiest crap to my dialplan that you have ever seen
08:32.57ZeeekI would win the trohy for the dirtiest dialplan hands down
08:33.06Zeeeks/trohy/trophy/
08:33.14giesenall in the name of setting caller id
08:33.20yacyachahaha
08:33.34giesenwell, it's a contextual caller id
08:33.58giesenso, if it's from an internal extension to one of our employees cell phones
08:34.09giesenit's set to 000000<ext>
08:34.24giesenit's a dirty, dirty hack.
08:34.26*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
08:34.41yacyachow much is the cost of the hardware used in asterisk
08:34.59giesenhaha the best is when I set the caller id for a single extension twice
08:35.08giesenyacyac: depends what you want to plug it into
08:35.23giesena single port analog fxo card is only like $20
08:35.31yacyacwhat are my options
08:35.33giesenbut the pri cards can get pretty pricey
08:35.50yacyacwhat does pri cards do
08:35.53giesenother than that, it's commodity pc hardware
08:36.05giesenyacyac: if you dont know, then you dont need it
08:36.18giesenif you just want to plug an asterisk box into an analog phone line
08:36.25giesenall you need is a single port analog fxo card
08:36.25yacyacis it pstn to voip adapter ?
08:36.39yacyacwhat does single port analog fxo card do
08:37.02giesenjust allows you to plug your asterisk box into a plain jane analog phone line
08:37.23giesenif you dont even need that much
08:37.35giesenthen all you really need is an ethernet card
08:37.54giesenin a standard pc
08:38.05yacyacasterisk handles incomming calls and fwds it ?
08:38.38giesenif you want to handle incoming (or make outgoing) calls on an analog phone line, you need an analog fxo card
08:38.46yacyacok
08:38.48yacyackool
08:39.00giesenif you want to do voip exclusively
08:39.03giesenthen you dont need one
08:39.18yacyacthat will handle 1 call at a time ?
08:39.28giesenassuming you either have IP phones or ATA (Analog Telephone Adapters) for your existing analog phones
08:39.32giesenyacyac: yes
08:39.58Zeeekyacyac what exactly do you want to do?
08:40.17Zeeekand in what country
08:40.25yacyaci am in india
08:40.39yacyaci want to recive and make calls
08:40.43Zeeekand you want to call/receive calls from where?
08:40.50yacyacrecive and fwd to some other local number
08:40.52yacyacsomething like that
08:40.56dangerareacan anyone help with this please?
08:40.57dangerareahttp://pastebin.com/734452
08:41.03yacyacall around the world
08:41.03yacyaclol
08:41.09Zeeekrceive on a normal POTS phone line or a new number?
08:41.48yacyacwhat is pots phone line
08:42.00giesenpots = plain old telephone system
08:42.05giesenbasic analog phone line
08:42.05dangerareayacyac: a normal analog line
08:42.11yacyacyes
08:42.21yacyaci want to transfer to pots
08:42.43giesenthe question is do you want to receive calls on your existing pots line
08:42.49yacyacyes
08:42.50giesenor are you fine with getting a new phone number
08:43.10yacyacyes
08:43.24yacyacnew phone number from the telephone company ?
08:43.34Zeeekso you need a cheap X100P (used or clone)
08:43.38giesenwell if you went for the second option
08:43.45giesenyou would get the new number from your voip provider
08:43.52Zeeekthat connects to one POTS phone line
08:44.07yacyacwe dont have any voip provider here i guess
08:44.18yacyacwhat if i want to do that
08:44.18giesenthere's definitely voip providers operating in india
08:44.19Zeeekfor foreign activity, I'd recommend getting an account in a country where you call
08:44.43Zeeekor, get Skype ;)
08:44.51yacyacwhat if i dont want to involve voip providers then
08:44.59yacyacplain funda
08:45.04yacyacone asterisk server here
08:45.07Zeeekyacyac is this business or pleasure if I may be so indiscreet?
08:45.27yacyacone asterisk sever any country
08:45.35yacyacand i just link them uo
08:45.38yacyacup*
08:45.42giesenyacyac: you can do that, yes
08:46.06*** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au)
08:46.19yacyacso from india i can call to pstn line of any other country where my box is
08:46.40yacyacand anyone can call to my pstn line my box
08:46.54Zeeekif that box has an FXO and a line or a voIP provider, yes
08:47.26yacyacsomething i wanna try out
08:47.51giesenyacyac: it's probably much more cost effective to get a voip provider in the country you want to call
08:47.59giesenthan to setup an asterisk server in every country
08:48.06yacyachmmmm
08:48.12yacyacwhat about in long run
08:48.19yacyacwhich is more good
08:48.45yacyacskype and other voip providers are good... how cheap can they go in bulk calling
08:48.59giesenI pay 1.1c/minute for calls
08:49.16yacyackool
08:49.19giesenwithin my local calling area
08:49.27giesenor 2c/minute for anywhere in north america
08:49.42yacyacgiesen ... whats the use of using asterisk if we use skype of anythinng
08:49.55yacyackool
08:49.57giesenwhat's the use of any pbx
08:49.58yacyacvonage is good
08:50.16giesenallows you to share resources and perform cool stuff
08:50.19yacyacgiesen ... sorry but i am new.. and my concepts are not very clear
08:50.24yacyacthansk
08:50.50giesena bunch of people on a pbx can talk between eachother without consuming any resources, they can share phone lines, etc
08:51.17yacyachmmm
08:51.17giesenyou can have 100 people sharing 10 phone lines
08:51.23giesenrather than have one for each person
08:51.23yacyackool
08:51.29yacyacok
08:51.54giesenhaha unless you work at my company
08:51.54*** join/#asterisk gan- (n=nmuller@195.70.21.58)
08:51.59yacyachahahahaha
08:52.00giesenin which case we need 2 phone lines for every person
08:52.00yacyaclol
08:52.04yacyacwhy
08:52.15giesenwe spend a lot of time on conference calls, etc
08:52.23yacyachahaha
08:52.26giesenso we're sometimes talking to 5 people at a time
08:52.31yacyacwow
08:52.32yacyaclol
08:53.15yacyacso you have a asterisk box up and running ?
08:53.38*** join/#asterisk oscarh (n=oscar@host-87-74-0-243.bulldogdsl.com)
08:53.58giesenyep
08:54.01gan-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?
08:54.48yacyackool giesen
08:54.49giesenyacyac: I actually have 2 setup, in an active-standby cluster
08:54.55yacyachome or office
08:54.57yacyackool
08:55.00oscarhhi, 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 :(
08:55.09yacyacwhat do you do with 2 then
08:55.22giesenone is there in case the other one fails
08:55.32yacyacwow
08:55.32giesenso if one dies, the other takes over
08:55.36yacyackoool
08:55.47yacyacwhat 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?
08:55.49giesenand I can take one down for hardware upgrades, etc, without taking down my phone system
08:56.00giesenyacyac: company pbx
08:56.07yacyacoh
08:56.09yacyackoooll
08:56.29*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
08:56.40yacyacso which voip provider do you use
08:57.08*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
08:57.10yacyaci have heard alot about vonafe
08:57.13yacyacvonage
08:57.26benjamin7062can * work with vonage?
08:58.04giesenyacyac: unlimitel.ca
08:58.15giesenbenjamin7062: I think there's some hacks to make it work
08:58.20*** join/#asterisk creadurx (n=creadure@196.82-134-19.bkkb.no)
08:58.21giesenthough I've never tried
08:58.35yacyackool
08:59.03yacyacand is there any setup cost for setting up a voip account with them ?
08:59.15giesencheck the pricing page
08:59.31giesenfor the setup I got, it was a minimum $50 to start
08:59.45giesenbut I got the a la carte package
08:59.49giesenwhere I pay per minute
08:59.58giesenrather than a flat rate per line
09:00.19yacyacoh
09:00.19yacyacok
09:01.45yacyacgiesen which distro have you used for asterisk
09:01.55giesengentoo
09:02.22benjamin7062They do support * (or provide ATA rather) for bus customers starting at $150
09:02.54giesenbenjamin7062: either that or you can just use the ata they give you, can get fxo cards for your asterisk box
09:03.03giesenand use single line analog trunks between them
09:03.06giesenit's kinda dirty
09:03.08giesenbut it works
09:03.24yacyacthanks a lot giesen
09:03.29giesenno problem
09:03.42yacyacany recommandation on using any specific distro
09:03.45benjamin7062yeah... but that's .. well, bleh
09:03.47benjamin7062=)
09:03.56giesenhey, I said it was dirty =)
09:04.00benjamin7062I know -- teasing
09:04.04yacyacgentoo... i dont like it much.... never was able to hold it for long on my machin
09:04.04giesenalmost as dirty as my dialplan
09:04.17giesenyacyac: have you tried asterisk@home?
09:04.22giesenit may be to your liking
09:04.29giesenautomates some of the tough stuff in asterisk
09:04.34giesenwith a point and click interface
09:04.44benjamin7062yacyac, I have excellent success with debian testing (etch)
09:04.47giesenit's based off of centos
09:04.47benjamin70622.6 kernel
09:04.57yacyacoh
09:04.57yacyacok
09:05.00benjamin7062trixbox is also something to play with
09:05.09giesentrixbox = asterisk@home
09:05.11benjamin7062(which is asterisk@home)
09:05.16yacyaci have used debian .. it was too heavy on my laptop
09:05.17yacyaclol
09:05.23yacyacso i used arch
09:05.25benjamin7062too heavy?
09:05.28giesenall depends on what you install
09:05.30yacyacnow i am just on this windows
09:05.30yacyaclol
09:05.37yacyacyeah i had only 256mb ram
09:05.47yacyacand i wanted to run gnome
09:05.47yacyaclol
09:06.01yacyacarch was pretty fast
09:06.05*** join/#asterisk bmg505 (n=leon@c1-199-1.rndf.isadsl.co.za)
09:06.16giesenbenjamin7062: you have a decent enum macro?
09:06.30giesenI had to hack the one on voip-info.org up a bit to get it to work reliably
09:07.02yacyacgiesen is it relaibly to use asterisk at corporate level ?
09:07.11benjamin7062yacyac, my X session is 160megs right now pushing 6 screens (3 vid cards)  ... just fyi
09:07.29yacyacbenjamin7062 what do you run boy
09:08.01giesenyacyac: lots of people do
09:08.06*** join/#asterisk n3glv (n=Omega__@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net)
09:08.10giesenif you know what you're doing
09:08.15yacyackool
09:08.26giesenI still have a few kinks Im working out in my system
09:08.30giesenbut it's definitely getting there
09:08.37yacyacgiesen: i am not a pro or something... but i just try to learn
09:08.45yacyackool
09:08.50yacyacwhat are you trying this time
09:09.12giesenI've still got some bugs I have to work out with my music on hold and conference calling
09:09.23yacyackool
09:09.30giesenand I dont like the way asterisk does queue
09:09.32giesen*queues
09:09.35yacyacwhy
09:09.57giesenbecause of the way it handles penalties
09:10.05yacyachmmmmm
09:10.28giesenI had to use cascading queues
09:10.40giesento get "priority" levels to work the way I want
09:10.40yacyacwhat is cascading queues
09:10.45yacyacoh
09:10.46yacyackool
09:10.50giesenfor example
09:11.03giesenIve got a couple employees who are the front line tech support guys
09:11.07benjamin7062giesen, for us, we aren't doing ENUM
09:11.11yacyackool
09:11.11benjamin7062inbound call center mostly
09:11.17giesenbut if they dont pick up the phone
09:11.24yacyacoh yes
09:11.25yacyaci know
09:11.25giesenI want it to fail over to some other poeple
09:11.35giesenbut asterisk doesnt do it in a useful fashion
09:11.37giesenat least not for me
09:12.02giesenso I actually had to create 3 different queues
09:12.16benjamin7062giesen, I had the same issue, had to use a combination of Queue(|||timeout) with maxlen, and strategy strict... along with looping in the dialplan
09:12.18giesenand timeout from one queue to the next
09:12.49giesenI think I've got a workable solution now
09:12.53yacyacif 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
09:12.57giesenwith cascading queues
09:13.19giesenyacyac: be prepared to spend a LOT of time getting your first asterisk server going
09:13.25yacyacyeah
09:13.31yacyaci know that lol
09:13.45bmg505Hi guys, any of the dev's here?
09:13.47benjamin7062giesen, We ended up writing a server daemon that simply checks the queue status and uses the manager API to add second level agents... HAX!  =)
09:14.02giesenhaha
09:14.04giesennice
09:14.08giesenI dont have the resources to do that
09:14.26giesenand the queue/agent penalties were useless to me
09:14.33yacyacinternational call center in india use's one of the best queueing system
09:14.48yacyacyou will never see a call drop in it
09:14.51benjamin7062yacyac, 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.
09:15.23giesenI pity the fool who has to try and figure out what the hell my dialplan is doing
09:15.26yacyacbenjamin7062:  then its no fun .. if they do it for me....
09:15.31bmg505I 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?
09:15.43benjamin7062yacyac, trust me... 2 hours isn't enough to do 'shite' for you.  =)
09:15.44bmg505MixMonitor for queues
09:15.50yacyachahahahaha
09:16.01giesenyacyac: be prepared to spend like 20 or more hours
09:16.05giesenfiguring stuff out
09:16.15bmg505giesen, only 20 hours?
09:16.19giesenhehehe
09:16.24yacyacbenjamin7062 lets see... first let me get thing started... if i am fucked i will catch hold of them
09:16.25giesenIve probably spent about 150 so far on mine
09:16.25benjamin7062giesen, Problem is... if you do cascading queues like that.. it breaks hold times I think... avg hold times.. etc.. the stats.
09:16.31yacyachahaha
09:16.31giesenyeah
09:16.36yacyactime is no problem
09:16.37yacyaclol
09:16.42giesenbenjamin7062: it's not that important for me right now
09:16.49giesento have hold time stats
09:16.53giesenbut it will be in the future
09:17.22benjamin7062giesen, the server daemon is easy if you know perl... their manager interface is EASY.. clear text ascii with \r\n terminators
09:17.30giesenyeah
09:17.37giesenit sounds like a cool way of doing it
09:18.03benjamin7062but I honestly can't confirm if using queue timeouts in fact DOES break avg hold times
09:18.18benjamin7062it might build those by channel timers rather than queue timers.
09:18.33giesenactually, that would be very cool if it did use channel timers
09:18.36giesenbut I doubt it
09:18.39stoffellbenjamin7062, a queue timeout means you bail out of the queue, so you go somewhere else... end of call to that queue
09:19.06benjamin7062stoffell, that's what I assumed.. so I just went the daemon route instead of figuring it out
09:19.08benjamin7062=)
09:19.11giesenstoffell: yeah, which is a problem when you use cascading qeues
09:19.23giesensince the timers are probably reset every time you change queues
09:19.42stoffellgiesen, and (out of interest) cascading queues, would be useful to do ... what ?
09:19.47benjamin7062I suspect spillover 'stuff' will be coming soon
09:20.02giesenstoffell I have different tiers of tech support staff
09:20.08giesenso if my front-line guys are too busy to pick up
09:20.11giesenor unavailable
09:20.11benjamin7062stoffell -- high priority agents that are only bugged if queue 1 is full
09:20.13stoffellgiesen, yes, they are.. changing queue is going out of 1 and going into a new
09:20.18giesenit fails over to some other people
09:20.55benjamin7062we had the same issue
09:21.09stoffellso if q1 (first line) is full, the calls go to q2 (2nd line support)... ? so that's correct?
09:21.15giesenyeah
09:21.18Zeeekmaybe 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
09:21.18benjamin7062yeah
09:21.18yacyac;kool
09:21.26giesenfull or unavailable, etc.
09:21.43benjamin7062Zeeek, that's what I wrote instead of doing a tiered queuing structure.
09:21.48stoffellgiesen, but the time doesn't get reset then, does it ? it goes straight to q2 ??
09:21.51benjamin7062because of the hold time situation
09:22.04giesenstoffell: the first queue times out, then it goes into a brand new queue
09:22.08giesenI dont know if it gets reset or not
09:22.12giesenI havent tested it yet
09:22.12benjamin7062stoffell, you have to inject a call to queue 1 first to check if someone is available
09:22.16giesenbut I would suspect it does
09:22.43stoffellbenjamin7062, yes, correct, but it takes 1 second to see if it its, if it's full, it goes to q2 ...
09:22.47benjamin7062right now there is now application or global var that gives status on QUEUE's... so you can't check prior to injection
09:22.49stoffellif it is ...
09:22.56Zeeekbenjamin7062 so where do we download it? :) Or is this the dirty dialplan?
09:23.06benjamin7062stoffell -- you can't use 1 sec.. because the timeout includes call termination
09:23.20stoffellbenjamin7062, ah, yes, indeed.. very correct... good point.. (on having an overall queue status)
09:23.43giesenI actually like benjamin7062's idea of using the manager interface to do it
09:23.44benjamin7062You have to leave the call in the queue long enough to 'try' extensions...
09:23.45stoffellbenjamin7062, but you can say in queues; if full (limited callers), go to q2..
09:23.49giesenyou might also be able to do it with an AGI
09:23.57benjamin7062what I did, since you can't set maxlen to 0.. is set it to 1 instead
09:24.19benjamin7062then.. at least... you can't put a call in that queue if someone is already waiting.. so it jumps to 2
09:24.22benjamin7062immediately
09:24.30stoffellan agi could do it, you could poll the queue and the available agents.. but it's some work/debugging
09:24.30benjamin7062and only 1 caller gets screwed with a wait time
09:24.31benjamin7062=)
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)
09:25.37benjamin7062the 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
09:25.41benjamin7062but it's hax
09:25.41stoffellbenjamin7062, waiting is 'good', it's a queue! ;)
09:26.20benjamin7062stoffell, 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
09:26.24benjamin7062it ends up looping
09:27.02stoffellyep, that's right.. can't do it easily right now.. good idea though.. could use it here myself.. :)
09:27.10benjamin7062Basically, 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
09:28.48benjamin7062BTW -- did I mention I LOVE the polycom 601's?
09:28.49benjamin7062=)
09:29.04yacyacwhat is polycom 601's
09:29.15dlynes_officebenjamin7062: spread the love...donate some polycom 601's to a good cause :)
09:29.37Zeeekyacyac phones you can't afford
09:29.40*** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net)
09:29.43Bert-hello there
09:29.50yacyacZeeek and they are
09:30.03ZeeekVery nice SIP phones
09:30.17benjamin7062We 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
09:30.22Bert-I have this card : VIC2-2BRI-NT/TE (cisco). It is a 2 port card. Txo ports means 2 lignes or 2x2lines plz ?
09:30.47Zeeekbenjamin7062 tell us about manager<--> Polycom
09:30.52Zeeek601
09:31.11yacyachey all cards and everything works in india ?
09:31.14drraythe polycom's are ugly
09:31.16drrayIMO
09:31.23Zeeekget out!
09:31.25benjamin7062Zeeek, sky is the limit... write a CGI that converts manager API to miniture version of xhtml... get 'whatever' you want
09:31.30dlynes_officeyacyac: i would imagine so, yeah...they work in Lahore
09:31.36yacyaci had heard that most of the card cant
09:31.45yacyacdlynes_office : where you from
09:31.51benjamin7062for 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
09:31.51dlynes_officeyacyac: 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)
09:32.09yacyackool
09:32.11dlynes_officeyacyac: I'm from vancouver, canada, but there's two or three guys on here from Pakistan
09:32.18giesenbenjamin7062: that sounds really cool
09:32.23yacyackool
09:32.24giesenI may do something like that for my cisco phones
09:32.40dlynes_officeyacyac: you're setting up routes in India?
09:32.41giesenso I can manage some stuff that I can do with the SIP loads on them
09:32.42benjamin7062Zeeek, i have the 301.. also, same quality... but obviously lacking functionality
09:32.52giesen*cant do
09:33.04Zeeekyep, sad ain't it? We're in the ghetto of Polycom low rent
09:33.07yacyacyep ... setting up a asterisk server in bombay
09:33.10giesenand chan_sccp is just dirty
09:33.21benjamin7062giesen, 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!
09:33.27dlynes_officeyacyac: ah...will you be doing anything in Punjab/Amritsar?
09:33.34Zeeekyacyac so the club would be basterisk ? (like bollywood)
09:33.43giesenbenjamin7062: yeah, that's part of the reason I wont touch chan_sccp
09:33.48giesenthat and the lack of dynamic reloads
09:33.49benjamin7062yup
09:33.59giesentaking down my whole box to make a change to a phone
09:34.04giesenis not my idea of a feasible system
09:34.17dlynes_officebenjamin7062: apparently chan_skinny is a lot more stable
09:34.18benjamin7062=)
09:34.33giesenchan_skinny doesnt have the functionality
09:34.36dlynes_officebenjamin7062: and if you're having problems with it, there's an active developer for chan_skinny
09:34.45dlynes_officebenjamin7062: it's Qwell/Qwell[]
09:34.50benjamin7062dlynes_home, yeah, but we tried that too... and, well, Digium was on my box when it crashed... I laughed... he sighed... we moved on
09:35.04dlynes_officecool
09:35.13benjamin7062He logged it though
09:35.14yacyacdlynes_office i dont know
09:35.16benjamin7062just a bug of some sort
09:35.34yacyacdlynes_office i have a friend who is there in amritsir who is going to help me
09:35.44dlynes_officebenjamin7062: yeah...seems those intel epro100's don't just have a problem with digium cards
09:35.55benjamin7062We decided away from the cisco due to the lack of xml support in the SIP images...
09:35.56dlynes_officebenjamin7062: their drivers seem to be buggy in general
09:36.09benjamin7062It works... but to a lesser degree per my reading of other's experiences
09:36.15dlynes_officeyacyac: can you talk to him about maybe setting up a server in Amritsar?
09:36.23yacyacsure
09:36.24yacyacwhy
09:36.24dlynes_officeyacyac: 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)
09:36.32benjamin7062Sangoma all the way...
09:36.43yacyacdlynes_office : we as in ?
09:36.43benjamin7062The EC on this Sangoma is 'perfect'
09:36.49giesenbenjamin7062: the sip loads do have xml
09:36.53giesenit's just somewhat broken
09:37.05benjamin7062giesen, yeah -- that's what I meant.  It's minimal support
09:37.06dlynes_officebenjamin7062: actually running a sangoma card on a box with two epro100's, and got a kernel trap after three hours
09:37.28benjamin7062one of the guys on here has great insite into it... they use only cisco.. he swears by 'em but acknowledges some of the bugs
09:37.30dlynes_officeyacyac: our company; we provide voip service and pstn in Vancouver, Canada
09:37.35benjamin7062xhtml is 'so' much easier to deal with
09:37.39dlynes_officeyacyac: That's close to Surrey, Canada
09:37.53yacyacdlynes_office : cool
09:37.53giesendlynes_office: you're in surrey?
09:37.54benjamin7062dlynes_home, shush, that won't happen to me
09:38.12dlynes_officegiesen: no, but we plan to start marketing in surrey to businesses shortly
09:38.16giesenah
09:38.19giesenwe're in toronto
09:38.20yacyacdlynes_office you already have a peer in bombay/mumbai ?
09:38.20dlynes_officegiesen: the owner of our company is Indian
09:38.28yacyacoh
09:38.29dlynes_officeyacyac: nope...don't need one atm, either
09:38.34yacyachahaha
09:38.43yacyacdlynes_office whats your companys name
09:38.48dlynes_officeyacyac: All of our Indian customers are from Punjab, or are of Punjabi descent
09:38.52benjamin7062Is that how everyone offers so much 'free' local calls?  Just a bunch of cooperative peers?
09:39.01dlynes_officeyacyac: and even then, 95% of them are Jutts
09:39.20dlynes_officeyacyac: 24/7 Communications
09:39.25yacyacdlynes_office: hahahhaha lol...
09:39.30yacyacgot a site to look at ?
09:39.37dlynes_officeyacyac: www.247communications.com
09:39.42yacyacdlynes_office are you an indian or canadina
09:39.52dlynes_officeyacyac: I'm Canadian; the owner is Punjabi
09:40.02dlynes_officeyacyac: but he moved here when he was 2 or 3 from Amritsar
09:40.06yacyacoh
09:40.07benjamin7062I'm mutt?  is that cool?
09:40.08yacyackool
09:40.36dlynes_officeyacyac: He's a Canadian citizen, but originally from Amritsar
09:40.37yacyacdlynes_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)
09:41.06dlynes_officeyacyac: yeah...basically we can provide local dial tone in Vancouver for you, in exchange for local dial tone in Punjab for us
09:41.37dlynes_officeyacyac: so that would even allow you to call north american toll-free numbers that are callable from Vancouver
09:41.44benjamin7062dlynes_home, is that how everyone does it?  A shit ton of peers?
09:41.47yacyackool
09:41.54dlynes_officebenjamin7062: I doubt it
09:42.00benjamin7062oh
09:42.06dlynes_officebenjamin7062: dundi
09:42.26benjamin7062I read about that
09:42.36dlynes_officebenjamin7062: dundi is for sharing your local extensions with the world
09:42.40benjamin7062basically, you agree to allow people out to the PSTN through you.. if you can go through them?
09:42.43dlynes_officeso that if anyone needs to call one of your customers
09:42.43*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)
09:42.55dlynes_officethey just dial into your switch, and dial them as a sip extension
09:43.03dlynes_officeso the call never goes to pstn
09:43.09benjamin7062ahhhhhhh
09:43.11benjamin7062GOTCHA
09:43.17dlynes_officebut that won't work for peering (pstn)
09:43.32dlynes_officethat you need to set up a manual agreement for
09:43.32benjamin7062wouldn't that be cool though.
09:45.06*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
09:46.49benjamin7062Just wait till Dundi is 'huge'...
09:47.10benjamin7062The US Gov... will attack it like napster for 'stealing' telecom from the bells.. =)
09:48.40drrayI don't get dundi
09:49.36benjamin7062Do you understand BGP?
09:49.47benjamin7062It's similar
09:50.26benjamin7062I thought it allowed PSTN sharing.. but apparently it's just peer->peer dial route propagation
09:51.17benjamin7062if 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)
09:53.56n3glvwhat's dundi?
09:54.18n3glvif the gov will hate it, I will prob love it
09:54.36n3glvis that like sipbroker? or more like ENUM?
09:54.40n3glvsounds like enum
09:56.06benjamin7062enum
09:56.08benjamin7062pretty much
09:56.13benjamin7062only distributed
09:56.24n3glvhmmm
09:56.31n3glvsounds like it's up my ally
09:56.46n3glvthey do any iax support? u got a url?
09:57.17n3glvbeen wanting to get sipbroker running
09:57.21benjamin7062http://www.dundi.com/dundi.txt
09:57.42n3glvenum never called me back to ver my number, but enum trunk works
09:57.53L|NUXfor DUNDi need peering :(
09:58.11benjamin7062true
09:58.24benjamin7062but 'anyone' can be a peer
09:58.33n3glvhmm
09:58.41L|NUXwell
09:58.44benjamin7062Not that it's the only or best way.. but it's creative
09:58.45benjamin7062=)
09:58.49n3glvthey implement dunid in  * yet?
09:58.50L|NUXif he/she allow your machine :)
09:59.02benjamin7062dundi was written by the same guy that wrote *
09:59.09n3glvcool
09:59.19n3glvthey got a wiki I wonder
09:59.19benjamin7062Mark Spencer / Digium
09:59.35AltnTabWhy Asterisk could not find recorded files from Monitor() thru external encoder executed with System()
09:59.45AltnTabthe output path is correct shoun in CLI
09:59.49benjamin7062Anyone know why my hold music sould sound like an alien sending static distortion?
09:59.51AltnTaberr shown
09:59.58benjamin7062or.. hmm, maybe that's just what it is?
10:00.07n3glvI have a dream of a Ham Radio voip that uses a peering cached callsign to ip system
10:00.51benjamin7062Ham over Ethernet?
10:00.54benjamin7062HoE?
10:01.01n3glvben, is your zaptel stuff working? ztdummy etc?
10:01.16n3glvhaha echolink.org for one
10:01.22n3glveqso for another
10:01.27Zeeekn3glv the Digium distrubutor in Paris is a ham
10:01.31n3glvbeeing done, but not very robust to failure
10:01.37NLiningtonCan somebody give me a hand getting incomming faxing working with app_rxfax. Everything looks like its working ok but faxes being received always fail.
10:01.45dlynes_officedundi and enum are almost the same thing
10:02.20dlynes_officejust basically two competing standards, from what I understand
10:02.20benjamin7062Hmm, perhaps that's my problem... I'm using a PRI... so figured timing would be handled by the card?
10:02.20benjamin7062maybe not
10:02.20ZeeekNLinington it may be a question of version. About 2 out of three fail for me
10:02.27Zeeekon X100P
10:02.37dlynes_officeNLinington: app_rxfax/app_txfax are both highly unreliable
10:02.43n3glvdo a lsmod and make sure that the zaptel dummy is there
10:02.48dlynes_officeZeeek: you actually get it to work at all on an x100p?
10:02.51n3glvI'm no expert
10:02.52dlynes_officeZeeek: you must be blessed
10:03.10n3glvbut I know MOH and Meetme and some other stuff relies on zaptel or ztdummy
10:03.20dlynes_officebenjamin7062: is your pri working fine?
10:03.38NLiningtonI have a X100P and zaptel dummy is not loaded, do I need this?
10:03.50dlynes_officeNLinington: no...don't load it
10:03.59dlynes_officeNLinington: ztdummy is if you don't have another zaptel driver loaded
10:04.00benjamin7062dlynes_office, yeah -- can make receive calls fine.. loaded ztdummy (modprobe) just to try it.. still static.
10:04.02benjamin7062poop
10:04.12benjamin7062sounds aweful
10:04.14n3glvok
10:04.17n3glvsorry
10:04.21dlynes_officebenjamin7062: yeah...you only need wct4xxp loaded or whatever...don't load ztdummy
10:04.27ZeeekMurphy'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!
10:04.30dlynes_officebenjamin7062: ztdummy will just interfere
10:04.34n3glvI am trying to compile new kernel, getting hardware errors
10:04.35benjamin7062Not your fault
10:04.45n3glvnew (used) parts here
10:05.12n3glvseems to get a bit farther each time
10:05.16n3glvso just hammering on it
10:05.21benjamin7062actually.. nothing like wct is loaded either
10:05.25n3glvmay be a ram problem
10:05.25benjamin7062perhaps that is it
10:05.34dlynes_officebenjamin7062: wait
10:05.41dlynes_officebenjamin7062: you're using sangoma not digium, right?
10:05.44NLiningtonso whats the best ways to get faxing working via *
10:05.46benjamin7062yes sir
10:05.52dlynes_officebenjamin7062: yeah...one second
10:06.04n3glvNL I think the fax machine has to be a certain spec
10:06.15n3glvsupport some protocol
10:06.20n3glvread that somewhere
10:06.27dlynes_officen3glv: 9600 baud or lower for app_?xfax.so
10:06.34benjamin7062t.136 or t.36.. or something
10:06.58n3glvyeah, knew I saw someting on that
10:07.09benjamin7062zaptel and wanpipe modules loaded
10:07.11NLiningtonbut it should handshake to that level?
10:07.12n3glvnot high priority on my box
10:07.19dlynes_officebenjamin7062: you should have af_wanpipe, wanpipe, wanrouter, zaptel, crc_ccitt, and sdladrv loaded
10:07.56dlynes_officen3glv: anyways, if you want something more reliable
10:08.07dlynes_officen3glv: use iaxmodem in conjunction with hylafax
10:08.07benjamin7062thank you for that list
10:08.10benjamin7062all okay
10:08.11n3glvit's 6am and my stomach is trying to talk me into baking a frozen pizza
10:08.27benjamin7062Timing on both PRI channels set to 'NORMAL'.. cards work
10:08.28benjamin7062hmmph
10:08.46benjamin7062the mp3 player is build into * right?
10:08.49dlynes_officebenjamin7062: wanrouter status -> CLK: EXT?
10:09.26NLiningtonok I am going to have a go with iaxmodem again, last time I tried it looked like it had the same problem.
10:09.28benjamin7062dlynes_office, yeah.. both PRI's..
10:09.58benjamin7062NLinington, I am going to try the Linksys analog->sip thing.. I'll let you know how many fail for us in a few days
10:10.32n3glvlinksys PAP2 ?
10:10.51benjamin7062yes sir
10:11.12benjamin7062I don't have high hopes based on reading
10:11.20benjamin7062but hell, it's $60.00 .. might as well try it
10:11.43NLiningtonbenjamin7062, 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.
10:11.45benjamin7062if it sux.. we'll hang it on the wall and watch the blinky lights
10:12.12benjamin7062NLinington, ahhh -- touche.. efax.  =)
10:12.37n3glvwell. that big ass V company uses a lot of PAP2's
10:12.54n3glvI have no 1st hand info on reliability
10:13.04benjamin7062I will in a few days
10:13.09dlynes_officebenjamin7062: your span line looks like span=1,1,0,esf,b8zs?
10:13.29benjamin7062span=1,1,0,esf,b8zs
10:13.29benjamin7062span=2,2,0,esf,b8zs
10:13.29benjamin7062dchan=24,48
10:13.29benjamin7062bchan=1-23,25-47
10:14.21benjamin7062Maybe I need to specify somewhere to use 'one' for timing
10:14.24dlynes_officebenjamin7062: what kinda chip is on the machine?
10:14.37benjamin7062AMD 4000 i think
10:14.45dlynes_officeand you've got plenty of memory i'm guessing?
10:15.09benjamin7062Only 1gig.. but 900free
10:15.11n3glvI have heard that * is not very good on non intel
10:15.23dlynes_officen3glv: you heard wrong
10:15.27benjamin7062n3glv, nonsense..  * lives in user land
10:15.33dlynes_officen3glv: lots of people are running fine on amd's
10:15.37n3glvok
10:15.46dlynes_officen3glv: same for zaptel
10:15.50n3glvcool
10:16.05dlynes_officen3glv: however zaptel doesn't work well on non-Linux OS's
10:16.11n3glvjust heard that tonight
10:16.13dlynes_officen3glv: or is completely non-existent
10:16.17benjamin7062n3glv, expecially if you use a generic 386ish build on your kernel... the apps wouldn't know the difference.
10:16.25dlynes_officen3glv: it's only been ported to FreeBSD
10:16.35n3glvwell, my SMP kern won't run ztdummy
10:16.38dlynes_officen3glv: and FreeBSD's porting effort is only considered beta quality at best
10:16.47n3glvhence my attempt at building a new kern
10:16.56benjamin7062n3glv, SMP breaks a lot in linux sometimes.. hehe
10:17.05n3glvyeah
10:17.22dlynes_officebenjamin7062: peoples' inability to install the right packages is what breaks in linux sometimes, not smp :)
10:17.38n3glvran me around in circles untill I found u guys and found out about ztdummy needed for meetme
10:17.44dlynes_officen3glv: paste uname -a's output to the channel
10:18.08n3glvit's a trixbox, well known issue I think
10:18.25dlynes_officetrixbox still runs asterisk under-the-hood
10:18.32benjamin7062dlynes_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)
10:18.36dlynes_officetrixbox's config files are just all fubar
10:18.53dlynes_officebenjamin7062: i would imagine it's the code, not the smb
10:18.54benjamin7062trixbox is * with all the crap loaded for you... on CentOS...
10:18.55dlynes_officebenjamin7062: i would imagine it's the code, not the smp
10:19.07n3glvLinux asterisk1.local 2.6.9-34.EL #1 Wed Mar 8 00:07:35 CST 2006 i686 i686 i386 GNU/Linux
10:19.23dlynes_officen3glv: ummm....that's not an SMP kernel
10:19.36benjamin7062n3glv, normally smp kernels specify
10:19.36n3glvnot running both cpu-s right now
10:19.46n3glvu want kern ver of smp kern?
10:19.47benjamin7062dlynes_office, STOP TYPING FASTER THAN ME!  o.O!
10:19.50dlynes_officen3glv: doesn't matter...it's not an smp kernel
10:19.54benjamin7062hehe
10:20.03dlynes_officen3glv: what kernel-dev do you have installed?
10:20.08*** join/#asterisk ASTERISKNEWBIEXX (n=chatzill@rrcs-67-52-187-18.west.biz.rr.com)
10:20.18n3glv2.6.9-34.EL-smp-i686
10:20.26*** join/#asterisk SHad|Work (n=kvirc@84.255.228.2)
10:20.37benjamin7062hmm?
10:20.43dlynes_officen3glv: that's your smp kernel version, or the version of kernel-dev you have installed?
10:20.45n3glvthat is kern that was failing
10:20.54SHad|Workgood day
10:20.55dlynes_officen3glv: what kernel-dev version do you have installed?
10:20.59n3glvhow I find that?
10:21.13dlynes_officen3glv: beats the funk out of me...I don't use an rpm-based distro
10:21.17ASTERISKNEWBIEXXcan someone fix my asterisk system ?  Ill pay them some money if  they can do it now.
10:21.19SHad|Workdoes anyone here have any experience with configuring multiple zaptel cards in a single server?
10:21.23benjamin7062rpm -<some arguments>
10:21.50benjamin7062ASTERISKNEWBIEXX, what's wrong?
10:21.59dlynes_officeASTERISKNEWBIEXX: type /nick asterisknewbiexx before I go blind, please
10:22.07n3glvyeah, I prefer debian most of the time
10:22.19benjamin7062dlynes_office, What did you just say, ?  I couldn't see it?
10:22.36n3glvhaha, yeah, shouting
10:22.37benjamin7062my eyes hurt
10:22.40benjamin7062ouch
10:22.46*** join/#asterisk MarcPtz (n=MarcPtz@18.Red-80-35-146.staticIP.rima-tde.net)
10:23.03dlynes_office~suggestions
10:23.08jbotit 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 ...
10:23.15benjamin7062Man, if I were hold music on my * box... I would wanna play clear... this hold music is rude
10:23.34dlynes_officebenjamin7062: i'm thinking you've got other issues, though
10:23.44dlynes_officebenjamin7062: i don't think it's a question of your drivers
10:23.56dlynes_officebenjamin7062: what else are you running on that box besides asterisk?
10:23.59n3glvI finally managed (don't know how) to get my box to play something else (cat stevens)
10:24.18benjamin7062dlynes_home, nothing.. clean install.. tftp, ftp, ssh, normal kernel poop, cron, etc
10:24.38dlynes_officehrm
10:25.05dlynes_officeasterisknewbiexx: now, what seems to be your problem?
10:25.14benjamin7062can you call US 800 numbers toll free?
10:25.27asterisknewbiexxi messed up my conf files
10:25.28dlynes_officebenjamin7062: who?
10:25.50dlynes_officeasterisknewbiexx: freepbx/trixbox/asterisk@home/amp?  or regular asterisk?
10:25.52n3glvben, I can
10:25.53benjamin7062dlynes_home, you... you can hear it by calling 877-87-photo...
10:26.02benjamin7062i'll have to go figure what those numbers are...
10:26.13AltnTabHow come asterisk cannot find a file when the wxact location is specified !?
10:26.23benjamin70628778774686
10:26.25n3glv48686
10:26.31dlynes_officeAltnTab: maybe your permissions are screwy?
10:26.44n3glvivr is fine
10:26.46asterisknewbiexxfedoracore 4 /asterisk 1.2.9/astguiclient-1.1.11/vicidialer
10:26.47benjamin7062AltnTab, Sometimes it's relative path also
10:26.56AltnTabdlynes_home, the permissions are default ones on Monitor()
10:26.58dlynes_officeasterisknewbiexx: and which particular config files are screwed up?
10:27.17dlynes_officeAltnTab: 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
10:27.25n3glvgot female
10:27.35n3glvtalking about call something for live talk
10:27.38dlynes_officeAltnTab: i was thinking more that you might not have execution bits on one of the parent directories
10:27.39n3glvsounds fine
10:27.47asterisknewbiexxsip.conf ,iax.conf ,zapata.conf
10:27.58benjamin7062WHAT?.. really.. one sec
10:28.04AltnTabdlynes_home, could be, i'll check
10:28.18dlynes_officeThe number you have called cannot be reached from your calling area
10:28.22dlynes_officebastards
10:28.43dlynes_officeI guess you don't want any business from Canada :p
10:28.54n3glvhaha, sounds 100% on my cell
10:29.00benjamin7062damnit.. gave you wrong number..  877(3)7photo
10:29.09*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
10:29.09n3glvif it's supposed to do a commercial for live talk that is
10:29.15n3glvhaha
10:29.17benjamin7062No, i'm just dumb
10:29.39asterisknewbiexxthink I messed up my zap channnels
10:29.55dlynes_office"The number you have dialed is not available in your area at this time."
10:29.56benjamin7062hmm
10:30.22n3glvcan u do that one numeric too?
10:30.59benjamin7062weird... our 800 number goes to some random guy
10:31.04benjamin7062i'm pretty sure that's broken
10:31.07benjamin7062damnit
10:31.13n3glvguy answered, had no idea what i was talking about
10:31.44*** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin)
10:31.48n3glvdo u have a failover set, like a no answer or busy number?
10:31.50benjamin7062OMG  -- our 877 number must forward to our president if our lines are broken
10:31.57n3glvYEAH
10:32.00n3glvI bet
10:32.00benjamin7062i didn't set it up.. I wouldn't know
10:32.04benjamin7062that's hawt
10:32.09benjamin7062I just had you spam my boss...
10:32.11n3glvheeheehee
10:32.13benjamin7062heh
10:32.14benjamin7062cool
10:32.15n3glvwoke his ass up I bet
10:32.20benjamin7062it's only 5:30 am
10:32.24benjamin7062better wake his ass up
10:32.30n3glvhope I didn't get u in trouble
10:32.36benjamin7062nah
10:32.39benjamin7062he'll live
10:32.43n3glvI said trying to test an asterisk box
10:32.57benjamin7062weird that it goes to him
10:33.02n3glvpretty funny though
10:33.06benjamin7062makes me wonder if perhaps ONE of our pri's is broken
10:33.10n3glvcould be
10:33.19benjamin7062they both show okay though
10:33.21benjamin7062so weird
10:33.58benjamin7062our direct dial works fine
10:34.28n3glvwho feeds you your did?
10:34.38benjamin7062time warner telecom
10:36.05n3glvok, and it was working? just crappy singnal to noise ?
10:36.09MarcPtzHi 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?
10:36.20benjamin7062n3glv, yup
10:36.26benjamin7062calls sound excellent
10:36.34benjamin7062hold music sounds like ass in a tunnel
10:36.48*** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au)
10:36.50n3glvu have some echocancelation stuff turned on?
10:36.56benjamin7062yes
10:37.05n3glvtry with it off?
10:37.08benjamin7062does that break hold music?
10:37.17n3glvgrasping here
10:37.31n3glvbut should or could affect audio qual
10:37.58n3glvstarting to get into my area, (live sound engineer here)
10:40.03benjamin7062whew
10:40.05benjamin7062okay
10:40.08benjamin7062same without EC
10:40.10benjamin7062I'm glad
10:40.16benjamin7062cause that would have sucked if that was the problem
10:40.25benjamin7062thing is.. it plays the 'prompts' fine
10:40.28benjamin7062it's just hold music
10:40.57n3glvhmm
10:41.08n3glvtry other music files?
10:41.24benjamin7062I've tried 3
10:41.30*** join/#asterisk kpettit (n=keith@69.15.174.114)
10:41.31benjamin7062they play clear on my system
10:41.32kpettitgood morning
10:41.37benjamin7062what is the actual 'player'
10:41.42benjamin7062maybe that's my problem
10:42.23n3glvthink it depends on install, but a lot use mpg123 or something
10:42.44*** join/#asterisk michael-i (n=michael@141.41.38.58)
10:42.49benjamin7062hmm
10:42.51benjamin7062tis isntalled
10:42.53benjamin7062installed
10:42.54n3glvGM kpetit
10:43.45kpettitanybody know some good places to look to hire asterisk folk?
10:43.48*** join/#asterisk backblue (n=igor@82.102.1.42)
10:43.52kpettitI need to find some people inm Houston
10:43.57benjamin7062www.digium.com
10:43.59backbluehi, morning all
10:44.09backbluedoes anyone it's trying to use hudlite?
10:44.11benjamin7062<-- in austin tx
10:44.19kpettitwanna move :)
10:44.32benjamin7062No way, not after Katrina...
10:44.34n3glvI'll move
10:44.37kpettithaha
10:44.39benjamin7062Houston = Lousianna now
10:44.42n3glvnot expret, but get by
10:44.42benjamin7062yuk
10:44.52backbluehudlite anyone?
10:45.00n3glvexpert that is <ha>
10:45.06kay2How 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)
10:46.04benjamin7062kay2, easier using Manager
10:46.33benjamin7062or maybe I'm lazy
10:46.44*** join/#asterisk jpeeler (n=thepeel1@host81-149-2-72.in-addr.btopenworld.com)
10:47.19kay2benjamin7062 ??? what u talking about
10:47.40kay2benjamin7062: I just say can my AGI write something in the CLI
10:48.04benjamin7062I know
10:48.16benjamin7062was 'teasing'
10:48.56benjamin7062It was sarcastic in my .. you can't read my mind?
10:49.11benjamin7062[head]
10:49.15benjamin7062man, I'm getting tired...
10:49.30*** join/#asterisk duckz (n=duckz@193.192.47.26)
10:50.49benjamin7062kay2 -- what language is your AGI in?
10:51.18benjamin7062kay2, couldn't you drop to system and run asterisk -rx'CLI Command'
10:51.19benjamin7062?
10:51.21kay2no matter
10:51.32kpettitwhy not use AMI?
10:52.00benjamin7062I got smacked for saying that
10:52.11dlynes_officeami?
10:52.19dlynes_officeoh...nvm
10:52.20dlynes_officeduh
10:52.42kpettitbenjamin7062, I like ami, works pretty well for me.  Better than using asterisk -rx "" anyways
10:52.44benjamin7062I assume that means * Manager Interface?
10:52.50kpettityes
10:53.00kpettitit works nice with PHP which is mainly what i program in
10:53.03benjamin7062kpettit, I agree... I said that... and she said I was crazy
10:53.09benjamin7062That is what _I_ use
10:53.16benjamin7062AMI
10:53.25kpettitI use it to get queue stuff, transfer/place calls, etc.
10:53.30benjamin7062exactly
10:53.41benjamin7062she apparently is writing something that needs a voice channel I imagine
10:53.45benjamin7062thus, she is using the AGI
10:53.57benjamin7062so I'm guessing she needs something for a voice prompt or something
10:54.00benjamin7062She didn't say
10:54.25benjamin7062But an alternative from the AGI would be `asterisk -rx'something'` since she didn't like my AMI suggestion
10:55.12benjamin7062<kay2> How can I write something in the CLI from an AGI ?
10:55.12benjamin7062* peterm22 (n=petermit@219-90-223-69.ip.adam.com.au) has joined #asterisk
10:55.12benjamin7062* FaithX has quit (Connection reset by peer)
10:55.12benjamin7062<benjamin7062> kay2, easier using Manager
10:55.51kpettitgot ya
10:56.23kpettitMy big thing i'm trying to program this week is seeing if I can log a agent in thro0ugh a web page
10:56.33benjamin7062easy as pie
10:56.39benjamin7062if you are using 1.2.9.1
10:56.42kpettitI can't find any AMI or asterisk -rx type commands i can use to log a agent in.  Logging out was easy enough
10:56.52benjamin7062AgentCallBackLogin
10:56.54kpettiton some machines i am.
10:57.36*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
10:57.42kpettithttp://www.voip-info.org/wiki-Asterisk+cmd+AgentCallbackLogin
10:57.52benjamin7062http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+AgentCallBackLogin
10:58.10peterm22hi everyone, quick q. what app can i use for receptionist to "set callerid name"
10:58.48benjamin7062Few lines.. especially if you already wrote the functions to connect, login, and communicate which it sounds like you did
10:58.49kpettitpeterm22, real time asterisk with a web interface for the sip section
10:59.05kpettitbenjamin7062, sweet this looks nice
10:59.23benjamin7062has to be 1.2.1+
10:59.37kpettitbenjamin7062, 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
10:59.55kpettitoh yeah I've got that on 90% of my systems.  I think everything I'm doing queus with is 1.2.4 or better
11:00.03benjamin7062tis 'exactly' what we do from our managers polycom 601 xhtml browser.. =)
11:00.17kpettityou do that from a polycom???
11:00.21kpettitthat's freaking cool
11:00.33benjamin7062I've written more code in 5 days than ever in my life at one time
11:00.39kpettitwow.
11:00.56kpettithow does that work?  do you have to have the attachment module?
11:01.15benjamin7062no... xhtml browser is on all polycom 601's
11:01.24benjamin7062the expansion is kinda useless
11:01.33kpettitI 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
11:01.38benjamin7062most you can do is call 'status' for people.. not functions
11:02.03benjamin7062kpettit, you mean in the sip.cfg?
11:02.03kpettitbenjamin7062, I use those like nuts for call status.  People seem to be so used to those old style key systems
11:02.05benjamin7062features
11:02.15kpettitbenjamin7062, either in there or the individual phones.
11:02.26benjamin7062hmmph.. didn't know the phone would let me do that.
11:02.28benjamin7062intersting
11:02.34kpettitI 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
11:02.44kpettitbenjamin7062, its' saved me soooo much heartache
11:03.11kpettitagents would login then press DND, or forward there phones to some random extension, wich of couse the phone dosen't check
11:03.28benjamin7062I've been mildly dissapointed at the button flexibility on almost all the available sip phones... I wish you could make buttons do 'anything'
11:03.43benjamin7062the 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)
11:03.48benjamin7062that I have found thus far
11:03.57kpettitI know I can disable the buttons, which worked great.  But that's as much as i've done with them
11:04.13benjamin7062I want a 'park' button.. =)
11:04.25kpettitmy pain in the ass right now is parking as well.
11:04.58kpettitdoing 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
11:05.14benjamin7062It works... I just wanna' BUTTON for it.  But not possible I don't think
11:05.24kpettitso 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
11:05.25benjamin7062kpettit, you can bump the timer for parked calls to 1000000
11:05.34benjamin7062and then have a web interface on the 600 show parked calls.
11:05.47kpettitI thought aobut that, but in a resturant that's impossible
11:05.53benjamin7062true
11:05.54benjamin7062good point
11:06.13kpettitthey 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
11:06.37benjamin7062yup, and if you bump the timeout... they might forget about the call
11:06.54benjamin7062have it call back there phone for 30 sec... then forward to park again?
11:07.07kpettitexactly.  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
11:07.24benjamin7062You could capture the ring back in your dialplan
11:07.28benjamin7062with some trickery
11:07.31kpettithows that
11:07.32benjamin7062then do something else with the call
11:07.52benjamin7062it will ring back the same phone.. but what you do AFTER that ring back doesn't have to be voicemail
11:08.14kpettitI 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
11:08.31kpettitoh got ya
11:08.42kpettitughh I see a painfull alternative.
11:08.48kpettitit could workthough
11:09.18benjamin7062It 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)
11:09.39benjamin7062otherwise, hit up voicemail
11:09.41kpettitwhen it rings back thogh where do i control that in the extensions.conf
11:09.58kpettitit's the page app that does the ring back, how can i set a second priority on that?
11:10.04lilalinuxdoes anyone know a cheap way to connect a mobilephone as a channel?
11:10.10peterm22kpettit, i only need to change incoming call callerid name, do i need realtime for that ?
11:10.15benjamin7062well, 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
11:10.38benjamin7062lilalinux, heh, call a channel?
11:10.38kpettitpeterm22, unless you want ot had edit sip.conf everytime you want to change caller id
11:10.43benjamin7062lilalinux, does that count?
11:10.56benjamin7062lilalinux, just kidding
11:11.04*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
11:11.36kpettitbenjamin7062, 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
11:11.37lilalinuxbenjamin7062: 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)
11:13.00kpettitlilalinux, 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.
11:13.01benjamin7062when 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
11:14.14kpettitbenjamin7062, yeah I think some tinkering is in order.  I know in features.conf I specify the context for "parkedcalls" to use
11:14.19benjamin7062lilalinux, 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 *
11:14.21kpettitjust haven't tried putting that in extensions.conf yet
11:14.42benjamin7062kpettit, yeah, but that context is different than the ring back I believe... let me test real quick
11:14.43kpettitor rather I've done the include but not put a [parkedcalls] in there
11:15.56benjamin7062parkedcalls is just an include I think
11:15.59kpettitbenjamin7062, I wonder if i could jsut have a [parkedcalls] with a exten => t,1,,,,
11:16.01benjamin7062so you can include the parkinglot
11:16.20kpettit?
11:16.21benjamin7062i 'think'
11:16.31benjamin7062one sec
11:16.36n3glvI could not get ringback to work for me
11:16.46n3glvam guessing I missed something
11:18.14benjamin7062Hmm, 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
11:18.29n3glvso, how do I set the level of verbosity on the disconnected cosole screen?
11:18.30benjamin7062I'm trying to capture the context of the returned call from park
11:18.37n3glvthe call progress monitor etc
11:18.46benjamin7062set verbose 13
11:18.49benjamin7062set debug 13
11:18.56kpettitok cool.  I'm experimenting with timeouts in the parkedcall context
11:19.53kpettitit looks like it's using "park-dial" to do the ring back
11:20.16n3glvgot any ideas why I keep seting the server apear to try login 2 times to my host?
11:20.35n3glvregister 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)
11:21.26tomtom_hi
11:21.31n3glvREGISTER attempt 1 to 1412etc@neptune.vtnoc.net
11:21.44n3glvthen agin
11:21.47n3glvagain
11:23.02*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
11:23.20*** join/#asterisk anna-- (n=nmuller@195.70.21.58)
11:23.55anna--hello
11:24.11kpettitbenjamin7062, 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
11:24.14backbluehttp://lists.digium.com/pipermail/asterisk-dev/2006-May/020904.html
11:24.20anna--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?
11:25.02backblueanna--: define groups, and use it.
11:25.08backblueinstted of MSN's
11:25.17backblueand define the msn on the group
11:25.35anna--in zapata.conf you mean?
11:26.25backblueanna--: what driver are you using?
11:26.34anna--I'm using the zaphfc module
11:26.37backbluewhy use CAPI?
11:26.46backblueso, you are using bristuff
11:26.51anna--I was using CAPI before because I had a AVM card with chan_capi
11:27.13anna--but now I'm using a HFC card, so i'm using bristuf and zaphfc
11:27.21backblueforget capi, forget bristuff, use misdn
11:27.34backblueanyway, you can do the same, on zapata.conf
11:27.41backblueif you want to use bristuff
11:28.01anna--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
11:28.02backbluefrom my experiance, misnd lot better.
11:28.06tomtom_anyone any experience with ISDN BRI Gateways, such as Mediatrix, Vegastream or Voxtream Parlay .. any recommendations for use with Asterisk?
11:28.19anna--never tried it with the HFC though...
11:28.21backblueanna--: lspci, and show me the card id.
11:28.34backbluehfc cards, works great
11:28.38kpettitbenjamin7062, I figured it out!!
11:28.42backbluewhat card do you have? single S0?
11:28.49anna--00:0b.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02)
11:28.57backblueyes, use misnd
11:28.57benjamin7062kpettit, using the parkedcalls context?
11:29.00backbluewill work great
11:29.15backbluei have plenty of machines with cards like that
11:29.18kpettitin the console you can see it setting a priority 1 extension
11:29.23anna--and with mISDN I don't need to patch the asterisk sources?
11:29.54benjamin7062yeah
11:30.16kpettitbenjamin7062, in park-dial i set a exten => 701,2,SayUnixTime as a text and it worked
11:30.32benjamin7062Ahh... I see
11:30.36kpettitbad thing is the timeout.  It's got to ring for like 60 seconds before it goes to the priority 2
11:30.53kpettitI 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)
11:30.57BertZhi again
11:31.32kpettitbenjamin7062, without hardcoding what asterisk is already doing automatically with parked calls
11:31.35benjamin7062change the timeout for the extension it is calling
11:31.51kpettitit's not calling a extension but the sip peer directly
11:31.52benjamin7062see if that affects this?
11:32.02benjamin7062oh, duh
11:32.03benjamin7062right
11:32.16kpettitI wish I could make it dial a exten
11:32.28benjamin7062I didn't even catch that
11:32.34benjamin7062my original thoughts wouldn't be valid
11:32.44*** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au)
11:33.23kpettitbenjamin7062, 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)
11:33.35kpettitbut sense it never rang back the extension my ideal didnt woprk
11:33.47kpettitthis kind of works but the timeout is awfull
11:34.10benjamin7062wonder if you can override priority 1?
11:34.14kpettitI guess it's nce at least to be able to send a 2nd priority at all
11:34.18kpettitbenjamin7062, that would be nice.
11:34.22kpettitor rather modify it
11:34.33benjamin7062what if you specify a priority of 1..would that overrule?
11:34.38kpettitactually I'm going to try that.  I wonder if it'll bugger a parked call
11:34.44kpettittesting...
11:34.47kesaHi. 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?
11:35.13benjamin7062This is a scenario I would have never tested so this is good for me to know!  =)
11:35.31Zeeekkesa you mean with software phones on each computer?
11:35.38ZeeekIn that case, absolutely
11:35.44kpettitbenjamin7062, well it's still parking.. so that's good
11:35.49benjamin7062heh
11:36.52kpettitbugger, 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)
11:37.03benjamin7062hmm
11:37.09benjamin7062HAS to be a way to configure that
11:37.15kpettitIt must write that on the fly and overwrite what I do
11:37.41kesaZeeek: So one computer could act as a switching board and just forward calls to different computers through the use of Asterisk?
11:37.47kpettitwhen that priority 1 failes it's still doing my SayUnixTime for priority 2 though
11:38.23Zeeekkesa that's something it does very, very well assuming you have softphones on each PC with headsets or USB phones
11:38.25benjamin7062so the dial back (park-dial) is hard coded as a Dial command
11:38.26benjamin7062beh
11:38.38redaxhi
11:38.59benjamin7062I'm going to peak at the source
11:39.10redaxis the Queue is suitable to
11:39.36redaxuse when too many incoming calls, and no internal to ring?
11:39.44redaxwithout Answer() of course
11:39.46kpettitbenjamin7062, yes it seems to be.  I added my own park-dial context so I could add the second priority
11:40.04kpettitbenjamin7062, so I could try to match what it seems asterisk was automatically doing
11:40.48kpettitthis is what it does when the parked call "times out" by default...
11:40.58kesaZeeek: I see. Thanks! :) So that function is built-in already?
11:41.04kpettit<PROTECTED>
11:41.04kpettit<PROTECTED>
11:41.04kpettit<PROTECTED>
11:41.43benjamin7062I'm looking for park-dial in the source
11:41.49Zeeekkesa look here: http://asteriskdocs.org - read the intro of the book there
11:42.12kesaZeeek: Alright. Thanks again :)
11:42.26kpettitis this a valid exten?   exten => SIP/2009,1,SayUnixTime ??
11:42.38Zeeekalso there is no better FIRST introduction to asterisk than this: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
11:42.51Zeeekit's old, but very easy to understand
11:43.11benjamin7062Umm, ?.. I honestly can't say... Also have not tried that.  I have yet to see that exist
11:43.21benjamin7062where you specify the channel for an extension
11:43.23benjamin7062but maybe?
11:43.26kpettitthat's what it looks like it's doing. I'm testing...
11:43.49*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
11:44.39benjamin7062it actually creates that context on the fly (park-dial)
11:45.20kpettitbenjamin7062, yeah but it listens to what I put in there.  it just overwrites what I set as priority1
11:45.34kpettithumm setting a exten => Sip/2009 didn't work for me
11:46.00kpettit<PROTECTED>
11:46.08kpettitthat's the part asterisk doing that I need to overwrite
11:46.27benjamin7062you could modify the source in res_features.c to do what you want
11:46.57kpettitI think I'll have to tell the customer ot go to hell first.  haha
11:47.10benjamin7062it doesn't look that bad actually
11:47.16kpettitI've got to many boxes to keep track of to figure out different source code modificatoins
11:47.52dlynes_officekpettit: use a centralized svn server and create a separate repository for each server
11:48.13dlynes_officekpettit: have them download new configurations every night
11:48.26dlynes_officekpettit: 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)
11:48.50kpettitdlynes_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
11:49.17benjamin7062looks like you'd change this one line
11:49.19benjamin7062ast_add_extension2(con, 1, peername, 1, NULL, NULL, "Dial", strdup(returnexten), FREE, registrar)
11:49.21dlynes_officekpettit: custom programming always warrants a higher per hour rate, too :)
11:49.25kpettitdlynes_office, I know there is other junk I'll want to modify, it's just a scary game to get into
11:49.42kpettitdlynes_office, I wish.  We just charge by system not by features.
11:49.58dlynes_officekpettit: tell them that feature's only included in the deluxe edition
11:50.08dlynes_officekpettit: and the deluxe edition is a higher price
11:50.20kpettitdlynes_office, for the extra $$godawfullammout fee
11:50.27dlynes_officekpettit: exactly
11:50.32*** join/#asterisk userdefined (i=jr000430@shell1.phx.gblx.net)
11:50.34dlynes_officekpettit: now you catch my drift :)
11:50.41kpettitbenjamin7062, did that specify a timeout?
11:50.48dlynes_officebenjamin7062: ummm....no???
11:51.27kpettitbenjamin7062, 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
11:51.44Zeeekkesa I'm there
11:52.19Mw3hi. 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
11:52.19benjamin7062dlynes_home, You don't think that'd work?
11:52.36dlynes_officebenjamin7062: use ast_strdupa, or you're going to have a memory leak
11:52.37benjamin7062dlynes_office, dang, wrong name
11:53.02benjamin7062=) I just copied that line from the source.. didn't modify it
11:53.10benjamin7062So, they might have a leak
11:53.16dlynes_officebenjamin7062: strdup allocates memory for the pointer and returns a pointer to the malloc'ed memory
11:53.49dlynes_officebenjamin7062: ast_strdupa allocates memory on the stack, not the heap
11:53.55dlynes_officebenjamin7062: and so no need to free it
11:54.30*** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135)
11:54.32benjamin7062right now I'm trying to comprehend what they are doing and why
11:54.51benjamin7062but I think that method probably has the ability to set a timeout
11:55.04benjamin7062assuming it's the "add a call to the queue" method
11:55.05benjamin7062not sure
11:55.06dlynes_officeah wow
11:55.10dlynes_officewhoever wrote that should be shot
11:55.19dlynes_officeor wait
11:55.23kpettitbenjamin7062, I'm going to check in setting a timeout in the sip peer
11:55.31benjamin7062heh
11:55.31kpettitI wonder if that's doable
11:55.53benjamin7062kpettit, hmm?  suppose that'd be phone specific?
11:56.00kpettithttp://www.voip-info.org/wiki/index.php?page=Functions
11:56.10kpettitI'm wondering if I can use the TIMEOUT type function specified here
11:56.30kpettitwhoops wrong page
11:57.07*** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au)
11:57.27kpettithttp://www.voip-info.org/wiki/view/Global+Varialbes
11:57.43dlynes_officeah...nvm
11:57.44kpettitif not in extensions.conf maybe in sip.conf.
11:57.52dlynes_officeI think they free the memory later on
11:58.05benjamin7062kpettit, 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
11:58.20benjamin7062I don't have ctags on this machine
11:58.27kpettitah
11:58.27dlynes_officenormally though, unless you need the memory to persist past the current function, you should only use ast_strdupa instead of strdup
11:58.30benjamin7062so jumping around is .. poop
11:58.57kpettithttp://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)
11:59.02kpettitwhat do you tihink of that var?
11:59.08dlynes_officekpettit: what is it you're trying to do?
11:59.12benjamin7062dlynes_office, I think someone should sign their name to each piece of code so we have a target to shoot at
11:59.28dlynes_officebenjamin7062: 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)
11:59.35benjamin7062dlynes_home, hook a return call from park and give it an action other than return the call
11:59.35alucard064re all
11:59.43kpettitthe 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
11:59.50alucard064i try to find a software
12:00.02alucard064its name is ipswitchboard
12:00.02dlynes_officekpettit: ah
12:00.13dlynes_officealucard064: yeah...it's a .NET client application
12:00.17alucard064someone have this software
12:00.18alucard064yes
12:00.27alucard064its a free client
12:00.28kpettitdlynes_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
12:00.35dlynes_officealucard064: not quite
12:00.42dlynes_officealucard064: he might be charging for it in the future
12:00.49dlynes_officealucard064: but it's free for now, but not opensource
12:00.54alucard064yes
12:00.58alucard064it s free
12:01.01*** join/#asterisk nothinman (i=shakey@aczw177.neoplus.adsl.tpnet.pl)
12:01.06alucard064but i want it to try it
12:01.08kpettitdlynes_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
12:01.12nothinmanhry
12:01.15nothinmanhey*
12:01.19alucard064but i cna find him
12:01.30alucard064i cant find it
12:01.51alucard064so if someone have a version of this software
12:01.55alucard064please send me
12:02.04*** part/#asterisk Joe__11 (n=develope@host217-114-154-220.pppoe.mark-itt.net)
12:02.09dlynes_officealucard064: http://ipswitchboard.thorben.dk/
12:02.16alucard064lol
12:02.23alucard064yes i know this adress
12:02.26dlynes_officeThat's the official homepage for it
12:02.39nothinmanguys, 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...
12:02.40dlynes_officeIf you know the address, why haven't you downloaded it?
12:02.47alucard064but th software its not avaible and in return it give me the links of easypbx
12:03.08alucard064there is no links to download the software
12:03.32alucard064so i thought that someone can give me ipswitchboard
12:03.38alucard064if it s possible
12:04.02dlynes_officealucard064: go to easypabx.com, and then click on 'Login'
12:04.04znoGquestion: 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
12:04.13dlynes_officealucard064: It's a link in the upper righthand corner
12:04.14znoGoops, sorry, typed a bit too much for one message.
12:04.24alucard064yok
12:04.26benjamin7062kpettit, beh... that method doesn't allow timeouts
12:04.34alucard064but when can i download it
12:04.42kpettitbenjamin7062, bummer
12:04.43dlynes_officealucard064: no idea...i haven't signed up
12:04.47dlynes_officealucard064: i dont' use windows
12:04.53kpettitbenjamin7062, so without changing the source I'm kinda screwed then it looks like
12:04.56alucard064because i want to developp a software as example ipswitchboard
12:05.18benjamin7062only 50... you still have a 60 second solution
12:05.19kpettitis there any good RTA web interfaces out there?
12:05.23kpettitI hven't checked in awhile
12:05.27benjamin7062make the parked timeout like 15 secs.
12:05.28benjamin7062heh
12:05.53kpettitbenjamin7062, that just sends it back to the sip peer in 15 seconds, which will ring that sip peer for ever
12:06.08benjamin7062not if there's a priority 2, right?
12:06.08kpettituntil it finaly fails over to my 2nd priority which rings all the phones again
12:06.16alucard064so
12:06.26benjamin7062so 75 seconds... and they are back on the phone
12:06.30alucard064if someone have ipswitchboard
12:06.35benjamin7062There's another hax you could do
12:06.40kpettit?
12:06.42alucard064can he give me the software
12:06.45alucard064thanks
12:06.57benjamin7062But you'd only kinda have 1 call park
12:07.06benjamin7062or 1 per button
12:07.13benjamin7062this is REALLY hax though
12:07.15kpettitI 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
12:07.29kpettitgo on.... sounding good.,...
12:07.52benjamin7062create 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
12:07.55kpettitbenjamin7062, paging through the ip501's and 601's kicks ass by the way.  I sooo love that
12:08.05benjamin7062when they hit the 'call park' button.. it basically does a queue login
12:08.10backblueanyone with hudlite?
12:08.11benjamin7062agent login rather
12:08.18benjamin7062they get the call..
12:08.23benjamin7062make the queues max len = 1
12:08.28benjamin7062heh
12:08.32benjamin7062it's hax though
12:09.33kpettitah I got ya.  I'm buzzing that to see if that's optional there.
12:09.36kpettitI love queues though
12:09.37dlynes_officealucard064: try here:  http://ipswitchboard.thorben.dk/index.php?option=com_simpleboard&Itemid=42
12:09.49benjamin7062It would accomplish your park goal
12:09.52benjamin7062but it's uglyt
12:09.53benjamin7062ugly
12:10.08dlynes_officebenjamin7062: and asterisk source code isn't? :)
12:10.14benjamin7062lol
12:10.16benjamin7062touche
12:10.18kpettitman they just need a key system.  these old bastards that can't learn new tricks drive me nuts
12:10.34dlynes_officekpettit: so why do you need to do all this parking crapola then?
12:10.52dlynes_officeJust use autoattendant, if they don't feel like answering the phone, throw it into voicemail
12:11.07dlynes_officeAnd use hinting on the extensions, so they can see each others' statii
12:11.19kpettitdlynes_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
12:11.39dlynes_officekpettit: ok
12:11.54benjamin7062hell, 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
12:12.03benjamin7062I know that's not the right word for it but you get the point
12:12.04kpettitso 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
12:12.10dlynes_officeand it's just timing out on the park, instead of beeping you to remind you that you have a parked call?
12:12.29benjamin7062he wants it to ring back to 'all' sip phones
12:12.32kpettitbenjamin7062, I haven't heard that term before
12:12.39dlynes_officebenjamin7062: ah
12:12.48benjamin7062there is a way to force a channel to connect to another channel
12:12.48alucard064thanks dlynes_office
12:13.01benjamin7062but i'm going to have to dig in my cobwebs
12:13.05kpettitdlynes_office, if it just rings back to the one sip phone that made the park, chances are that person isn't there anymore
12:13.06benjamin7062(the back of my brain)
12:13.09*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
12:13.23kpettitbenjamin7062, is this like remote call pickup?
12:13.24dlynes_officekpettit: and how does it do that?  sorry...i'm not familiar with parking
12:13.31dlynes_officekpettit: does it ring a Local extension?
12:13.50benjamin7062it's hard coded to ring back the sip/<ext> so you can't capture the extension even
12:14.01benjamin7062it rings the channel directly
12:14.04*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
12:14.08dlynes_officeic
12:14.10kpettitdlynes_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)
12:14.59kpettitthen 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)
12:16.55kpettitdlynes_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
12:17.44kpettitso 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?
12:17.46creadurxi <3 click 2 dial
12:18.12*** join/#asterisk hwt (n=hwt@195.139.204.157)
12:18.13kpettitI have no idea where it's getting that onw
12:18.14hwtwhat 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?
12:18.27hwti have compiled zaptel, and it is confirmed loaded.
12:18.28benjamin7062Don'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
12:18.30hwtthe kernel module.
12:18.47kpettithwt, sounds like you don't have a zaptel card or zaptel modules loaded.  I'm just guessing though
12:19.04benjamin7062that function is defined in pbx.c if you want to peak
12:19.13hwtkpettit: it is confirmed loaded.
12:19.15*** join/#asterisk nortex (n=nortex@64.136.65.142)
12:19.18kpettitbenjamin7062, the park or the sip timeout?
12:19.24hwtkpettit: i don't have a zaptel card, though.
12:19.31benjamin7062the park-dial timeout
12:19.32kpettithwt, you dong zt_dummy?
12:19.37kpettitah.
12:20.20dlynes_officekpettit: rtptimeout=30
12:20.34dlynes_officeerm 60 i mean
12:20.36tomtom_so no one any experience with bri gateways
12:20.57dlynes_officetomtom_: it sounded like you wanted an isdn->gsm gateway
12:21.03benjamin7062I know I saw a way to take a channel and force termination to another channel... or force a transfer
12:21.15dlynes_officebenjamin7062: there is
12:21.19kpettithwt, I'm not sure.  It just just be yoru mpg123 version or a sound file recorded at the wrong settings
12:21.42hwtkpettit: maybe, i'll look into it.
12:22.06kpettitdlynes_office, I just tried that one actually
12:22.26hwtcan i have extensions loaded both from extensions.conf AND realtime?
12:22.33kpettithwt, yes
12:22.34dlynes_officekpettit: »·»·»·»·»·»·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
12:22.37hwtor does that only work with sip peers.
12:22.40hwtkpettit: k, thanks.
12:22.44hwtcool.
12:23.23dlynes_officekpettit: 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)
12:23.40kpettitdlynes_office, I'm alittle lost
12:23.53dlynes_officekpettit: i guess you're not a c coder?
12:23.55kpettitwhat do you mean by peername
12:24.06kpettitPHP is and basic C is about all I have in me right now
12:24.43DovidHow do I start a console debug ?
12:24.55dlynes_officekpettit: say like SIP/2009
12:25.02dlynes_officeDovid: set debug 9999
12:25.15dlynes_officeDovid: then sip debug
12:25.21benjamin7062Dovid, or asterisk -vvvvvvvvvvvvvvvvvvvddddddddddddddddddddr
12:25.21dlynes_officeDovid: or iax2 debug
12:25.22kpettitrtpholdtimeout  I'm going to try that.  A parked call is on hold... Havne' done that one yet
12:25.29dlynes_officeDovid: or pri intense debug
12:25.39kpettitdlynes_office, got ya
12:25.46*** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au)
12:26.13benjamin7062Driving me crazy.. I can't find the 'thing' that I saw that would force two channels to join/terminate
12:26.19dlynes_officekpettit: so in your case, you would have a line in your features.conf that looked like 'ringbackext=Local/ringall'
12:26.30dlynes_officekpettit: and use that instead of peername if it's set to anything
12:26.52kpettitbut I'd have to do the C code to have that right?
12:26.54dlynes_officekpettit: it's a pretty simple change to res_features.c anyways
12:27.02dlynes_officekpettit: correct
12:27.21kpettitok
12:27.39dlynes_officeand then once you're happy with it
12:27.46tomtom_dlynes_office: no, not an isdn->gsm gateway, basically a voip->bri gateway, instead of using  bri pci cards
12:27.49dlynes_officeyou might want to post it to bugs.digium.com and share it with others
12:27.52benjamin7062in your case, you could just hard code the return ext into res_features.c for that particular install
12:28.09dlynes_officebenjamin7062: yeah, but it's better to do it the proper way
12:28.23dlynes_officebenjamin7062: and then kpettit doesn't have to keep patching the code with every new release
12:28.36benjamin7062yeah.. but I just realised.. that function doesn't take 'extension'.. it takes 'channel'.. anyway, so I'm wrong either way
12:29.22benjamin7062wonder if it'd accept sip/234&sip/234&sip/234
12:29.30dlynes_officebenjamin7062: of course
12:29.36dlynes_officebenjamin7062: it's only parameters to the dial command
12:29.47dlynes_officebenjamin7062: but to keep it simple
12:29.59dlynes_officebenjamin7062: i'd still do ringbackext=Local/ringall
12:30.16benjamin7062dlynes_office, I didn't follow it through what it was doing
12:30.25benjamin7062just 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)
12:30.30kpettitId' be nice if I could jus tdo exten => sip/2009,1,,,,  type of thing
12:30.39benjamin7062lol
12:30.43dlynes_officebenjamin7062: and then in [default], do exten => ringall,1,Dial(SIP/2009&SIP/2010&SIP/2011)
12:30.44kpettitso if it called sip/2009 but I think that's not going to happen
12:30.47dlynes_officeerm
12:30.51kpettitexactly
12:30.52benjamin7062you can if you're willing to accept 60sec timeout
12:30.57kpettithah
12:30.57*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
12:31.00*** join/#asterisk graphyx (n=mkling@24.144.57.161)
12:31.00dlynes_officebenjamin7062: and then in [default], do exten => ringall,1,Dial(SIP/2009&SIP/2010&SIP/2011,,t)
12:31.02kpettityeah that's what I've got now
12:31.09graphyxanyone familiar with sivus?
12:31.36dlynes_officekpettit: what is?
12:31.50graphyxThe Sip Protocol security scanner.
12:32.16kpettiti've got a second priority that goes after 60 seoncds that does a ringall to all the phones
12:32.46kpettitthe 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
12:32.59kpettitunless I do some C hacking that is :)
12:33.49benjamin7062it's application transfer it think...
12:34.06benjamin7062if the call were sitting in a queue... you could have a button that 'transferred the call the that phone' upon pressing it
12:34.09dlynes_officekpettit: well, what i suggested is probably exactly what you want
12:34.10benjamin7062there you go
12:34.13benjamin7062=)
12:34.18dlynes_officekpettit: and then you can change it easily, too
12:34.23*** join/#asterisk mut (n=animenod@65.111.222.120)
12:34.24benjamin7062that completes my UGLY ass way to accomplish this
12:34.36dlynes_officekpettit: you don't have to modify the code every time you want to change the extensions it rings
12:34.43dlynes_officekpettit: or the dial pattern, or whatever
12:35.25benjamin7062kpettit, dlynes_office's solution is far more accurate and elegant than mine... =)
12:35.25kpettitdlynes_office, I understandthat part.  I just don't think changing code is going to be a good option
12:35.56dlynes_officekpettit: well, to each his own, i suppose
12:36.14kpettitdlynes_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)
12:36.33dlynes_officekpettit: yeah...but something like that would take about 1/2 hour of C hacking, if that
12:36.40kpettitand 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)
12:36.51Lord_Drachenbluthello
12:37.00dlynes_officeyeah...that's something you'd change on your test machine only, until you're happy with it
12:37.06dlynes_officethen when you're happy with it
12:37.07Lord_Drachenbluthave a question to ask if you guys have the time
12:37.14dlynes_officerepackage your asterisk and redeploy
12:37.21dlynes_office~suggestions
12:37.34jbotfrom 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 ...
12:37.34kpettitdlynes_office, I apprecaite it.
12:37.47kpettitdlynes_office, I need to find another body to help me out with this stuff.
12:38.03Lord_Drachenbluti am trying from cli to use the playback command to play an audio file but i keep getting  -bash: syntax error near unexpected token `('
12:38.10dlynes_officekpettit: heh...i'd help, but i'm already overloaded with work as it is
12:38.29kpettitdlynes_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)
12:38.38dlynes_officeLord_Drachenblut: ummm...why are you trying to use playback from the cli?
12:39.10benjamin7062<-- just worked another 40 hr shift
12:39.14benjamin7062and I have to stay all day
12:39.16benjamin7062and stay late
12:39.16benjamin7062sigh
12:39.17Lord_Drachenblutdlynes_office, i am working on a graphical frontend that when you hit a button will call a person and playback an audio file
12:39.20dlynes_officebenjamin7062: ring a ding
12:39.28dlynes_officebenjamin7062: i've been up since 8am yesterday morning
12:39.36benjamin7062Me too!  =)
12:39.40dlynes_officebenjamin7062: that's how it goes in the it industry
12:39.47benjamin7062I wondered why you were always in chat while I am!
12:39.53*** join/#asterisk fenlander (n=fenlande@82.152.81.57)
12:40.01benjamin7062thought maybe you were in Australia or something
12:40.02dlynes_officeIt's 5:40am here
12:40.15graphyxso nobody has tried out the sivus scanner at all?
12:40.16benjamin70628:00 here
12:40.26dlynes_officeI'm busy building a bunch of packages to deploy for troubleshooting some buggy aastra firmware
12:40.31benjamin7062well.. 7:40
12:40.32mutthats only how it goes when ya don't know what you're doing
12:40.43dlynes_officestupid companies that make windows-only voip debugging solutions
12:40.51dlynes_officelike wtf?
12:40.58benjamin7062vmware!
12:40.59kpettitdlynes_office, benjamin7062 must just be this line of work
12:41.02benjamin7062that's how I run windows
12:41.07dlynes_officehow the hell are you supposed to run Windows apps remotely?
12:41.13benjamin7062kpettit, and doctors!
12:41.14creadurxwhat better way to run windows than in a window
12:41.31dlynes_officeespecially when you don't have access to do port mapping?
12:41.52benjamin7062dlynes_home, oh, i see..  the windows app is on someone 'elses' machine
12:41.58dlynes_officeno
12:42.00kpettitoh god I had the works sip bug that was driving me freaking nuts for weeks
12:42.02benjamin7062VNC?
12:42.14dlynes_officeI have to run a .net app to access an aastra phone that's sitting behind a firewall at a remote site
12:42.23kpettitour T-1 provider is our SIP provider whcih is nice (fax works great over sip), but DTMF wasn't working with somepeople
12:42.24benjamin7062lol
12:42.26benjamin7062hawt
12:42.47dlynes_officeand 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
12:42.56kpettitdtmfmod=auto and every other option, and we did all other sorts of debuging, but the fix ended up being setting canreinvite=yes
12:43.16benjamin7062I'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
12:43.20kpettitthat was killing me.  I guess the telco didn't like asterisk sending the dtmf so doing canreinvite=yes fixed it
12:43.32dlynes_officekpettit: canreinvite=yes makes asterisk get its ass out of the media path
12:43.51kpettitbenjamin7062, I'm with yha man.  The closest thing to a windows box I have in work or house is SuSE
12:43.55dlynes_officebenjamin7062: ummm
12:44.02*** join/#asterisk Dovid_Laptop (n=none@barak.cellcom.co.il)
12:44.09Lord_Drachenblutdlynes_office, no idea's?
12:44.10dlynes_officebenjamin7062: i haven't used a windows desktop on any of my machines for as long as i can remember
12:44.13dlynes_officeLord_Drachenblut: ?
12:44.22kpettitdlynes_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
12:44.29benjamin7062I feel so... validated!
12:44.30dlynes_officeLord_Drachenblut: oh....why not use fop?
12:44.56dlynes_officeLord_Drachenblut: I think fop will allow you to do all that
12:45.02dlynes_officeLord_Drachenblut: no point reinventing the wheel
12:45.20kpettitbenjamin7062, I feel guilty when I use SuSE, instead of a source distro.  Like I took the easy way out or something.  haha
12:45.21benjamin7062now 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)
12:45.24Lord_Drachenblutdlynes_office, there is a point but thanks for the help
12:45.29benjamin7062and windows has decent multi mon support
12:45.33*** part/#asterisk graphyx (n=mkling@24.144.57.161)
12:45.46benjamin7062don't get me wrong
12:46.05benjamin7062but just starts to die after a while.. no matter how much ram you throw at it
12:46.10kpettitbenjamin7062, 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
12:46.13dlynes_officekpettit: what source distros is there besides Gentoo, Sourcemage, and FreeBSD ports collection?
12:46.27benjamin7062x11vnc you mean?
12:46.39kpettitdlynes_office, I'm not sure, it seems to change daily.  Gentoo is what i use for work servers
12:47.00kpettitbenjamin7062, I thnk that's it.  it's just the keyboard and mouse that is shared
12:47.02dlynes_officekpettit: ok...was just making sure you weren't including slackware in that ugly list :p
12:47.24kpettiti haven't used that one in forever
12:47.24benjamin7062<-- still a debian vagina
12:47.28kpettithaha
12:47.39kpettitI like doing the Knoppix debian thing
12:47.50dlynes_officeyeah...Knoppix is cool
12:47.51kpettitI switch dekstop distro's every few months it seems.
12:47.56dlynes_officeit's my favorite rescue cd distro
12:48.05kpettitthe knoppix install is pretty sweet
12:48.09benjamin7062I'm so lazy... I compile production services always... but for tools.. I hate compiling.. (like tcpdump, less, etc)
12:48.15dlynes_officekpettit: i've never installed it
12:48.15*** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org)
12:48.20kpettitthen you have a pre-installed working system that you can apt-get install...
12:48.20dlynes_officekpettit: only ran it on the live cd
12:48.34kpettitI installed it and used it for a few months when the first DVD version came out
12:48.34tzafrirkpettit, if you refer to something you can use on a server, count knoppix out. You can use it, but not maintain it
12:48.39benjamin7062You guys know about systemrescuecd.iso right?
12:48.45kpettittzafrir, I'm just talking desktop
12:49.02kpettitbenjamin7062, Yeah I like that partionmagic clone on there.  that's pretty sweet
12:49.06dlynes_officebenjamin7062: no idea, but my main use for a rescue cd is for backing up windows filesystems that got hosed
12:49.30dlynes_officebenjamin7062: because somebody clicked "OK!  Clean the 56 viruses from my system!"
12:49.37benjamin7062hahahahha
12:49.43tzafririt'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
12:49.45benjamin7062I laugh.. because you speak truth
12:49.55dlynes_officebenjamin7062: i know i speak the truth
12:50.12dlynes_officebenjamin7062: that's a popup window from internet explorer asking you to click ok to install the latest spyware
12:50.12kpettitI did gentoo KDE desktops for all my sales guys
12:50.35kpettitdid the kororaa KDE install thing which worked out nice.
12:50.35Dovid_LaptopBesides for these lines what do I have to put in to meetme.conf  ?
12:50.35Dovid_Laptophttp://astcc.dovid.net/meet.txt
12:50.54benjamin7062I run etch on servers... instead of sid
12:51.12kpettitetch?
12:51.15benjamin7062OMG -- when they broke pam on unstable a while back... I wanted to spit
12:51.18benjamin7062etch = testing
12:51.21benjamin7062it's in the middle
12:51.22op3rdoes anyone know agentcallbacklogin here?
12:51.29benjamin7062stable is always 'OLD'
12:51.29dlynes_officeDovid_Laptop: none of those belong in meetme.conf; those are all for extensions.conf
12:51.38kpettitop3r, I wish, that's on my todo list
12:51.41Dovid_LaptopI know
12:51.46Dovid_LaptopIts in exten.conf
12:51.51benjamin7062op3r, yes, what do you want to know?
12:51.52kpettitop3r, http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+AgentCallBackLogin
12:51.53tzafrirDovid_Laptop, do you have an existing meetme root for every number that will get there?
12:51.55Dovid_LaptopMy question is what do I add to meetme.conf ?
12:51.56dlynes_officeDovid_Laptop: no such file in asterisk
12:52.12Dovid_Laptoptzafrir: can I pm ?
12:52.23tzafriryes
12:52.56kpettitbenjamin7062, 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)
12:53.19kpettitbenjamin7062, asterisk -rx "sip show peers" or basically any command just shows the first line
12:53.24dlynes_officelittleball: you had a problem earlier....i forget what it was
12:53.33dlynes_officelittleball: but you logged off before i had a chance to reply
12:53.34kpettitmaybe that's fixed now, but on all the 1.2.9's I have it has the problems.    kind of anoying
12:53.34littleballhi dlynes_office
12:53.39op3rbenjamin7062: because I am having this problem that the agent can login twice
12:53.41hwtis realtime stable enough for production use now? roughly 1000 users.
12:53.56op3rbenjamin7062: even if he is already logged in on another extensions
12:54.08littleballhi dlynes_office, i forgot the problem, maybe it is related to "hint" prioroty
12:54.10kpettitop3r, if they sit at the same phone I'd hardcode the sip peer to be logged into the queue.
12:54.19kpettitop3r, agents have a way of buggering uip the login
12:54.20dlynes_officelittleball: oh yeah...that's what it was
12:54.30dlynes_officelittleball: what problem exactly were you having?
12:54.32littleballcan 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
12:54.38benjamin7062kpettit, hrmm?  I get all lines?  weird?
12:54.45op3rkpettit: so the best way is to hard code the phones to queues?
12:54.47dlynes_officegood morning, anthony
12:55.00kpettitbenjamin7062, which release do you have?
12:55.05op3rkpettit: because need to be able to monitor the agent login logout using qmetrics
12:55.12trelane_what are the general thoughts on asterisk and preempt? does the lower latency from preempt help?
12:55.13benjamin7062kpettit, 1.2.9.1
12:55.16littleballi just don't understand when i read http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
12:55.18anthmhi
12:55.20kpettitop3r, in queues.conf you specify
12:55.20*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
12:55.30dlynes_officelittleball: what don't you understand?
12:55.34kpettitop3r, pm me and i'll paste you a example
12:55.43nfi|ermeswhere can i read about spacial characters, like "_" "."  ecc
12:55.47nfi|ermes?
12:56.02benjamin7062kpettit, do agents only have one station?
12:56.11littleballdlynes_office, start from the begining what is the purpose of "hint"?
12:56.19kpettitop3r, basically in queues.conf just define   member => Sip/206
12:56.21nortexStrange 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?
12:56.23benjamin7062sorry.. op3r do agents only have one station
12:56.27dlynes_officelittleball: blf (busy lamp field)
12:56.40dlynes_officelittleball: well, presence detection
12:56.47op3rbenjamin7062: 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)
12:56.49dlynes_officelittleball: in sip's case, it's called BLF (busy lamp field)
12:56.58dlynes_officeerm presence awareness
12:56.59littleballexten => 200,hint,SIP/phone1, in this case, how to understand this
12:57.08benjamin7062op3r, right.. but do they have ONE desk...
12:57.14benjamin7062like, is a person assigned to a desk?
12:57.16*** join/#asterisk Lord_Drachenblut (n=Lord@74.129.228.28)
12:57.24littleballdlynes_office, in jabber, it is called presence, in SIP, it is called BLF?
12:57.31op3rbenjamin7062: nope its first come first serve
12:58.02dlynes_officelittleball: 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)
12:58.14trelane_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
12:58.33dlynes_officelittleball: it also means if someone hits the button associated with that busy lamp field, it'll dial SIP/phone1
12:58.47littleballdlynes_office, ok. understood this.
12:59.23littleballdlynes_office, next question, subscribecontext=
12:59.26benjamin7062op3r, perhaps you should have them auto log them out before logging them in?
12:59.32littleballwhy subscribecontext=  is needed?
12:59.38benjamin7062even if they aren't logged in
12:59.43*** part/#asterisk oscarh (n=oscar@host-87-74-0-243.bulldogdsl.com)
13:00.07dlynes_officelittleball: 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)
13:00.08benjamin7062op3r, run the logout prior to login... for your login extension
13:00.21op3rhmm
13:00.24benjamin7062op3r, not sure if that would work...
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13:00.27benjamin7062but maybe
13:00.33dlynes_officelittleball: subscribecontext is needed so that asterisk knows which dialplan context to look in for hint subscriptions
13:00.50benjamin7062I'll try real quick
13:00.50op3rbenjamin7062: can i pm you my agentcallbacklogin info from my extensions.conf?
13:00.50hwtare there any good web-based management interfaces for realtime?
13:00.59op3rhwt: fop?
13:01.04benjamin7062one sec
13:01.09benjamin7062let me check if this even works
13:01.15dlynes_officeop3r: fop isn't a realtime web based management interface
13:01.17hwti don't want to use A@home or AMP.
13:01.31dlynes_officehwt: fop doesn't require A@home or amp
13:01.38hwtop3r: no, to edit dialplans, extensions, sip users, etc.
13:01.39dlynes_officehwt: but it's not a realtime management interface, either
13:01.47hwtdlynes_office: yup, i know.
13:01.49littleballdlynes_office, let me think first. not understand this yet
13:02.05*** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au)
13:02.28kpettitop3r, check out my agent logoff, it's the ugliest logoff hack ever.
13:02.32kpettit<PROTECTED>
13:02.44nortexStrange 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)
13:03.14littleballdlynes_office, can u describe how a sip phone can subscribe another sip phone's presence
13:03.15op3rhwt: check out the php edit config that was used by trixbox
13:03.17littleball?
13:04.10littleballdlynes_office, the whole flow is important for understanding the whole sip presence thing. :-)
13:05.32benjamin7062kpettit, actually -- that's what I do too... since there is no way to auto log someone off
13:05.34dlynes_officelittleball: one sec...just bringing up a page
13:05.40littleballthanks
13:06.34xxttxxttxxttHey dlynes_office , Do you know where I could insert the IVR into my dial plan ?
13:07.01dlynes_officelittleball: try this link...it's a wiki on there that I've started but haven't had a chance to finish yet
13:07.04dlynes_officelittleball: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+subscribecontext
13:07.22dlynes_officelittleball: it involves a few screenshots of Aastra 9133i's for setting up SIP presence (BLF's)
13:07.30dlynes_officexxttxxttxxtt: wherever you want
13:07.51littleballthanks
13:07.58hwtop3r: so trixbox is the new name of AMP?
13:07.58xxttxxttxxttdo i need ne extra commands
13:08.08kpettitbenjamin7062, that's funny, I thought that was soo ugly
13:08.11dlynes_officehwt: new name of Asterisk@Home
13:08.17hwtdlynes_office: ah, right.
13:08.19op3rhwt: no its the new name of A@H
13:08.22dlynes_officehwt: FreePBX is the new name of AMP
13:08.29hwtit's probably easier to just write something yourself.
13:08.31benjamin7062kpettit, really, it's the only way to auto do it.. or create an AGI that calls the AMI
13:09.07kpettitit's just funny you can't code that directly I think
13:09.32*** join/#asterisk }btorch{ (n=btorch@208.63.19.179)
13:09.48benjamin7062op3r, 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)
13:10.28benjamin7062kpettit, the person who coded it had a one track mind.. only invisioned people would want to do the login/logoff via phone... not dialplan
13:11.03benjamin7062kpettit, same thing; you can't log someone in automatically without prompting for a password
13:11.38benjamin7062I understand you may want security but it should be 'my' decision .. not set in stone... but there are always hax around the system
13:11.40benjamin7062=)
13:11.41*** join/#asterisk feld_ (n=feld@12.148.212.157)
13:11.54benjamin7062like a custom perl app that calls the AMI
13:11.58*** join/#asterisk Katty (n=aisaacs@64.82.232.54)
13:12.31Kattyhey dlynes (=
13:12.38op3rbenjamin7062: nice
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13:12.46benjamin7062op3r, work for you?
13:13.16op3rbenjamin7062: 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
13:15.24kay2if 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 ?
13:15.50SheriF_WorKi want to understand something about buying the G729 codecs what dose it mean per channel ?
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13:16.40dlynes_officealucard064: you still there?
13:16.43}btorch{any way to do that ?
13:18.24benjamin7062kay2, perhaps transfer them both to a meetme?
13:19.48dlynes_office}btorch{: maxgreet, saycid=no, sayduration=no, saydurationm=5, envelope=no
13:20.05dlynes_office}btorch{: check your sample voicemail.conf file for more info on those five options
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13:23.40Kattyariel_: (=
13:24.05X-Genis there a chan skype yet ?
13:24.09X-GenSkype-to-Asterisk
13:24.17boddyhii all I configure extensions conf --> _1XXX,1,Dial,Zap/1/${EXTEN}
13:24.27boddy_5XXX,1,Dial,Zap/1/${EXTEN}
13:24.49znoGquestion: 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
13:24.55boddyI can call telephones that begin with 1
13:25.02boddybut I cant 5
13:25.29Kattyare they 4 digits long, starting with 5?
13:25.43ariel_Katty, morning
13:25.49Kattyariel_: how's it goin, hun?
13:26.01}btorch{dlynes_office, thanks
13:26.08ariel_ruff been very busy... but not enough income.. normal
13:26.11Kattyhey iDunno (=
13:26.12boddyKatty you sayin me ?
13:26.19Kattyariel_: aww, i'm sorry to hear that )=
13:26.38Kattyariel_: stuffs a little tight around here too...windshield got cracked beyond repair.
13:26.49Kattyariel_: and my license renewal is up soon, so it had to be replaced right away ;)
13:27.11Kattyboddy: 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)
13:27.38PersilonHi
13:27.42benjamin7062boddy, do you have anything else in your dial plan that could possibly be catching _5XXX first?
13:27.44boddyI just want redirecet telephon numbers start with 5 to zap
13:27.46ariel_Katty, well in my state windshields are required to be changed and paid for by the insurance co. without any deductable to the driver.
13:28.11a20060628how can i find the dial command result?
13:28.27Kattyariel_: wow, that's pretty nice.
13:28.35Kattyariel_: i have glass coverage, but my deductable is 500.
13:28.42kpettitI'm tryijng to parse through a cdr record to see how many voicemail's were left for a extension
13:28.43PersilonI'm having troubles with Playtones(), I'm building an ivr and it only works whitin another extention, not on the ivr extention
13:28.46ariel_You should check with your state most of them have the same rule.
13:28.53Kattyariel_: i already did check.
13:28.55littleballanyone using firefly? i can call out from firefly, but cannot call to firefly. everything works for xlite
13:28.58kpettitany 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.
13:29.00ariel_argh
13:29.02Kattyariel_: i'd have to pay 500 to get a 200 job done ;)
13:29.09Kattyariel_: and it's already taken care of now anyway.
13:29.13kpettitGrepping for the voicemail extension in Master.csv dosen't seem to do the trick
13:29.14ariel_I see
13:29.26boddy?
13:29.32nortexStrange 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?
13:29.40benjamin7062kpettit, 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)
13:30.13benjamin7062kpettit, as an alternative
13:30.52kpettitThis 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.
13:30.57kpettitnot sure the best way to go about that.
13:31.19kpettitI 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.
13:31.20Kattyariel_: you any good with mounting win2k3 shares? (smbfs)
13:31.50creadurxis there anyway i can tell my ip10s SIP phone to answer a channel that is ringing? via the manager interface..
13:31.51ariel_you mean setting up samba on a linux box?
13:31.58Kattyariel_: nono.
13:32.05Kattyariel_: it technically /mounts/
13:32.10Kattyariel_: but gives me errors about stale nfs
13:32.29a20060628can u help me, pls ?
13:32.31ariel_I don't deal much with that...
13:32.33*** part/#asterisk jpeeler (n=thepeel1@host81-149-2-72.in-addr.btopenworld.com)
13:32.37Kattyariel_: m'kay.
13:32.45Kattyariel_: i'll find someone else to pester, don't worry ;)
13:32.52nortexKatty, Is it still not working?
13:32.56a20060628how can i find the asterisk dial command result
13:32.59Kattynortex: no, no it's not.
13:33.00a20060628???
13:33.11nortexKatty, but it did mount
13:33.16Kattynortex: still at the same point i was yesterday. mounts but gives me a stale nfs error on dir
13:33.21Kattynortex: it /has/ mounted.
13:33.22ariel_a20060628, more info is needed. I don't understand
13:33.27Kattynortex: always has.
13:33.31Kattynortex: but just not properly.
13:33.38nothinmanguys, 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? :/
13:33.58Kattyanthm, dear, are you around?
13:33.59benjamin7062kpettit, 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)
13:34.11anthmyes hon
13:34.21a20060628in php i use the command exec("dial","") ..., and i want to know what is the result of this
13:34.38kpettitbenjamin7062, I can see calls in the cdr but I don't know how to verify if a voicemail was left
13:34.41a20060628somebody answered, busy somthing like that
13:35.08Kattyanthm: ever heard of a "stale nfs file handle" when you mount a win2k3 share?
13:35.22anthmyah
13:35.36ariel_a20060628, there is very good dialparties.agi out there on the wiki that has allot of info about what your looking for.
13:35.38Kattyanthm: can you put that into kat for me so i can understand what it's trying to tell me?
13:35.41benjamin7062kpettit, 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)
13:35.46BertZHmm
13:35.49MACscrhello everyone
13:36.22anthmit means the nfs link was lost while something was using the file
13:36.28MACscranyone know of a managed asterisk hosting provider?
13:36.29a20060628ariel_ can you send me a link or something ?
13:36.40BertZI'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 ?
13:36.41anthmand windows is prtty lame about letting go of file locks
13:36.46Kattyanthm: hmm.
13:36.47*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
13:36.48ariel_~google dailparties.agi
13:36.57a20060628thanks
13:37.08Hmmhesaysoh man these e2's kickass
13:37.10anthmso you gotta close every application that may have been using a file on that
13:37.18anthmbefore it will let you mount it again
13:37.21Kattyanthm: 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.
13:37.25anthmor reboot the box
13:37.32Kattyanthm: oh ah.
13:37.36Kattyanthm: that's something to try. i'll do ti.
13:37.45anthmnfs over windows is a pita
13:37.46Hmmhesays$100 bucks a set then damn well better be kickass
13:37.59Kattyanthm: i'll give that a try. thanks for the heads up hun
13:38.03anthmthey usually mount nfs as a weak user
13:38.05BertZ~queues
13:38.09*** join/#asterisk m4rk___ (i=mark@mir.stevecole.org)
13:38.12BertZ~queue
13:38.14jbotInnovative load-balancing/batch-processing system and rsh replacement. URL: http://bioinfo.mbb.yale.edu/~wkrebs/queue.html
13:38.14ariel_Hmmhesays, 2e's
13:38.27anthmyou 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
13:38.29HmmhesaysShure E2 ear monitors
13:39.04benjamin7062anthm, not so bad if you get smb to auth using secure=domain or whatever...
13:39.05Kattyanthm: 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)
13:39.26benjamin7062anthm -- don't remember exactly but if it auths against AD basically
13:39.29anthmso you are using the windows disk from unix then ?
13:39.31Kattyanthm: still trying to hash out the problems... i simply presumed it was a win2k/2k3 difference in permissions or something.
13:40.15Kattyanthm: 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
13:40.15ariel_Hmmhesays, expensive ear devices.
13:40.28Hmmhesaysariel_: well worth it
13:40.36anthmyah it's gonna be 2k3 being lame
13:40.39anthmin that case
13:40.48Kattymew )=
13:40.51m4rk___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?
13:41.08m4rk___what i mean is: can i just do a Dial() and have asterisk handle passing on the call progress
13:41.21cef_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)
13:41.30anthmi was dumb enough to use 2k3 back in 2k3 so i havent looked at it much since
13:41.38Kattyanthm: 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.
13:41.45dlynes_officealucard064: anyways...if you're still there, try these links:  http://www.ipdanmark.dk/ and http://www.ipdanmark.dk/IPSwitchBoard/IPswitchBoard%20Manual.pdf
13:41.50redaxhi.
13:41.57cef_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)
13:42.31cef_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
13:42.37Kattyanthm: i'm really getting sick of 2k3...but the company wants me to keep using it since i'm mcped on 2k3 stuff
13:42.40redaxwhat's wrong with this: s,2,Dial(mISDN/g:Intern/101) ; s,3,Dial(mISDN/g:Intern/102)
13:42.48Kattyanthm: i appear to be doomed ;)
13:42.53redaxit should ring the ext 102, if the 101 is busy, right?
13:43.14dlynes_officecef_: i've only found that to be the case when you have canreinvite=no
13:43.36dlynes_officecef_: oh...nvm
13:43.51dlynes_officecef_: you're actually able to convert...it just doesn't sound right
13:44.09dlynes_officecef_: the problem i was thinking of was where it doesn't autonegotiate codecs
13:44.18cef_yup.. i don't think that's my problem
13:44.49dlynes_officeand anthm will probably say but freeswitch does that, no problems :)
13:44.51cef_asterisk seems to know what i'm trying to do a 'show channel xxx' indicates the correct info including transcoding
13:45.08dlynes_officeor not :)
13:45.55cef_i verified this works fine in the branches/1.2 release
13:46.03Persiloncan anyone help me with playtones ?
13:46.04cef_it only brakes in the trunk
13:47.31redaxor should I use the "j" parameter in Dial() to jump +101 if the Dialed party is busy ?
13:48.21littleballhello, who knows how to force an sip peer expire in asterisk CLI console?
13:49.02*** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com)
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13:50.46znoGare my messages too long to bother reading or nobody has any suggestions?
13:51.31fulgashi
13:52.19Lord_Drachenblutwhere can i find info on callfiles at
13:52.52ariel_znoG, ???
13:53.07*** join/#asterisk Modcuts (n=bob@lan.proporta.com)
13:53.36ManxPowerLord_Drachenblut, Google the mailing lists, the Wiki, sample.call in the Asterisk source
13:53.43znoGariel: 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)
13:54.16ManxPowerznoG, I think it's so location specific that we reallly can't tell you.  your idea makes some sense.
13:54.21asteriskmonkeyanyone know of some asterisk mail client software
13:54.27nortexCan some one help me diagnose why some polycom phones are auto answering.
13:54.35ariel_znoG, you need what I call a rollover macro
13:54.40*** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au)
13:54.45ManxPowernortex, no.  Polycoms do not auto answer unless you tell them to.
13:54.56ManxPowerthe Wiki and the mailing list archives have info on making them autoanswer
13:55.15ariel_ManxPower, morning
13:55.17asteriskmonkeyavc was a good stand alone java client for asterisk but is no longer made
13:55.17nortexManxPower, Oh, but they do in a queue with a sipheader.
13:55.41ManxPowernortex, um, adding a SIP header would be considered "telling them to"
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13:55.43znoGariel_: right, which would do more or less what I proposed?
13:56.41ariel_znoG, do you have one you need help with?
13:57.23nortexManx 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.
13:57.33znoGManxPower: 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.
13:57.47znoGariel_: not yet, haven't really implemented the idea, yet.
13:57.51*** join/#asterisk qdk (n=qdk@213.237.44.34)
13:58.18znoGariel_: 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.
13:58.34ariel_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
13:58.58ariel_znoG, use group counts in a macro
13:59.04ariel_you don't need an agi for it.
13:59.23*** join/#asterisk DrkShdw (n=DrkShdw@fl-209-26-20-205.sta.embarqhsd.net)
13:59.33ManxPowerznoG, we don't let our users forward to internal extensions.  End of problem
14:00.12ManxPowernortex, what makes you think the Queue app even sends the header you are setting?
14:00.17ManxPowerI thought only DIAL did that.
14:00.25znoGariel_: interesting, time to go to the url you pasted
14:00.30znoGariel_: thanks
14:00.38nortexznoG, I have 3 receptionist all in a queue with ring-all.
14:00.59ManxPowerQueue looks smart.  It is not.
14:01.06nortexManxPower, Becuase the ring tone is changed on the phone.
14:01.21znoGnortex: yeah, that could work I guess
14:01.25ManxPowernortex, then perhaps the PHONE config is different.
14:03.23nortexManxPower, 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.
14:04.19nortexManxPower, 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.
14:04.21ManxPowernortex, 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?
14:04.25iqGood Morning
14:04.35Kattyhey iq (=
14:05.06iqhey Katty - whats up
14:06.36nortexManxPower, 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)
14:07.03Hmmhesaysgxp-2000 thoughts?
14:07.05nortexManxPower, I mean the same sip.cfg (Polycom file)
14:07.08nXORhi ppl is someone here to help me with a little problem
14:07.21nXORi knwo its prolly pebkac but i need enlightning on the matter
14:07.26ManxPowernortex, Well, make sure all phones are loading that file, yes
14:07.41ManxPowerCustomer: We are on our way to Jackson MS, will you be available in 2 hrs?
14:07.51ManxPowerIt 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)
14:08.11dlynes_officeManxPower: don't you mean more than?
14:08.32ManxPowerdlynes_home, Correct.  Apparently I didn't have enough coffee yet.
14:08.37Kattyiq: nada, just trying to get some work done ;)
14:08.43dlynes_officeManxPower: and i've had too much
14:08.47*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
14:08.51dlynes_officeManxPower: 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
14:09.03nXORi 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
14:09.08iqKatty: :)
14:09.11dlynes_officeGood morning, Russell!
14:09.16ManxPowerdlynes_home, I don't do that anymore without large wads of cash in hand.
14:09.19nXORand my soft client shows NAT/Firewall of an unknown type
14:09.20russellbgood morning :)
14:09.36nXORany pointers would be much appreciated
14:09.41ManxPowernXOR, you want to tell the SIP client there is no NAT.
14:09.49ManxPowerThen use asterisk's built in NAT features
14:09.52nXORManxPower,nat=no ?
14:10.09nXORManxPower, shed some light on this please
14:10.18ManxPowernXOR, I don't know how to configure your softphone.  All I can tell you is how to configure Asterisk.
14:10.24nXORtell me that
14:10.34nXORor at least point me to a resource
14:10.43ManxPowernXOR, Is Asterisk behind NAT?
14:10.51nXORwell no not really
14:10.59*** join/#asterisk [TK]D-Fender (n=joe@CPE000d3a2c3061-CM00080d8dba84.cpe.net.cable.rogers.com)
14:11.02nXORits like this --isdn line ----- asterisk box ----- lan
14:11.11jbalcomb~websae
14:11.15ManxPowernXOR, NAT is like being pregnant.  You can't be "sort of"
14:11.17jbalcomb~seen websae
14:12.00jbotwebsae is currently on #asterisk (37m 52s), last said: 'get a VoIP provider and ATA/SIP phone'.
14:12.02nXORmaagic,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)
14:12.02jbalcombnXOR NAT or PAT?
14:12.02nXORManxPower,the asterisk box taps directly to an isdn line via an TA
14:12.03ManxPowernXOR, and ALL phones and softphones are on the LAN
14:12.03nXORManxPower,indeed
14:12.04*** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net)
14:12.05nXORjbalcomb, neither for asterisk box
14:12.06DandreHello,
14:12.08ManxPowernXOR, then you do not have NAT.
14:12.12nortexManxPower, 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)
14:12.27ManxPowernortex, correct.
14:12.32nXORManxPower, so everythign should work ?
14:12.51nXORbut somehow it doesnt ...... maybe its the TA ......
14:12.54DandreI need some help with grandstream bt100 firmware update. Is there any one here that culd help me?
14:12.56ManxPowernXOR, Assuming you have all the other billion things that need to be set up correctly.
14:12.58jbalcombnortex: you have your system set up to ring distinctively per queue?
14:13.00nXORbut ui see visdn raising up the visdn0 interface
14:13.21nXORthen closing it .....
14:13.30jbalcombDandre: No one can know until you tell us what the problem is.
14:13.32ManxPowernXOR, So you have Softphone makes a call -> Asterisk -> HOW DOES ASTERISK DIAL THE PSTN?
14:13.56nXORextensions.conf has the dialplan
14:13.57ManxPowernXOR, looks to me like you have an asterisk/ISDN problem that has nothing to do with the phones.
14:14.24ManxPowernXOR, and I can't help you with ISDN since I'm in the USA.
14:14.25nXORyeah i think so too, but i cant really pin point it, do you knwo of any good visdn + asterisk tuts ?
14:14.40nortexjbalcomb, I only have one queue, but that is what I hoped for.
14:14.50*** part/#asterisk m4rk___ (i=mark@mir.stevecole.org)
14:15.00nXORi cant fidn any good ones online, ones that i find are scarce
14:15.21Dandrejbalcomb: 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
14:15.22ManxPowernXOR, No.  I'm in the USA.  I think there are 3 people in the entire country that have any kind of ISDN BRI.
14:15.36eleslo all, does anyone know if it is possible to transfer a call in progress to another extension via console ?
14:15.38nXORshould i forward traffic from lan1 to visdn with iptables ?
14:15.45DandreI have a blank
14:15.48Dandrepage
14:15.53littleballhello, for sip, CLI#sip show peers. what is the difference for status : UNKNOW and UNREACHABLE? what packets change the status from "UNREACHABLE" to "OK"?
14:15.53ManxPowereles, you cannot.
14:16.13ManxPowereles, you can via the manager interface (aka AMI)
14:16.23nortexjbalcomb, 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.
14:16.50ManxPowerlittleball, unknown = phone is not registered.  Unreachable = phone was registered, but now it's not responding,  OK = phone is registered and is responding
14:17.11anthmeles, app_changrab from http://www.pbxfreeware.org
14:17.17ManxPowerunrechable and OK only happen when you use the qualify= option
14:17.31littleballManxPower, how to define "Unreachable" and not responding? Responding to what?
14:18.07ManxPowerlittleball, not responding to the SIP OPTIONS packet Asterisk sent
14:18.31littleballManxPower, thanks. I think the same and let me test now
14:19.37elesManxPower: know where i can maybe get a reference or some documentation for the AMI ? I have tried looking but havent really found anything nice
14:19.41jbalcombnortex: 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?
14:19.43elesanthm: tanks i will take a look
14:20.04ManxPowereles, that would be on the Wiki and in the Asterisk source, prolly /path/to/src/asterisk/docs
14:20.10littleballManxPower, i found that OPTIONS from asterisk to peer packet has the same CSeq: 102
14:20.11jbalcombDandre: Does the phone work otherwise? Have you port scanned the phone to see if anything is responding?
14:20.14littleballwhy?
14:20.57littleballManxPower, 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
14:21.02Dandrejbalcomb: the phone seems to work as before but doesn't register to my asterisk box
14:21.36anonymouz666anthm: lots of apps in that site
14:22.33nortexjbalcomb, 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.
14:22.41anthmyah that's where i keep my out of tree apps
14:22.41*** join/#asterisk snowy_owl (i=0@200.218.196.2)
14:23.18nortexjbalcomb, I can pastebin what I have at this point if you want.
14:23.24snowy_owlThose who know me have no need of my name
14:23.26jbalcombnortex: Hrmm.. can I get a copy of the relevant configs?
14:24.31jbalcombnortex: yeah, a pastebin would be great
14:25.00*** join/#asterisk anna-- (n=nmuller@195.70.21.58)
14:27.40anonymouz666snowy_owl: that's for sure
14:29.19jbalcombnortex: if pastebin.com is still down you can use http://sial.org/pbot/
14:30.48Dandrejbalcomb: only 80 and 4144 ports respond to a port scan
14:31.48*** join/#asterisk a1fa (n=a1fa@207.210.210.202)
14:31.49a1fayo
14:32.11a1fai found a site a month ago that offered $1/did + $0.01 per minute
14:32.20a1fa25 calls max at one time
14:32.34a1faanybody know the site i am talking about
14:33.11*** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca)
14:33.57MACscralfa, you might want to check your browser history or next time bookmark it =P
14:34.42dlynes_officeMACscr: dood...don't be making sense...that's not allowed
14:35.01MACscrlol, sry =P
14:35.17a1falol
14:35.23a1fanah.. 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)
14:39.05MACscryou cant search your browser history?
14:40.06MACscrdlynes: you seem pretty logical, have you heard of anyone offering asterisk hosting and management?
14:40.20MACscrbasically "managed asterisk hosting"
14:40.22a1faMACscr : this was 2 months ago
14:40.28a1faMACscr : i do
14:40.33MACscrgoogle hasnt turned up anything
14:40.36a1faMACscr : $5/month
14:40.45a1famsg
14:40.47*** join/#asterisk mut (n=animenod@65.111.222.120)
14:40.57dlynes_officeMACscr: nope
14:41.09*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.94.Dial1.SanJose1.Level3.net)
14:41.15dlynes_officeMACscr: 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)
14:41.27jbalcombDandre: does the phone work?
14:41.30dlynes_officeMACscr: 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)
14:42.11Dandrejbalcomb: it doesn't register to my asterisk box but I can place calls
14:42.21littleballwhat is xen?
14:42.34jbalcombXen is a virtual server technology
14:42.35dlynes_officelittleball: virtual machines
14:42.48littleballok
14:43.00MACscrmy issue is that i dont have the time or patience to setup asterisk they way i want it
14:43.09dlynes_officelittleball: as is uml and vps
14:43.19jbalcombMACscr so just contract [TK]D-Fender to do it for you
14:43.22MACscrand im needing to custom of a configuration to do any vanilla service
14:43.22dlynes_officelittleball: user mode linux, and virtual private server
14:43.50dlynes_officeMACscr: why do you need to custom configure everything?
14:44.30jbalcombDandre: I assume you have rebooted the phone since?
14:44.40Dandresure
14:44.48littleballdlynes_office, does it mean xen runing on a few clustered hardware?
14:44.57dlynes_officelittleball: not necessarily
14:45.03dlynes_officelittleball: could be all on one machine
14:45.04*** part/#asterisk bkw_ (n=bkw_@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net)
14:45.07MACscrwell, 1) im running multiple companies with the same staff, which the staff are remote
14:45.18dlynes_officeMACscr: ok
14:45.24dlynes_officeMACscr: and?
14:45.27littleballdlynes_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.
14:45.32MACscrso i need caller id rewritten so the staff member knows what company he is answering for
14:45.38MACscri also need to record all the calls
14:45.49*** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl)
14:45.49dlynes_officeMACscr: how many different companies?
14:45.55MACscronly 2 right now
14:46.06MACscrand only a handful of extensions
14:46.07jbalcomblittleball: Xen allows you to run multiple servers on one piece of hardware
14:46.07dlynes_officeMACscr: do you expect it to be more than three?
14:46.11MACscras in 6 at most right now
14:46.21dlynes_officeMACscr: i.e. more than three companies?
14:46.25MACscractually i only have 2 for sure
14:46.28MACscrnot right now
14:46.34MACscrbut i definately see it happening
14:46.44dlynes_officeMACscr: you can handle that with three lines if you want, too
14:46.48jbalcombMACscr: I have 5 companies on my system but only 7 remote users. Also, have two call centers with 17 call queues
14:46.56dlynes_officeMACscr: one company per phone line
14:47.06dlynes_officeMACscr: so when you see line 1 ringing, you know it's for company #1
14:47.09MACscrright, i was planning on going with 3 lines
14:47.20MACscr2 lines for 2 companies and 1 for fax
14:47.36dlynes_officeMACscr: a line on a voip phone is not the same as an analog line
14:47.39littleballjbalcomb, 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
14:47.44dlynes_officeMACscr: it's not a 1:1 mapping
14:47.44jbalcombdlynes_home MACscr: Why different lines? Why not just different phone numbers and match the DID?
14:48.24dlynes_officejbalcomb: 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
14:48.32MACscrsry, meant lines
14:48.34MACscrwoops
14:48.37MACscrnumbers i mean
14:48.37jbalcomblittleball: 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-FenderMACscr : Have you picked your phone already?
14:48.59littleballjbalcomb, yes.cater different requirement
14:49.07MACscrFender, i went cheap and got a GXP-2000 for myself
14:49.16jbalcomblittleball: 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.
14:49.20littleballjbalcomb, reliable?
14:49.22MACscrbut 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-FenderMACscr : 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
14:49.49a1fajbalcomb : xeon is good
14:49.56a1fajbalcomb : i got couple of xeon servers
14:50.09a1fa<PROTECTED>
14:50.20jbalcombdlynes_office MACscr: yeah but you can do that based on the DID just as easy as line
14:50.25dlynes_officejbalcomb: you can do all that by setting four instances each of each server, listening on different interfaces, too
14:50.32littleballalfa, jbalcomb, is it reliable? and is it compatible with existing linux applications?
14:50.32peterm22anyone using the new sipura one phones ?
14:50.34a1faof course
14:50.36a1fait is i386
14:50.38[TK]D-FenderMACscr : Line 1-2 = Company A, 3-4 = Company B
14:50.51dlynes_officejbalcomb: yeah...I was just thinking doing it by the lines is less kludgy
14:50.56*** join/#asterisk N3WWN (n=N3WWN@ns1.futuretek.cx)
14:51.02dlynes_officejbalcomb: then the caller id is left untouched
14:51.05littleballalfa, you mean the virtuam server is exactly one i386 box?
14:51.16a1fawell yeah
14:51.21jbalcombdlynes_office: but if you do it based on DID then each company can use all the lines
14:51.22a1faits like running wmvare on your box
14:51.29a1favmware*
14:51.36a1fasame thing
14:51.48a1faon a 3ghz zeon with 10gb of ram
14:51.49*** join/#asterisk mogorman (n=mogorman@gateway.digium.com)
14:51.50nortexjbalcomb, http://pastebin.com/734897
14:51.56a1fayou can run 20 virtual servers if not more
14:52.00[TK]D-Fenderjbalcomb : not by PSTN lines, but PHONE "lines" or "appearances"
14:52.13jbalcomblittleball: 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.
14:52.13dlynes_officejbalcomb: you've lost me...how is doing it by did, not modifying the callerid for incoming calls?
14:52.28nortexjbalcomb, I just had it work on a fresh phone.
14:52.40jbalcomb[TK]D-Fender: ah, guess I'm tryign to do too much at once to keep it all straight.
14:52.49dlynes_officejbalcomb: or are you getting call appearances and pstn lines mixed up?
14:52.51littleballalfa, how about hardware resource like udp/tcp port?
14:53.12a1falittleball : depends, you can assign them each individual ethernet
14:53.22Dandrejbalcomb: I have put 1.0.8.12 version on my tftp server and now I have recoverd admin console but it shows:
14:53.24DandreProgram-- 1.0.6.8    Bootloader-- 1.0.8.9    HTML-- 1.0.6.8    VOC-- 1.0.1.0
14:53.25littleballalfa, true
14:53.27a1faor you can use 1 ethernet -> nat/routing ->
14:53.36dlynes_officeMACscr: these are all inbound calls, right?
14:53.38a1faDandre : reboot your phone again
14:53.45a1faDandre : 1.0.8.23 is out
14:53.57a1faDandre : i hate grandstream.. so many issues with their firmware
14:54.04littleballanyone 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)
14:54.15littleballs/reply/response
14:54.31znoGs/response/respond
14:54.37znoG:)
14:54.40littleballhehe
14:54.41Dandrebut I have had issues with the admin console with 1.0.8.23: I had no access to it
14:54.57a1fathey dont recomend you to downgrade
14:54.58MACscrdlynes: 98% of traffic will be inbound
14:55.19littleballznoG, so, other peer cannot call Firefly if it is behind firewall
14:55.22N3WWNAnyone having "channel.c:787 channel_find_locked" errors with 1.2.8 and the vpb channel driver?
14:55.28dlynes_officeMACscr: so you don't mind prefixxing outbound calls with a certain digit to differentiate between different phone lines then, right?
14:55.39jbalcomba1fa: are you the guy that was working on an auto-provisioning system for IP phones?
14:55.54dlynes_officeMACscr: i.e. to force asterisk to dial out on a specific analog line?
14:56.16MACscrdlynes: i dont see that being an issue, but im also not going to be using any analog lines
14:56.16a1fajbalcomb : no
14:56.25dlynes_officeMACscr: oh, ok
14:56.36[TK]D-Fenderdlynes_office : Again not needed with seperate call appearances.
14:56.37dlynes_officeMACscr: what kind of lines will they be then?  pri, or voip?
14:56.47MACscrsry, i come from a PTSN background when it comes to phone systems
14:56.59a1fajbalcomb : i did upgrade firmware on my remote locations in austria and bosnia :)
14:56.59MACscrso i apologize if i dont use the right logic sometimes
14:57.06[TK]D-Fenderdlynes_office : My home IP 501 has 1 line key / customer of mine I'm working with actively supporting 4 calls at a time each.
14:57.08littleballwho can recommend a cheap but good :-) router to seperate voice traffic and data traffic in the office?
14:57.26a1fajbalcomb : both phones were behind two firewalls and two nats.. nasty, but i was able to upgrade them remotely
14:57.28dlynes_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)
14:57.51dlynes_office[TK]D-Fender: but at the same time, sometimes you just want to grab the first available line, too
14:58.01dlynes_office[TK]D-Fender: and you don't care which one it is
14:58.07*** join/#asterisk masonf (n=masonf@dungle.vineyard.net)
14:58.09dlynes_office[TK]D-Fender: other times you might want to grab a specific line
14:58.11a1faintelligent call routing
14:58.24masonfdid asterisk cvs change recently?
14:58.25dlynes_officea1fa: yes, i already use that
14:58.29a1faif cost <> "" { use }
14:58.41dlynes_officea1fa: not an issue for cost
14:58.50a1fawell, availability and realibility
14:58.51dlynes_officea1fa: for an issue of which did you want showing up on the customer's display
14:59.00[TK]D-Fenderdlynes_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)
14:59.08dlynes_officea1fa: so that they know which business you're calling from
14:59.13tzangercool, there's a virtual tour of my house
14:59.20a1fadlynes_home: callerid not enuff?
14:59.27tzangerhttp://www.venturehomes.ca/VirtualTour.asp?TourID=5956
14:59.33littleballwho can recommend a reasonable router to seperate voice traffic and data traffic in the office?
14:59.36dlynes_office[TK]D-Fender: yeah, but if you're using all analog lines, and you cannot set your callerid, then it is an issue
14:59.50a1falittleball : cheap?
14:59.53a1faor good?
15:00.02littleballreasonable, alfa
15:00.12dlynes_officelittleball: any router that supports vlans
15:00.13a1fano such thing :) sacrifice something
15:00.25a1faany router that, well routes
15:00.26littleballthe budget is below 500USD
15:00.31a1falittleball : hahah
15:00.39a1falittleball : you can use QOS if you need to
15:00.51a1farouting+qos via packetshaping router
15:00.52dlynes_officea1fa: qos doesn't separate it...it only controls it
15:00.57littleballsmall office (30 persons). i need to improve the voice quality
15:01.10a1fadlynes_home: right, qos to improve pass-through
15:01.13nortextzanger, I have in-laws in ontario whose house is almost identical, must be a Canadian thing.
15:01.21a1fayou can use anyswitch that supports VLANS
15:01.23dlynes_officelittleball: qos to control it, vlans to separate it, and sip jitterbuffer if you've got huge files going across
15:01.27a1fayou dont even need routers
15:01.39*** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au)
15:01.50[TK]D-Fenderdlynes_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
15:01.50a1fadlynes_home: he wont need a router.. pick up a good cisco switch, or HP switch
15:01.54tzangernortex: probably just a common house plan for that time
15:02.00a1fathat supports .q and qos
15:02.04littleballalfa, the problem is that if a few people retrieve emails, the voice quality become bad
15:02.17a1fawhat firewall you got?
15:02.24a1famaybe you can qos through your firewall
15:02.27a1faTOS/QOS
15:02.31littleballno idea, it is in customers side
15:02.34dlynes_officelittleball: with huge attachments?
15:02.35*** join/#asterisk marv[work] (n=timr@64.89.118.139)
15:02.38littleballyes
15:02.44a1fablock email access
15:02.46a1fa:)
15:02.47littleballnot huge, 2-to 5 m
15:02.57a1falittleball : let me guess, the customer has DSL/CABLE?
15:02.59dlynes_officelittleball: look into patching in the sip jitterbuffer from trunk
15:03.14a1fai'm out
15:03.15a1fameeting
15:03.19a1fabbl
15:03.28littleballDSL
15:03.39a1falittleball : DSL and what kind of phone?
15:03.49a1falittleball : sipura firmware supports QOS in box
15:03.53littleballno branding :-) get from Taiwan
15:04.01a1faok, that could be your problem
15:04.13a1famy linksys router supports QOS
15:04.28a1fatoo bad halflife2 is QOS-ed on my home network
15:04.51a1fano need to QOS voip sinc i got 6mbits :)
15:04.51N3WWNlittleball:  have you looked into MikroTik?
15:05.00littleballdoes i need to configure QOS?
15:05.07a1fayou may have too
15:05.15littleballso that voice has higher priority than data
15:05.17*** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net)
15:05.34MACscralfa, you have 6mbits upload?
15:06.09littleballN3WWN, what is  MikroTik? :-( google now
15:06.54N3WWNlittleball:  Linux based router options.  You can get embedded linux systems (RouterBOARD 500, etc) made by MikroTik or install the software on a PC
15:06.58littleballi see. no, i cannot change router for customers until they agree
15:07.11littleballmaybe try it out
15:07.12*** join/#asterisk assert_true (n=Sunil@59.176.57.79)
15:07.34N3WWNlittleball: SOHO license is free and they work like a charm!
15:07.59littleballN3WWN, does it support wifi?
15:08.20N3WWNyup... AP, client, WDS, bridge, etc
15:08.38littleballdid u try it before?
15:08.40littleballgood?
15:09.07littleballhow much is the total cost for such a router?
15:09.27littleballi like the wifi option. it is interesting
15:09.35N3WWNlittleball:  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)
15:10.08fourcheezeHmmhesays: got your mixer working yet?
15:10.11*** join/#asterisk glLoadIdentity (n=tyn@81.214.255.57)
15:10.18Hmmhesaysdidn't work on it last night
15:10.26Hmmhesayssex0red the girlfriend and play some disc golf
15:10.30fourcheezeyou've made me want to get one now
15:10.32littleballN3WWN, did your company pay for wifi abilities?
15:10.33N3WWNlittleball:  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 $250
15:10.45fourcheeze(mixer not girlfriend)
15:10.47littleballcheap
15:11.12Hmmhesayslol
15:11.16N3WWNIf you want to take more about them, can you msg me offline?  We're off-topic here ;)
15:11.34littleballwith QoS? N3WWN. i need to improve voice quality urgently
15:12.25N3WWNlittleball: yes, you can prioritize many ways
15:12.33NuggetI've found QoS to be invaluable on my consumer dsl (768 up, 6m down)
15:12.40NuggetI wouldn't try to use voip without it
15:13.04Persilonis there a s-UNAVAILABLE extension ?
15:13.45littleballN3WWN, thanks. i will check and get one
15:14.01N3WWNNo problem, littleball...
15:14.06littleballNugget, must i go to customer size to configure the QoS?
15:14.17littleballi need to check with them about their router
15:14.19NuggetI don't understand your question.
15:14.36littleballBecause i have no idea what router the customer using now.
15:14.51N3WWNAnyone have experience with VPB channels?
15:15.13littleballmaybe their router has QOS. I am not familiar with such thing. So just ask whether QOS need to be configured
15:15.27NuggetHow would I know?
15:22.16*** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it)
15:24.51*** join/#asterisk florin2703 (n=aaa@florin-jurma.tm.ew.ro)
15:28.44*** join/#asterisk Vorondil (n=cwaldeck@mail.yhamerica.com)
15:29.46PersilonI need some help with s-NOANSWER and s-BUSY, I can't get them to work
15:30.13*** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com)
15:30.21*** join/#asterisk bjohnson (n=bjohnson@i216-58-92-225.cybersurf.com)
15:31.08znoGexten => s-NOANSWER,1,....   doesnt work?
15:31.15znoGPersilon: be more specific
15:31.35*** join/#asterisk salviadud (n=ralfalfa@201.145.29.99)
15:31.47PersilonznoG: in my dialplan I have exten => s-NOANSWER,1,Voicemail(${MACRO_EXTEN}@default,b)
15:32.24*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
15:32.36PersilonznoG: but it doesn't seem to work... if I don't pickup, it hangs up
15:32.55QwellPersilon: Do you have a Goto(s-${DIALSTATUS}) anywhere?
15:33.38PersilonQwell: nope, it should go at the top of the dialplan ?
15:33.57QwellPersilon: read the example in macro-stdexten
15:35.09PersilonQwell: thank you
15:35.16znoGPersilon: you probably need to do the Goto
15:35.20znoGoh, Qwell jumped in :)
15:35.29PersilonznoG: yes, thank you :)
15:35.35*** join/#asterisk tamp4x (n=tampon@64.201.13.51)
15:35.47PersilonI have another question regarding playtones() I can't get it to work either, when I call from a pstn phone
15:37.46*** join/#asterisk Qwell (n=north@unaffiliated/qwell)
15:39.55*** join/#asterisk brijn (n=brijnier@204.244.176.116.net-conex.com)
15:41.46*** join/#asterisk BekokBau (n=alamek@bb219-74-87-237.singnet.com.sg)
15:43.27PersilonQwell: I put the s-${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)
15:45.04lilalinuxdoes the register command support md5 passwords, too?
15:45.05PersilonQwell: nevermid, solved :)
15:45.22smackushas anyone had an issue with mixmonitor killing the asterisk server?
15:45.33BekokBaucan someone help me what is the problem with my configuration? here is my SIP debug http://pastebin.ca/74022
15:45.39dlynes_officesmackus: yes
15:45.46dlynes_officesmackus: 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-FenderPersilon : pastebin your macro
15:47.58[TK]D-Fender~pb
15:48.00jbothmm... 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)
15:48.28Persilon[TK]D-Fender: I forgot to uncomment the dial statement, it's solved
15:48.56Persilon[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-FenderPersilon : "Ringing"
15:51.01Persilon[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)
15:54.38znoGi 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
15:55.08jbalcombI need a project name for my IP phone provisioning system. Ideas?
15:55.22anna--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)
15:55.44masonfdoes anyone know the asterisk cvs servers address?
15:56.08Qwellmasonf: the asterisk cvs servers are dead, and removed from dns
15:56.11symlinkwe don't use CVS anymore
15:56.13Qwellsvn.digium.com
15:56.59znoGcould be an interesting idea... get a solution together that can relay T.38 and offer it in this country
15:57.33coppiceT.38 seems to be flavour of the month
15:57.39*** join/#asterisk sangee (n=rkuru@206.191.114.66)
15:58.06znoGyea, don't know why people insist on using faxes
15:58.08smackuswhere do i put the m flag for the monitor command?
15:58.23smackusnevermind... found it
15:58.30coppiceznoG: low IQ
15:59.38Persilonis there any way of knowing how many parameters were passed when calling a macro ??
15:59.57sangeeI 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?
16:00.11giesenyay
16:00.19sparkleytoneare the asterisk people the same people who program the GXP-2000 firmware?
16:00.21symlinksangee: sip show peer <name> load
16:00.23jbalcomb"ZIPP: Zee IP Phone Provisioner"
16:00.28jbalcombsparkleytone no
16:00.30sparkleytonei just noted that it invites you to this channel on its page
16:00.34sparkleytoneon voip-info
16:00.40giesenit 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
16:00.56jbalcombsparkleytone voip-info is not grandstreams page
16:01.07sparkleytoneyes i know jbalcomb ...i meant the wiki page
16:01.11sangeeok
16:01.18sparkleytonewhich has WAY more info than grandstream's site ;)
16:01.38sparkleytonei want to like this phone...but the firmware makes it impossible
16:01.59sangeewhen i dial to that extension it does not dialing? may be i need to update the database?
16:02.00jbalcombsparkleytone the phone makes it improssible
16:02.01nortexIs 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)
16:02.51sparkleytonejbalcomb: touch? i guess...the hardware must be crap...
16:02.53sangeeit registered okay, i can make outbound call to someone, but when someone call to that extension it does not dialing
16:03.18jbalcombok, 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)
16:03.38jbalcombsparkleytone: the Polycom SoundPoint IP 501 is 95% better
16:03.44sparkleytonei object...its way too good of a name for anything based on OSS ;)
16:03.55sparkleytonejbalcomb: how much is it, ballpark?
16:04.01jbalcombsparkleytone 200
16:04.06jbalcombwith PoE
16:04.13sparkleytoneyeah...cisco phones cost that much
16:04.29sparkleytonemeaning my boss would just make us buy the cisco phone
16:04.36sparkleytonei'll look it up tho
16:04.39jbalcombsparkleytone the cisco phones cost more
16:05.13jbalcombsparkleytone of course you might check out /linksys/sipura
16:06.06sparkleytoneyeah
16:06.15jbalcombsparkleytone plus it depends on how many lines you need, whether or not speakerphone is important, and what you voice quality requirements are.
16:06.17sparkleytonei saw they are finally getting some cheaper products in
16:06.42jbalcomb~seen avibani
16:06.45jboti haven't seen 'avibani', jbalcomb
16:06.54sparkleytonei 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-FenderIP430 = $170 incl PoE and suits most peoples needs
16:07.05nortexsparkleytone, Cisco phone plus the license cost much more then Polycom.
16:07.07jbalcomb[TK]D-Fender polycom?
16:07.11[TK]D-Fender~[av]bani
16:07.14sparkleytonehaving to work with the existing black-magic makes it ridiculous
16:07.15[TK]D-Fenderjbalcomb : Correct.
16:07.32[TK]D-Fenderjbalcomb : Go check it out... amazing little deal
16:07.33nortexsparkleytone, Plus 30 bucks for a power adapter.
16:07.39jbalcomb~seen [av]bani
16:07.41jbot[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?'.
16:07.47sparkleytonewhen i saw 'we' i mean...everyone...as in get rid of PSTN all together and redesign the entire communications system.
16:07.53jbalcombwow, that guy disappeared
16:08.32jbalcombhe was working on auto configuring gxp-2000s
16:09.02nortexjbalcomb, When will ZIPP be aVALIABLE ???
16:09.30Kattyhey 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)
16:10.12jbalcombnortex: if i have to do it all by myself it'll probably take 3 months
16:11.09stephane_re
16:11.31*** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw)
16:12.04jbalcombfeel free to join #ZIPP for more discussion and colaboration on ZIPP
16:12.14nortexjbalcomb, :( I figured since you were picking a name it was almost done.
16:12.56jbalcombnortex: HA! Picking the name comes first so as to encourage the procrastination!
16:13.15jbalcomb"can't start this project until I come up with a great name!"
16:13.40[TK]D-FenderKatty: Mew.
16:14.36*** join/#asterisk saftsack (n=saftsack@p54A7DF23.dip.t-dialin.net)
16:15.40saftsackhi
16:15.46saftsackwhere to find hylafax experts?
16:16.18dlynes_officesaftsack: maybe try #hylafax?
16:17.43saftsackthere isnt such channel ^^
16:21.02Mw3hi. 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
16:24.11*** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal)
16:27.20jbalcombthe IP430 is only $10 cheaper than the IP501
16:28.01jbalcombDoes this "Integrated IEEE 802.3af Power over Ethernet support" mean I dont need the $20 PoE cable?
16:28.38Katty[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-FenderKatty : Somewhat.  Come to the conclusion that I have to change soon and "just do it".
16:30.43[TK]D-FenderKatty : on vacation now, and just vegging away
16:30.52*** part/#asterisk userdefined (i=jr000430@shell1.phx.gblx.net)
16:31.13[TK]D-Fenderjbalcomb : yes, only $10 cheaper, but includes PoE, lighted indicators.
16:31.13*** join/#asterisk StevenL (n=steve@216.62.85.65)
16:31.22_ThorHello everyone
16:31.43jbalcomb[TK]D-Fender That does seem like a good deal then.
16:31.57masonfhow come there is no cdr_mysql in svn?
16:32.21jbalcombmasonf: thats covered on the wiki i believe
16:32.30jbalcombmasonf: licensing.
16:33.47ManxPowermasonf, Digium has a binary only Asterisk product.  Digium does not want two codebases and so cannot include MySQL stuff in Asterisk
16:35.37masonfmakes sense.. thanks
16:36.21Nuggetpostgresql!  :)
16:36.31jbalcombIs the AGI for use inside asterisk and the API for use outside asterisk?
16:36.37Nuggetbetter solution and free-er.
16:37.13jbalcombNah, I like L.A.M.P. rather than L.A.P.P.
16:38.06sparkleytoneSAOP
16:38.08sparkleytoneftw
16:38.14NuggetI was just pointing out that using postgresql would remedy digium's licensing challenges with mysql and provide for a more robust solution.
16:39.28masonfvoip-info says to get cdr_mysql from cvs.. sorry I have to ask another RTFM but where can I can cdr_mysql
16:39.45wunderkinaddons?
16:39.56ManxPowermasonf, all mysql stuff is in asterisk-addons
16:40.10*** join/#asterisk copantl (n=galel@190.4.22.82)
16:40.14jbalcombIs the AGI for use inside asterisk and the API for use outside asterisk?
16:40.14copantlhi guys
16:40.34ManxPowerjbalcomb, You mean AMI not API
16:40.42KattyNugget: prepare yourself. you're about to be hugged.
16:40.47copantlcan asterisk carrier a dedicated data channel?
16:41.02jbalcombManxPower AMI is Asterisk Management Interface?
16:41.03ManxPowercopantl, In theory.  Define "dedicated data channel"
16:41.09ManxPowerjbalcomb, correct
16:41.13Nuggethooray
16:41.36copantlhi ManxPower, can i have lised lines for data?
16:41.37Kattyand now i'm gone (=
16:41.41jbalcombManxPower ok, gotcha. So then what ths diff. between the AGI and the AMI?
16:41.53jbalcombManxPower As far as what to use them for really
16:41.56ManxPowerlised?
16:42.26jbalcombcopantl: there is some function/system out there for /sharing/
16:42.36ManxPowerjbalcomb, 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
16:43.02ManxPowercopantl, Zaptel allows you to DACS channel(s) to other channel(s)
16:43.09jbalcombManxPower: Very good, thank you.
16:43.11copantlManxPower: 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)
16:43.50ManxPowercopantl, yes, assuming your "dedicated some channels" just means "a digital version of cross connecting 1 channel to another"
16:43.56copantlwich protocols can use?, frame relay, HDLC?
16:44.08ManxPowercopantl, DACS is protocol independent
16:44.12copantlok
16:44.19ManxPowerfor example:
16:44.28ManxPowerTelco -> T-1 -> Asterisk -> Channel Bank
16:44.36ManxPowerwell...
16:44.41ManxPowerTelco -> T-1 -> Asterisk -> T-1 -> Channel Bank
16:45.17ManxPowerWe 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
16:45.17ManxPowerl
16:45.36ManxPowerthis assumes your server has at least 2 digium T-1 ports on it
16:46.08copantlcan i do this?: telco ->E1 -> asterisk -> channelbank -> DATA client
16:46.30ManxPowercopantl, yes, that is really no different from the setup I just described.
16:46.49ManxPowerassuming your Asterisk has at least 2 E-1 interfaces
16:46.58ManxPowercopantl, what KIND of data?
16:47.04ManxPowerIf you mean fax then say that.
16:47.17copantlframe relay or ip
16:47.25Nugget110 baud acoustic modem for a tandy model 100.
16:47.44ManxPowercopantl, for DACS it's just bits.  Zaptel does not know those higher level protoco.s
16:48.52copantlManxPower: is a clear channel?
16:49.04ManxPowercopantl, yes.
16:57.19florzwhen 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?!
16:57.28nothinmanexit
16:57.32nothinmansorry ;]
16:58.05florzerm, more clearly:
16:58.18florzwhen 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)
16:59.13rpmwhats the ibm clustering/redundancy project for asterisk called?
16:59.19japerryg'morning Cunningpike =)
16:59.34japerryany 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)
17:00.04gennarohi
17:00.38CunningPikeHey japerry - not really :(
17:00.39gennarosomeone can help me with _9xxxxxx pattern ?
17:00.47Juggie?
17:00.50*** join/#asterisk bernardovieira (n=bvieira@c911935d.static.bhz.virtua.com.br)
17:00.50Juggiewhat do you need help with
17:00.55gennaroin italy there are numer like this 0573416604 or
17:01.17japerryJuggie: something wrong with a polycom 601 dropping calls outgoing randomly
17:01.18gennaronumber more long
17:01.22gennaroin the same city
17:01.33gennarohow can i do?
17:01.44CunningPikejaperry: Did you get anything helpful from your debugs?
17:01.50Juggiegive me an example of a short and long number
17:02.00japerryCunningPike: nope
17:02.02gennaro0573416604
17:02.09gennaro05734166041
17:02.13japerryCunningpike: but then again, she hasn't reported it dropping yet
17:02.18Juggieoh hmmm
17:02.23Juggiethose are both valid?
17:02.39gennaro0573416604
17:02.44CunningPikejaperry: OK - you have 'pri debug span x' set, yes?
17:02.51jbalcombrpm: perhaps its 'System i'
17:02.54CunningPikejaperry: And a CLI session runnng?
17:02.57gennaro057325371
17:03.10gennaroand others...
17:03.15mutugh, 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
17:03.18gennaroin other city...
17:03.19japerryCunningpike: yup, I just called the front desk, told her to call me first before calling out
17:03.30CunningPikejaperry: Then get her to call you as soon as she gets a dropped call and scroll back in the PRI debug
17:03.37CunningPikejaperry: Perfect!
17:03.51CunningPikejaperry: It's tedious, but really the only way........
17:04.10gennarojuggie do u know how can i do?
17:04.10*** join/#asterisk speedwagon (n=Ariel@70.46.87.158)
17:04.28japerryheh
17:04.32Juggiegennaro, _0XXXXXXXXXX. perhaps
17:04.38japerrypri debug span 1 gives an error
17:04.41Juggieyou'll have a delay while asterisk waits for more digits
17:04.46japerryCunningPike: not a pri
17:04.46Juggiebut it should be ok.
17:05.11*** part/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net)
17:05.30CunningPikejaperry: 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)
17:05.57japerryyup, but its not a pri
17:06.03CrashHDI 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?
17:06.09gennarooh so if i do so
17:06.35japerryCunningpike: well its not 'technically' a PRI, its 4 regular lines over a T1 carrier
17:06.46nortexjaperry, Is it a channelized T-1
17:06.54gennarowith an analog phone i cant use SEND key and pbx wait for nothing and call dont start
17:06.59CunningPikejaperry: Using E&M, right?
17:07.10japerryYes. but we are only using 4 channels
17:07.19japerryso nortex, yes
17:08.04*** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
17:08.06gennarocan i use # as send key?
17:08.18CunningPikejaperry: I think your config might need tweaking then - your zaptel.conf looks like it's set up for a PRI.....
17:08.34Juggiegennaro, have you checked to see if * can read the send key?
17:08.52japerryCunningPike: if it was a PRI though wouldn't it have a B channel, etc?
17:09.04gennarohow
17:09.17japerryCunningPike: I believe e&m=9-12 in the /etc/zaptel.conf file is what makes it not a PRI
17:09.30Juggiei assume you are using a analog phone connected to an ata which is registered to asterisk?
17:09.49CunningPikejaperry: Can you pastebin your zaptel and zapata.conf again?
17:09.53gennarono directly with DIGIUM FXS
17:09.55japerryyah, one sec
17:09.59CunningPikejaperry: Or resend the link
17:10.05Juggieoh ok, either way your phone has dialtone correct?
17:10.12gennaroyes
17:10.13japerryZapata http://pastebin.ca/73561
17:10.23Juggieok, well just remove whatever extension logic you have and do only this
17:10.35Juggie_X.,1,Noop(${EXTEN})
17:10.43Juggiethen pick up the phone dial a few numbers and press send
17:10.46japerryzaptel: http://pastebin.ca/74075
17:10.47Juggiesee what asterisk sees
17:11.00gennaroNoop ?!?
17:11.09Juggienoop is just a way to echo debug output
17:11.12Juggieyou'll see :)
17:11.22Juggiewe are just testing, this isnt going to connect a call.
17:11.22gennaroi use DIAL(ZAP/4/EXTEN
17:11.26Juggiei know that
17:11.35Juggiebut i want to see what is being sent from the phone
17:11.52Juggieso do '_X.,1,Noop(${EXTEN})' pick up the phone, dial something, press the send button
17:11.53*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
17:11.56Juggieand then lets see what happens
17:11.58CunningPikejaperry: OK - taking a look
17:12.05Juggieso, exten=>_X.,1,Noop(${EXTEN})
17:12.09Juggiethats all you should have.
17:12.11gennaroi put down internet...
17:12.16gennarotanx
17:12.29Juggiehah.
17:12.31Juggieawesome
17:12.45Juggiei was trying to see what dtmf the send key sent.
17:12.52Juggiewhich i suspect is a,b,c,d
17:12.55Juggieone or the other
17:13.17*** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx)
17:15.14rene-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-Fenderrene- : Few clients will be able to pick up based on the eadsets
17:15.50Juggieummm.... this is support for asterisk :)
17:16.31creativxsoftphones is the shit
17:16.48rene-nobody i know wants to buy ip phones for call centers
17:17.35creativxi have yet to see a non-shitty softphone
17:17.40Juggiesoft phones with windows roaming profiles would be better.
17:17.52Juggieuser logs in, they get all their settings on whatever machine they log into and register.
17:18.36rene-[TK]D-Fender: what do you mean?
17:19.18creativxJuggie: but what softphone
17:19.32Juggiexten is fine.
17:21.25creativxi found it to be a bitch
17:21.37creativxtoo much eyecandy
17:22.38nortexcreativx, But did it work?
17:22.57creativxi dont remember ;)
17:23.11creativxi bought ip10s phones instead and forgot about it
17:26.08Heimidalis there a way to add a prefix to the caller id (using SetCIDNum) for all inbound calls from numbers outside our office?
17:26.13Qwell[]sure
17:26.23*** join/#asterisk jgoo (n=isec@ppp129-190.adsl.forthnet.gr)
17:27.13CunningPikejaperry: Have you played around with rxwink and similar settings?
17:27.21creativxtheres no way to answer a channel via the asterisk manager interface, is there?
17:27.23MACscrQwell: is it possible to rewrite the Name in the caller id, but keep the number the same?
17:27.27Qwell[]MACscr: sure
17:27.42CunningPikeMACscr: Set(CallerID(name))
17:28.45jgooCunningPike, as I newb, i am just curious, where would you put that Set(callerID(name)) ??
17:28.53Qwell[]jgoo: dialplan
17:29.03jgooreally? oh ok, thought as much
17:29.11jgoothat is extensions.conf? or another file?
17:29.14Qwell[]yes
17:29.25MACscrThe 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
17:29.29jgoook cool thanks
17:29.31MACscrer caller
17:29.58jgooMACscr, PM me more info what you are doing, I find that interesting
17:30.09jgooMAC, will this be DB driven_
17:30.10jgoo?
17:30.13*** join/#asterisk caloi (n=caloi@nat-66-218-1-201.usadatanet.com)
17:30.32MACscrright now im only going two companies, so it wont be that fancy
17:30.35smackusquick 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?
17:30.39MACscrbut i was planning on having it interfacing with a crm software
17:30.47MACscrsuch as SugarCRM
17:31.03smackusI am troubleshooting permissions issues on the box again.
17:31.03MACscrim pretty sure Trixbox already does this
17:31.13caloismackus - sip show (tab tab) will give you all the sip show options
17:32.10salviadudsmackus, sip show peers
17:32.23CunningPikejgoo: In your dialplan: exten => 1234,n,Set(CallerID(name)=foo)
17:32.38salviadudsmackus, sip show channels is for active sip channels
17:32.51japerryCunningPike: a litte, could that be causing the drop though?
17:33.26CunningPikejaperry: 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)
17:33.52MACscrCunning: could that be set per incoming DID instead of per ext?
17:34.12CunningPikeMACscr: Could what be set?
17:34.32MACscrthe callerID(name)
17:34.55HeimidalQwell[]: how can I achieve the alteration of just caller id from numbers outside our office?
17:34.55japerryCunningpike: hmm okay, I'll fiddle with it and see what outcoes
17:35.27MACscrHeimdal, thats what their talking about
17:35.28japerryCunningPike: so the other 'issue' is callerid which you probably see is statically set
17:35.29CunningPikejaperry: OK - it'd be really nice to get some debug info.......
17:35.30Qwell[]Heimidal: see above
17:35.52*** join/#asterisk crich1999 (n=crich@port-212-202-198-145.dynamic.qsc.de)
17:35.57CunningPikejaperry: Outgoing, right?
17:36.34jgooCunningPike, just roughly how tricky is it to get the dialplan to interface with sugarcrm or another database_ I guess the dialplan can call scripts?
17:36.47CunningPikeMACscr: I have a macro for incoming calls that sets the CID - what exactly are you trying to accomplish?
17:36.47HeimidalI 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)
17:36.48jgooand evaluate expression inside it etc...
17:36.58japerryCunningpike: 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)
17:37.18japerryjaperry: so it seems that outgoing callerid has to be handled by verizon
17:37.25CunningPikejaperry: Yes - are you getting incoming CID OK?
17:37.31*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
17:37.39PersilonHow 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
17:37.40CunningPikejaperry: The ability to set outgoing is determined by your telco
17:37.45japerryCunningpike: negative :-(
17:37.56MACscrCunningPike: 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
17:38.10MACscrbut i would like them to still be able to see the phone number that is calling
17:38.14japerryCunningpike: asreceieved doesn't seem to work
17:38.21CunningPikejaperry: Try adding a Wait(2) before you answer - CID takes 2 rings to appear
17:38.22MACscrespecially with calls that are fwded to cell phones
17:38.49Heimidal:(
17:39.18CunningPikeMACscr: So, in your incoming call context, use Set(CallerID(name)) based on the number called
17:39.56CunningPikeMACscr: exten => whateveryourincomingnumberis,1,Set(CallerID(name)=foo)
17:40.34CunningPikejaperry: Are you getting any kind of incoming CID at all? What are you seeing for incoming CID?
17:41.20CunningPikebrb
17:41.59japerrykk
17:43.30*** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br)
17:44.17caloiquick 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)
17:51.29Hmmhesaysso whoes going to swing me some cash for cluecon
17:52.08mogormanhmm akward silence.... i guess thats a no
17:52.30symlinkif you'll accept 1000% daily interest starting on the nanosecond it's loaned to you
17:52.31symlinksure
17:53.29KattyHmmhesays: you can nap in my room if ya want ;)
17:53.45KattyHmmhesays: if you bring any girls back, tho, you have to share.
17:53.54symlink!!!
17:54.00mutewww katty is a cannibal?
17:54.10Hmmhesayslol
17:54.22HmmhesaysFried or Baked?
17:54.31KattyHmmhesays: annnd, you have to teach me how to jump in billiards.
17:54.38mutand she does drugs?!
17:54.40Hmmhesayssymlink: do you know what a vagina is?
17:54.42mutNO WONDER!
17:54.47Hmmhesaysbecause with a comment like that surely you've never seen one
17:54.53KattyHmmhesays: i prefer baked...less fat haha
17:55.05KattyHmmhesays: i appologize, that was low.
17:55.07symlinkHmmhesays: I was surprised with Katty's response
17:55.30Kattysymlink: yeah well you don't know me that well either ;)
17:55.48symlinkKatty != Katty!
17:56.00japerryCunningpike: 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)
17:57.56rene-Katty's cute, cant say the same for the rubyonrails chick tho
17:58.04catch23_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
17:58.22MACscrKatty?
17:58.29catch23_I just figured there would be more server administrators in #asterisk than #ubuntu...
17:59.09MACscrlol, im blind, sry
17:59.10MACscr=P
17:59.29KattyMACscr: yes?
17:59.36KattyiDunno: you didn't hug it enough )=
17:59.43KattyHmmhesays: you thinking about coming to cluecon?
17:59.53HmmhesaysPossiblI
17:59.58MACscrkatty: someone was saying you were hot in comparison to the RoR chic
18:00.00KattyHmmhesays: i could use an escort downtown.
18:00.11MACscrdidnt see your name and was trying to figure out who they were talking about
18:00.14KattyHmmhesays: can't have any of those weird guys mumbling to themselves while i walk by ;)
18:00.38Hmmhesayslol
18:00.43KattyMACscr: i'm kinda surprised rene- has seen a picture of me, really.
18:00.43Hmmhesaysexcept me?
18:00.49KattyMACscr: i dunno who rene- is
18:00.53KattyHmmhesays: yes dear, i know you're insane.
18:01.02KattyHmmhesays: i've come to expect mumbling from you!
18:01.04HmmhesaysLOL
18:01.18MACscrlol
18:01.57MACscrlol, i didnt even know about cluecon
18:02.09MACscrits only about 2 and half hours from me
18:02.42Kattyit's about 8 from me.
18:02.46Hmmhesayshop on your vespa and come on down
18:02.49Kattyi'm sure someone will meet me at the amtrak station tho
18:02.52MACscrbut im also not in the telecom arena anymore really
18:03.05Kattysurely bkw will, if no one else.
18:03.11MACscrKatty, are you north?
18:03.18rene-Katty: you were @ astricon anaheim right?
18:03.20KattyMACscr: south west.
18:03.23japerryCunningPike: BINGO
18:03.24Kattyrene-: no, no i wasn't.
18:03.28japerryCunngingpike: Incoming call: Got SIP response 500 "Internal Server Error" back from 10.0.6.100
18:03.28japerry<PROTECTED>
18:03.36MACscrah, ok. Im in Peoria, IL
18:03.37KattyMACscr: i'm right around st. louis
18:03.53rene-oh, i have lived a live of deceit
18:04.01MACscr8 hours to chicago from st. louis?
18:04.08KattyMACscr: i'm 2 hours south of st. louis ;)
18:04.19MACscrlol, ok
18:04.22rene-well then you were right i havent seen you ever :)
18:04.40Kattyrene-: (=
18:04.44japerryCunningpike: I think when she makes an outgoing call and someone calls her, it drops the call she made
18:04.48KattyMACscr: you're roughly 5 hours from me.
18:05.34Kattyrene-: rubyonrails girl? i'd be interested in seeing that. post gifs.
18:05.53Hmmhesaysself esteem is one of the greatest offspring songs EVER
18:06.05KattyHmmhesays: i don't know about that.
18:06.10KattyHmmhesays: you better send it to me so i can compare
18:07.38Hmmhesayssent
18:08.05Kattyexcellent.
18:08.13Kattyi'm currently playing the hell out of lullaby by the cure
18:08.17Kattyever 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)
18:08.39phigworkhello hello.
18:08.42anthmumm yeah they made it when they were elderly
18:08.58phigworkis there a string name for an incoming callerid number?
18:09.10Kattyanthm: the cure?
18:09.18phigworki'm trying to set the text field of an incoming line to something, but I still want to pass the telephone number
18:09.22anthmyah =D
18:09.33Kattynot surprising...but i just heard of them about 2 or 3 months ago
18:09.52anthmoh
18:09.58anthmthere's more
18:10.21Kattyyeah, i have 6 albums now
18:10.26HmmhesaysYeah i've heard it
18:10.30Kattylovecats is a favorite too
18:10.37KattyHmmhesays, did you send it to my gmail?
18:10.53Hmmhesaysthe lady!@#!@#!@gmail yeah
18:10.54anthmin between days, head on the door
18:10.59phigworkand whats the difference between SetCIDName and SetCallerId?
18:11.19anthmplay for today
18:11.26anthmhanging garden
18:11.37sparkleytonewhat is/are the proprietary codec(s) that i'm not supposed to use again?
18:11.50KattyHmmhesays: 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)
18:16.57CunningPikejaperry: 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)
18:18.08CunningPikejaperry: 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
18:18.31CunningPikecaloi: What does your context for incoming calls look like now?
18:18.41bkervaskiHi 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.
18:19.04creativxbkervaski: look at the sample musiconhold.conf file
18:19.13japerryCunningpike: well I thought we were getting somewhere, but alss nope. the 500 server error is independant of hangups
18:19.18bkervaskiThat's what I'm using.  There's nothing about mpg123 in there??
18:19.18CunningPikebkervaski: Have you configured native MOH properly
18:19.29japerryCunningpike: also, incoming calls don't necissarly hang up her phone
18:19.33bkervaski@CunningPike: Probably not.  Do you have to specify it?
18:19.36CunningPikejaperry: That's what I suspected. Drats
18:19.56bkervaskiAhh.. got it.. thanks all, ;)
18:19.59CunningPikebkervaski: Oh, yes - look up 'native MOH' on the wiki - there is good into there
18:20.11CunningPike~nativemoh
18:20.20japerryCunningpike 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
18:20.48japerryCunningpike: and talking about rxgain, I've seen a suggestion to lower it to get callerid to work.. all other ways seem to not work
18:21.27CunningPike~moh
18:21.29jbotmoh 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
18:21.29bkervaskiI like mpg123, but from time to time it spikes out the CPU and won't let up until you kill it...
18:21.59CunningPikebkervaski: With the advent of native MOH, there is no longer any good reason to use mpg123
18:22.33CunningPikejaperry: Worth a shot - and there was nothing else in the CLI near the last hangup?
18:23.03CunningPikejaperry: Can you do a 'sip debug peer' on her phone and see if that shows anything.......
18:23.57bkervaskiSo 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)
18:25.01bkervaskiDoes 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?
18:25.16Qwell[]bkervaski: no, it streams it non-stop
18:25.25japerryCunningPike: I'll keep it running. she isn't calling out atm
18:25.30andrebarbosaanyone know how i call a dialplan function from an agi script?
18:25.35Qwell[]even when there are no calls on hold
18:25.45Zodiacalanyone 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?
18:25.56Qwell[]It's your fault, you know?
18:25.58andrebarbosai can call aplications using exec() ;)
18:26.00symlinkyup
18:26.06Qwell[]This is seriously just out of hand
18:26.26Qwell[]I'm pretty sure Edison has been screwing up on my bill...
18:26.43CunningPikejaperry: OK - hopefully we'll see something
18:26.44Qwell[]It's incredibly difficult to get a $300+ electric bill in a friggen apt
18:27.09Qwell[]That's more than double what it was 4-5 months ago
18:27.20Qwell[]nearly triple, in fact
18:27.27salviadudif 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?
18:27.29symlinkQwell[]: I snuck in
18:28.09salviadudif i did it with perl, would that be better?
18:28.26rob0file ... directory ... symlink ... I am seeing a pattern here
18:28.31xhelioxWhat'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. :)
18:28.33Qwell[]rob0: No you aren't
18:28.41symlinkrob0: I am not the person you are looking for
18:28.45Zodiacalqwell whats a normal electric bill in a house with central air in 90+ weather? just got an A/C :P
18:28.51rob0yes you are
18:29.07Qwell[]Zodiacal: I don't know...but my usage hasn't changed from before, when it was only like $150 or less
18:29.13CunningPikeGot 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
18:29.35Qwell[]Zodiacal: That's in CA though, where electricity is like $9/kwh
18:29.51salviadudforwarded what?
18:30.42creativxwe only got 80% of that story CunningPike :o
18:30.52CunningPikexheliox: Try operator=yes in voicemail.conf
18:31.12CunningPikecreativx: Oh - sorry. How far did you get? :)
18:31.26Zodiacalqwell yeah socal
18:31.41creativxCunningPike: up to "forwarded th"
18:31.42symlink"had forward t"
18:31.42creativx;)
18:31.50xhelioxCunningPike: Hmm, ok..
18:31.53salviadudyeah, man, you wrote it
18:31.56*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
18:32.09CunningPikehad forwarded their phone to a phone in Engineering..........
18:32.19symlinkpeople are smrt
18:32.29salviadudyou denied forwarding after that?
18:32.37CunningPikesalviadud: Bingo!  :D
18:32.44anthmKatty: http://www.cluecon.com/mp3/ made from videos of Live 8 in Paris last Summer
18:32.48salviadudhehe, nice one
18:32.52X-Gensalviadud, have u had a look at the management API to setup calls ?
18:33.03salviadudnortex, i have not
18:33.08salviadudi mean
18:33.11CunningPikesalviadud: The Polycoms allow you to deny forwarding on a per registration basis which is great
18:33.14salviadudX-Gen, no i have not
18:33.33X-Gensalviadud, http://www.voip-info.org/wiki/view/Asterisk+manager+experience
18:33.49salviadudCunningPike, so you tweaked the phone instead of the box?
18:33.55CunningPikesalviadud: Yes
18:34.06salviadudX-Gen, thanx man
18:34.13CunningPikesalviadud: We use the UA forwarding, not the asterisk one
18:34.37CunningPikesalviadud: Just happened to see the SIP 302 as it flashed past in the CLI
18:34.56creativxsalviadud: i have
18:34.56*** join/#asterisk NeonLevel (i=HydraIRC@200.52.142.184)
18:36.30salviadudcreativx, would you say the call manager api is flexible?
18:36.37*** join/#asterisk mogorman (n=mogorman@gateway.digium.com)
18:37.01creativxsalviadud: to some extent. how many clients are you planning to hammer the ami?
18:37.08X-Gensalviadud, it looks like its not suitable for multiple clients talking to it
18:37.18creativxthats why we use astmanproxy
18:37.21*** join/#asterisk pnlarsson (n=niklas@c83-248-2-120.bredband.comhem.se)
18:37.24creativxthen you can beat the shit out of it =)
18:37.25japerryCunningPike: okay got the log
18:37.25X-Genu should write an in-betweener
18:37.36X-Gen~astmanproxy
18:37.40CunningPikejaperry: Great - can you pb it?
18:37.40japerrydoesn't seem to show much to me, but getting it in pastebin now...
18:37.47CunningPikejaperry: Great
18:38.11X-Gen~waves
18:38.13vader--hmmm
18:38.25salviadudhehe, 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
18:38.25creativxsalviadud: i use the ami for originating calls, CID and hangup
18:38.32vader--i installed ntpupdate on my debian box and there is /etc/init.d/ntpupdate
18:38.39vader--but for some reason it never updates automatically
18:38.47vader--and the server's time is always off
18:38.56japerryCunningpike: http://pastebin.ca/74149
18:38.56vader--i run ntpupdate and all is fine
18:39.17X-Gencreativx, how about a CSTA intergace ontop of that ?
18:39.19salviadudso i'm wondering if i should pull out some duct tape and perl it. or just use the manager api
18:39.30creativxX-Gen: yet another acronym.. whats csta J
18:39.52creativxsalviadud: originate, then wait for hangup event, wait -> place new call, repeat?
18:40.02salviadudriiight!
18:40.25salviadudit's for a political campaign. haha
18:40.30creativxhehe
18:40.32creativxvicidial?
18:40.53salviadudi'm gonna dial a lot of freakin numbers
18:40.56creativxtheres already some campaign-dialers out there
18:41.12salviadudyeah, i've heard
18:41.21creativxwhere you can have 50 agents just attending outbound calls
18:41.21X-Genwhat about fax/answere machine detection ?
18:41.22salviadudand since it's not hard to do
18:41.34creativxnothing is hard with asterisk =)
18:41.38creativxyou just gotta know how.. hehe
18:42.05salviadudfax/answering machine... well, thats a big sorry for the guy with the money
18:42.54salviadudand the funny thing is. most automated calls get hung up
18:43.12creativxwell people in general hate talking to random people, dont they
18:43.25*** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin)
18:43.26salviadudi like prank calling, but, that's something else
18:43.30creativxhehe
18:43.40salviadudi like calling people with a british accent
18:43.47creativxbut are you planning to let a live person talk to some random number?
18:43.53creativxor play them some automated jumbomumbo
18:44.14creativxi dont know how these dial campaigns works.. except those who are out to ask questions/statistic analyzis
18:44.18salviadudjust play automated recordings
18:44.27salviadudi can't even get dtmf signals to work
18:44.32*** join/#asterisk jilott (i=jordan@24.79.192.187)
18:44.33salviaduddamn voipdiscount!
18:44.51creativxhehe
18:44.57salviadudit's easy, you get a range of numbers
18:45.01salviadudyou call 'em
18:45.09salviadudand hopefully they will interact with the IVR
18:45.20salviadudi can't get an ivr, cause i'm using SIP and not Zap
18:45.33creativxum
18:45.36creativxno
18:45.40salviadudmy provider is not the kind of provider i can ask for assistance on that
18:45.43CunningPikejaperry: I see a couple of SIP CANCEL messages in there - I'm just going to see if they are normal for this call progression
18:45.45creativxyou have problem with transferring dtmf?
18:45.45salviadudwell, its ugly sip
18:45.47X-Genu cant use IVR on sip, thats new ?
18:45.58salviadudyou can
18:46.03salviadudbut, not with voipdiscount
18:46.10*** join/#asterisk NotJohnDavid (i=dave@c-68-47-199-178.hsd1.tn.comcast.net)
18:46.14bkervaskiWill "show channels" from the * cli show *ALL* channels?
18:46.17salviadudi don't wanna confuse anybody here... dtmf signalling should work on any codec
18:46.17NotJohnDavidanyone use an sipura hardware?
18:46.28salviadudi use sipura
18:46.29creativxthen well if you need dtmf/ivr i would change service providers ;)
18:46.31salviadudsip 3000
18:46.52salviadudyeah. well, that does not go along with my "suppa cheap mexican prices"
18:47.02NotJohnDavidsalviadud: that's what I have.  trying to make it ring thru from PSTN->FXS
18:47.11NotJohnDavid(without using asterisk)
18:47.13jilottgood 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?
18:47.15creativxlike selling you an internet connection but not letting you use the smtp protocol
18:47.45salviadudnotjason, you gotta abilitate the pstn ring through line 1
18:47.50salviadudoops
18:47.51NotJohnDavidsalviadud: thought that if I put "*" in as ring1 caller under ring-thru distinctie caller it'd work but apparently not
18:48.27NotJohnDavidPSTN Ring Thru Line 1: is set to YES
18:48.38*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
18:48.46salviadudNotJohnDavid, how many rings before it rings it through?
18:48.49japerryCunningPike: thanks, yah I'm not sure if they are normal or not.. and I dunno how they'd be called
18:48.56salviadudthere's an option for that too
18:49.01NotJohnDavidit doesn't ring through
18:49.07*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
18:49.09CunningPikejaperry: 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)
18:49.27salviadudare both your channels registered?
18:49.38justinu|laptopcancel is what happens when you hang up before a call is answered
18:49.46CunningPikejaperry: Call termination is indicated with a BYE message, not a CANCEL
18:49.54salviadudi've had some trouble with the spa 3000.  if you don't have a channel registered, it kinda screws up
18:50.05CunningPikejustinu|laptop: But japerry is getting them halfway through a call......
18:50.08salviadudand if you do happen to have the sipura connected to asterisk
18:50.19NotJohnDavidoh 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
18:50.28justinu|laptopCunningPike: he's getting CANCELs after 200OK?
18:50.46salviadudNotJohnDavid, you can get the manual at their site
18:50.50rpmhow do i convert gsm sounds to g729 compatible?
18:50.51CunningPikejustinu|laptop: http://pastebin.ca/74149
18:51.03NotJohnDavidsalviadud: i don't have a proxy to register the pstn to?
18:51.27salviadudNotJohnDavid, if you don't register the pstn to something... it kinda freaks
18:51.32CunningPikejaperry, justinu|laptop: There's a '487 Request Terminated' in there, too
18:51.35NotJohnDavidoh i think i may have found some settings *Tries*
18:51.43salviadudNotJohnDavid, is your line1 connected to asterisk?
18:52.12NotJohnDavidsalviadud: no
18:52.47*** join/#asterisk lars-ut-away (n=lars-ut@70.103.228.158)
18:52.50NotJohnDavidsalviadud: all I want is for PSTN when ringing to pass through to the FXS on the spa3000
18:53.02justinu|laptopCunningPike: i don't see that call ever being answered
18:53.04NotJohnDavidoutbound VoIP and inbound VoIP works just fine
18:53.06justinu|laptopso the CANCEL looks valid
18:53.50japerryCunningpike, justinu|laptop: but the call does go through.... hmm
18:54.07justinu|laptopjaperry: for some reason, nothing is generating an answer... (200 OK)
18:54.40CunningPikejustinu|laptop: He's using E&M - it looks like there is something missing in the supervision
18:54.48salviadudNotJohnDavid, 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
18:55.08japerryCunning, so perhaps the wink settings in zapata aren't totally correct
18:55.17NotJohnDavidsalviadud: can I do that without asterisk?
18:55.29justinu|laptopCunningPike: ok, in that case, the far end is either not setting AB bits to 1 on answer
18:55.33X-Gencreativx, are there any apps that run ontop of astmanproxy ?
18:55.37justinu|laptopor asterisk isn't seeing the AB bits go high.
18:55.57justinu|laptopbut yes, the problem is that for some reason, answer supervision isn't working.
18:56.21salviadudNotJohnDavid, 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
18:56.26creativxX-Gen: no
18:56.39creativxX-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)
18:56.44salviadudcya later guys!
18:56.47CunningPikejaperry, justinu|laptop: Could be worth messing with wink settings.........
18:57.02CunningPikejustinu|laptop: It's em_w, apparently....
18:57.26justinu|laptopi've never seen e&m wink work right on asterisk
18:57.29japerrycunningpike, justinu|laptop: wink is 300, prewink is 20
18:57.34japerrylol
18:57.39justinu|laptopsome people say it does
18:57.40CunningPikejaperry: This would probably explain the CID issues, also
18:57.50*** join/#asterisk Zhadnost (n=tom@host-84-9-159-76.bulldogdsl.com)
18:57.51CunningPikejaperry: Can your telco provide partial PRIs?
18:58.02CunningPikejaperry: We have a 3-channel PRI here
18:58.07japerryCunningpike: probably. I'm wondering if thats what we should do
18:58.13justinu|laptopask your telco to switch to immediate, as a last resort
18:58.15japerryCunningpike, how much might I ask does that run?
18:58.21justinu|laptopbut go PRI if at all possible
18:58.36CunningPikejaperry: It's quite reasonable - let me check
18:59.30*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
18:59.32CunningPikejaperry: Well, hey, at least we got hints working! :D
19:00.08japerryCunningpike: tis true! hehehe
19:00.45japerrythe 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
19:01.58jilottgood 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?
19:02.40creativxjilott: codec conversion?
19:02.43Zhadnostare the incoming and outgoing calls using the same codec?
19:02.56jilottI assume so.  Is there a way to tell?
19:02.57justinu|laptopthe problem is likely on the ITSP side
19:03.08creativxsounds more like something i'd ask the service provider about
19:03.12CunningPikejaperry: CAD$140/mo for 3B+D
19:03.53jilottcool,  I've found that the problem exists for alaw,ulaw, and g729
19:04.00Zhadnostlook in your asterisk console (with verbosity set high) or iax2 show channels when in a call.
19:04.13creativxwhat codec is the calls delivered to you in?
19:04.26Zhadnost? ? ?
19:04.33jilottthanks for that.  I'm new to asterisk.  I thought it was ULAW that it was delivered in.
19:04.57CunningPikejaperry: 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
19:05.29japerryCunningPike: really for that price? damn.
19:05.47CunningPikejaperry: High?
19:06.08japerryCunningPike: oh no, thats very nice
19:06.19japerryCunningpike: we're paying something like $275/month-ish
19:06.43CunningPikejaperry: Ah - you'll be much better off with a partial PRI then - and you can add extra channels really easily
19:06.48ZhadnostI 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.
19:07.01Zhadnostdoes anyone know a good strategy to tweaking the settings for the line?
19:07.11japerryCunningpike: 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)
19:07.28CunningPikejaperry: Yes - we have a 13-channel in production that used to be a 3
19:07.47CunningPikejaperry: 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)
19:09.20rpmis it normal for festival to have a zombie process running along with the parent?
19:10.52rocheHello 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
19:13.12Hmmhesaysmake[2]: Entering directory `/usr/src/hmm0/buildroot/build_mipsel/asterisk-1.0.9/channels'
19:13.13Hmmhesays./gentone busy 480 620
19:13.13Hmmhesays./gentone: ./gentone: cannot execute binary file
19:13.13Hmmhesaysmake[2]: *** [busy.h] Error 126
19:13.16Hmmhesayswhat.. the crap
19:13.23*** join/#asterisk asterisknewbiezz (n=chatzill@rrcs-67-52-187-18.west.biz.rr.com)
19:13.43Hmmhesayswhy are we trying to execute gentone...hmmmmmmmmmm?
19:13.46asterisknewbiezzanyone know how to fax in asterisk?
19:13.52Hmmhesaysalksdfja;sljkf
19:13.53rob0hmm, I say!
19:13.55HmmhesaysThank you that is all
19:13.58Zhadnostwith hylafax?
19:14.41*** join/#asterisk funxion (n=nunya@63.214.236.169)
19:15.00asterisknewbiezzdoes that work with voip?
19:15.20Hmmhesaysanyone know what  ./gentone is supposed to do?
19:15.28Hmmhesayscause it sure as hell is not going to work when i'm cross compiling
19:15.39justinu|laptopit makes sound files
19:15.47funxionis there anyway to run a command in the dial plan on a call that is currently in use
19:15.49creativxgenerate tone
19:15.50creativxhehe
19:15.57*** join/#asterisk jorgeikeda (n=freebot@201.154.10.103)
19:16.02rob0Gentone was the predecessor of Gentwo.
19:16.07Hmmhesayswell duh
19:16.21Hmmhesaysrob0 was that a serious remark?
19:16.26justinu|laptopHmmhesays: why it's in the makefile, you'd have to ask digium
19:16.35justinu|laptopi'd just remove it
19:16.41Hmmhesaysyeah i'm going to
19:16.58*** join/#asterisk watchy (n=watchy@office2.gwhsi.com)
19:17.02watchyaynone here run an isp?
19:17.07Zhadnostasterisknewbiezz> hylafax? yeah. Look up iaxmodem on voip-info.org, it lets an iax channel act as a virtual modem.
19:17.24jorgeikedahi
19:17.44CunningPikefunxion: Say what?
19:17.53funxionlol
19:17.54rob0I confess, it was a lousy attempt at a joke.
19:18.01jorgeikedaanyone knows how to make the chan_unicall.so work? I´m trying to make it work with asterisk-1.2.5
19:18.40funxiononce I place a dial command is there anyway to update a table in mysql before the call hangs up?
19:19.44CunningPikefunxion: Not before the call hangs up I don't think so, unless you do some funky agi thing
19:19.55funxiontahts what I thought
19:20.01Hmmhesaysif I'm cross compiling can I pull *.h files from another platform?
19:20.07funxionbut was hoping someone in here might be able to say different
19:20.20CunningPikefunxion: You can do stuff after hangup.....
19:20.24funxionI know
19:20.27justinu|laptopapp_dial... asterisk's achilles heel
19:20.47funxionlol
19:21.09Hmmhesaysyeah you need a really really big microscope
19:21.11justinu|laptopHmmhesays: i don't see why not
19:21.15watchyno one here run an isp?
19:21.26funxionits like when osama bin laden got pantsed on south park
19:21.38rpmwatchy: an isp or itsp?
19:21.43funxionif anyone has seen that episode
19:21.50watchyisp. im curious if i should subnet my clients
19:22.01watchyand i would imagine someone here actually runs an isp
19:22.08justinu|laptopbridging is lame
19:23.17funxionis there no way to run two commands on one priority of a context?
19:23.26rpmsubnet your clients? you mean like give them a /30 subnet and over-populate your routing table?
19:23.48watchyrpm: yea, so i should just leave em on 255.255.255.0?
19:24.03watchywell 255.255.254.0 i mean
19:24.12watchyi got a /23 and a /24
19:24.50*** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com)
19:25.34rpmyes. just have network, most people have basic firewalls to prevent eachother from conencting to eachother
19:26.03rpm"just have one network"
19:27.15watchywell im having issues with broadcast issues i think
19:27.28justinu|laptopsee... bridging sucks
19:32.56mountainm2kanybody know of a good Polycom distributor?
19:33.22mountainm2kI've just been told that Atacomm, where I've gotten a few phones to test with, is _not_ authorized
19:33.24justinu|laptopwhat do you need? rock bottom price?
19:33.31justinu|laptopatacomm also rapes you on shipping
19:33.38mountainm2kWell, 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)
19:33.59mountainm2kI really don't care if they know how to support Asterisk, because I think I'm going to buy the Digium support
19:34.12justinu|laptopi 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
19:34.16mountainm2k(company, you know -- some crap about mission critical, yadda)
19:34.33mountainm2kare they authorized distributor?
19:35.12*** join/#asterisk denon (i=denon@synapse.subneural.net)
19:35.12*** mode/#asterisk [+o denon] by ChanServ
19:35.25justinu|laptopi believe so
19:37.35*** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net)
19:38.20[TK]D-Fendermountainm2k : Costs a bit more, but VoIPSupply
19:38.49mountainm2k<PROTECTED>
19:38.52[TK]D-Fendermountainm2k : Whats your troule with them?
19:38.56[TK]D-Fendermountainm2k : Apparently
19:38.58justinu|laptopi've been dicked by voipsupply too many times
19:39.15justinu|laptopsaying they have shit in stock, when its not
19:39.17TripleFFFFweird.. a rep at voipsupply is anmed justin
19:39.26justinu|laptopsending me bad phones, taking over a month to replace them
19:39.37TripleFFFFjustinu|laptop wich phones ?
19:39.44justinu|laptopcisco 7960
19:39.47TripleFFFFanyone here have the pap2t nam admin pdf ?
19:39.55TripleFFFFvoipcupply didnt send it.. lol
19:39.58TripleFFFFk JunK-Y
19:40.02TripleFFFFk justinu
19:40.02justinu|laptopthey actually sent me a bad 7960 and a bad power brick
19:40.15justinu|laptopeither way, the experience was unpleasant...
19:40.15[TK]D-Fenderjustinu : Taken me often nearly that long WITH my auth'd Polycom reseller
19:40.15TripleFFFFyeah on phone you need to consider 10% fail
19:40.22mountainm2khah, Atacomm on the phone...
19:40.42TripleFFFFthey are eorgansiing due to grotwh and seperating resel from end user
19:40.55TripleFFFFso they will be better.. myself got 30/60.. 30 pleasent 60 unpl
19:40.57luke-jr_Any suggestions for a SellVoIP-like ITSP?
19:41.07TripleFFFFluke-jr_ whant u need ?
19:41.10TripleFFFFwhat you need ?
19:41.15TripleFFFFtheclubvoip.com
19:41.17TripleFFFF;)
19:41.22justinu|laptopso i threw voipconnection a bone, and they came thru... very good communication, prompt shipping, etc.
19:41.34luke-jr_TripleFFFF: cheap DIDs and decent service?
19:41.40TripleFFFFok how cheap dids
19:41.41TripleFFFFand where
19:41.44TripleFFFFlocation
19:41.53TripleFFFFwhere are orig/term going to be
19:41.53luke-jr_SellVoIP's were $1/mo
19:42.12justinu|laptopso why not go with them?
19:42.12TripleFFFF1$ a month for how much.. and. cheaper you pay less support/service you get you know..
19:42.20luke-jr_justinu: they don't really exist anymore
19:42.21TripleFFFFi can get them at 0.10 cent
19:42.23justinu|laptopoh
19:42.29TripleFFFFbut i need 5000
19:42.36justinu|laptopwe pay 50c/mo for our DIDs
19:42.44luke-jr_TripleFFFF: heh, I don't have that many users
19:42.48TripleFFFFthats the point
19:42.56TripleFFFFusa luke-jr_ ?
19:42.59luke-jr_yeah
19:43.04TripleFFFFk
19:43.56luke-jr_oh, and no flash UIs =p
19:44.13*** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com)
19:44.35luke-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)
19:46.17Hmmhesaysbah why doesn't gcc give the full path it is looking for the headers in
19:47.36SparFuxI 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?
19:48.20X-Gensomeone reboot sourceforge please
19:49.02TripleFFFFlol
19:50.08*** join/#asterisk postel_ (n=jp@unaffiliated/postel)
19:50.09CunningPikeX-Gen: Doine
19:50.13CunningPike:)
19:50.32*** join/#asterisk skac (i=ash@kakistocracy.co.uk)
19:51.08CunningPike:)
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)
19:53.33southtelIs there a cli command to kill a given channel?
19:53.51TripleFFFFsoft hangup
19:54.08southtelBeautiful! Thanks.
19:54.17skacanyone 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)
19:54.33skacvonage is doing my head in, so i am going to go on to a new provider soon.
19:54.38Hmmhesaysgive me 50 bucks and i'll do it
19:54.44skacno thanks.
19:54.53HmmhesaysI use them all the time with no problems
19:55.20skacthey 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)
19:55.48skacHmmhesays: got any documentation on how you did it?
19:55.58justinu|laptopthey work like any other sip provider
19:56.04skaci see.
19:56.08Hmmhesayseverything you need is on the wiki
19:56.25justinu|laptopafaik, 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)
19:56.26Hmmhesaysi need a fscking cigarette
19:56.34stack_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?
19:56.36skacjustinu|laptop: damn
19:56.42Hmmhesaysor a business account
19:56.45skacyeah
19:56.48justinu|laptopskac: if you threaten to cancel, they might change their tune
19:57.01skac*nod*
19:57.10skacare the details not guessable?
19:57.13justinu|laptopno
19:57.17skacokey.
19:57.24justinu|laptopwell, yes
19:57.33justinu|laptopbut it would take hundres of years, unless you're the NSA
19:57.42skacah haha.
19:57.57Hmmhesaysmy password has an underscore in it
19:58.07justinu|laptopstack: progress means you've got inband tones available, generally
19:58.07Hmmhesaysso good luck with that
19:58.11CrashHDwhat 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)
19:58.13CrashHD?
19:58.26stack_justinu|laptop: and what does that mean? ;p
19:58.38justinu|laptopit means the network is telling you something on the voice channel
19:58.48justinu|laptoplike you call can't be completed
19:58.52smackusok, 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?
19:59.06smackusthe exten => is identical to other working phones.
19:59.08smackususing macro
19:59.11justinu|laptopsmackus: isounds like the phone isn't registered
19:59.17skacanyone know a good provider in Canada?
19:59.25smackusok.
19:59.30smackusso sip.conf?
19:59.31stack_justinu|laptop: ok, but the message for code 31 doesn't make sense to me... any idea?
19:59.51justinu|laptopstack: no, if you give me some context, i might be able to explain it better
19:59.59justinu|laptopsmackus: sip show peers
20:00.20CunningPikeCrashHD: Set(CallerID(num)=<insertDIDprefixhere${CallerID(num)})
20:00.52CunningPikesmackus: sip show peers
20:00.53stack_justinu|laptop... someone dialed a number and I got that on the console... you get a fast busy when it makes the attempt
20:01.13CunningPikeskac: Provider of what?
20:01.19justinu|laptopstack: they've probably called a number that's Non-ISDN equipment
20:01.28skacCunningPike: VoIP BPX?
20:01.29CrashHDCunningPike: so there is no general option I can set? I have to set it in the dialplan?
20:01.39skaci think.. i am very new to VoIP
20:01.40justinu|laptopstack: turn on pri debug and call it again
20:01.41*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
20:01.55CunningPikeCrashHD: Yes
20:02.04justinu|laptopstack and skac
20:02.05CrashHDCunningPike: ok thank you
20:02.09justinu|laptopwho's trying to confuse today?
20:02.12CunningPikeskac: Where are you?
20:02.22CunningPikeThere is only one CunningPike
20:02.30justinu|laptopthankfully
20:02.33justinu|laptop:)
20:03.08justinu|laptopmy 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)
20:04.13CunningPikejustinu|laptop: Hey! :P
20:04.22*** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com)
20:04.46CunningPikebiab - jog time
20:05.10justinu|laptopgo file!
20:05.29*** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home)
20:06.07smackuswouldnt sip show user show me if the phone was registered?
20:06.55smackusmy bad... you guys already answered that one
20:06.59smackusI missed it
20:08.22smackusok, so if my extension I am having issues shows up in sip show peers, is it then registered?
20:08.42SparFuxHow 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)
20:09.26Wazb^Hi all
20:09.46svenbartanyone tried sipX?
20:10.07Wazb^any idea from where i can get G729 codec (non-commercial)
20:10.49justinu|laptopsmackus: if it shows an IP address, yes
20:11.30smackusyes it does... what else could be the issue?
20:12.02justinu|laptopis qualify on?
20:12.11smackusit plays his voicemail recordings correctly. you can leave them. he can log in and listen to them.
20:12.22smackusjustinu|laptop: you asking me?
20:12.25justinu|laptopyes
20:12.34smackuswhere is that located?
20:12.40justinu|laptopsip.conf
20:12.50smackusI dont have that on any of my phones.
20:12.55justinu|laptopif it's on, you should see "OK (52ms)" in the status column
20:13.00smackuseven the ones that are working correctly
20:13.05smackusshould i add it?
20:13.10smackusqualify=on?
20:14.02justinu|laptopgive it a shot
20:14.15vader--can an fxs channel be used as a fxo channel?
20:14.31justinu|laptopno
20:15.02smackusdid not change it, still says unmonitored.
20:15.14Wazb^any idea from where i can get G729 codec (non-commercial) ?
20:15.14justinu|laptopsip reload?
20:15.23justinu|laptopWazb^: there's no such thing
20:15.26justinu|laptopyou hae to buy it from digium
20:16.00smackusyes
20:16.19justinu|laptopsmackus, something's not right then... you must have added the qualify in the wrong place
20:16.27smackuscould be, where should it be?
20:16.37justinu|laptopin the section for that phone
20:16.57smackusyeah, 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)
20:19.39vader--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)
20:19.46vader--and should i worry about it?
20:19.57*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
20:19.58X-Genvader-- yup, be afraid
20:20.02vader--really?
20:20.18X-Genit means the PC cant handle the IRQ fast enough
20:20.38X-Genmake sure card is not sharing an irq with something else
20:20.39vader--hmmm
20:20.39*** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158)
20:21.43CrashHDmusic 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)
20:22.58vader--whats the linux command to see what irq are being used by what cards?
20:23.07DrkShdwlspci
20:23.12justinu|laptopcat /proc/interrupts
20:23.14X-Gencat /proc/interrupts
20:23.18X-Gen*echo*
20:23.20X-Gen:P
20:23.41X-Genvader--, make sure your disks are using DMA
20:23.59vader--<PROTECTED>
20:24.00vader--<PROTECTED>
20:24.50X-Genanyone else care to comment ?
20:24.59*** join/#asterisk MatsK (i=MatsK@83.233.97.229)
20:26.43justinu|laptopyou 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?
20:30.06vader--ya
20:30.13vader--dual core
20:30.19*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
20:30.19vader--i believe
20:30.53vader--justinu i am providing analog for a couple lines
20:30.56vader--fax machines and stuff
20:31.12_alex_mx_vader--, are you dropping calls? How often do you see the message?
20:34.37websaeanyone 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)
20:39.23caloianyone feel like helping me through configuring ingress SIP channels?
20:40.09*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
20:40.32Dr-Linuxhi all
20:41.29*** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
20:42.10Dr-Linuxanybody knows about Cisco Voip conference phone?
20:42.39iqhi Dr-Linux
20:42.50*** join/#asterisk bkw__ (n=brian@asterisk/friend-and-developer/bkw)
20:42.54Dr-Linuxiq: ehh! how are you dude?
20:43.02iqDr-Linux: me good - how r u
20:43.59Dr-Linuxiq: 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
20:44.25Dr-Linuxso 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)
20:48.50iqDr-Linux: model number?
20:49.27Dr-Linuxiq: that i don't know yet sorry :)
20:49.29iqDr-Linux: I don't think it will be that different than a regular VoIP phone. These phones do have some nice Speaker system, etc.
20:49.47iqDr-Linux: array - koi baat nahi yaar. kal tak wait ker lo
20:50.24Dr-Linuxiq: 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)
20:51.20Dr-Linuxiq: bas as you know bro , mojhey iss filed say shok hai, phone kal daikhna hai, excited abhi say hoon,
20:51.21Dr-Linux:)
20:51.39iqDr-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
20:52.01iqDr-Linux: hota hai yaar... sab ke saath hota hai :)
20:52.16DaminDr-Linux: WTF?
20:52.37*** part/#asterisk Vorondil (n=cwaldeck@mail.yhamerica.com)
20:52.40Dr-LinuxDamin: STFU
20:52.49Dr-Linuxiq: hehe yeah
20:52.56Dr-LinuxDamin: sorry!
20:54.09iqDr-Linux: do you know anyone who gives DID in Lahore?
20:54.31Dr-Linuxiq: US DID?
20:54.43Dr-Linuxiq: i don't think if there is any
20:54.43iqDr-Linux: no
20:54.51Dr-Linuxiq: paki?
20:54.59iqDr-Linux: Lahore
20:55.08brijnIndia
20:55.27iqbrijn: do you know anyone offers DID in India?
20:56.00brijnNope, sorry
20:56.06brijnGoogle is your friend
20:56.15iqbrijn: I like Yahoo more
21:00.41clive-this cvs thing is confusing me:)...just checked-out a shit load of junk .:)
21:01.16CrashHDhow can I get music on hold to play while the call is parked?
21:01.20clive-I mean svn
21:01.47nortexclive-, 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)
21:03.22okdohi
21:03.30findlayhello
21:03.31*** join/#asterisk skraelings001 (n=skraelin@190.40.39.154)
21:03.43clive-nortex,  I did a "svn checkout http://svn.digium.com/svn/astcc
21:03.44clive-" , I guess I should have named the "trunk" as well
21:03.45okdoanyone have any recommendations for T1 cards when splitting voice and data channels?
21:03.52okdolike use the sangoma instead of the digium, etc.
21:04.47skraelings001hi folks
21:05.43skraelings001does direct pickup application wok with any chan?
21:05.49symlinkyes
21:06.37skraelings001it does work with sip and iax channels only for me
21:07.03symlinkif 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)
21:07.44skraelings001symlink: can it be a mysql cdr?
21:08.24symlinka CDR has to exist on the channel
21:11.41*** join/#asterisk tech9iner (n=hacim@unaffiliated/tech9iner)
21:11.56tech9inerg'day all..
21:12.18tech9inercan ye mates clarify something i seem ill equipped to discern on me own please?..
21:12.49skraelings001symlink: ok, i'll check this out. thanks
21:13.31tech9ineris 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) ''' ?..
21:14.03jbalcombtech9iner: Asterisk@home is prolly a simpler choice
21:14.06tech9inergoogled n read n read every bit o documentation i can find seeking this answer..
21:14.46tech9inerahhh.. sounds like it jbalcomb !! never heard of it.. lol.. prolly RIGHT on *'s homepage huh.. lmao..
21:15.01jbalcombHow can I use the AMI to get the IPs and Extensions of all my phones?
21:15.03tech9ineroi.. tx mate.. ill dig that up..
21:15.09jbalcombg'luck
21:15.16tech9inertx mate..
21:15.26jbalcombgood on ya mate
21:17.07tech9inerlmao.. handup for ye help jbalcomb lol http://www.koalanet.com.au/australian-slang.html wink.wink............
21:17.19tech9inercheerio then..
21:17.20*** part/#asterisk tech9iner (n=hacim@unaffiliated/tech9iner)
21:18.31jbalcombHow can I use the AMI to get the IPs and Extensions of all my phones?
21:21.03creativxAction: command ? :p
21:21.22X-Robsip show peers
21:23.29*** join/#asterisk Carp1 (i=Carp1@ip-204-97-151-80.modem.logical.net)
21:23.35Carp1+---- Asterisk Installation Complete -------+
21:23.35Carp1<PROTECTED>
21:23.36Carp1<PROTECTED>
21:23.36Carp1<PROTECTED>
21:23.36Carp1<PROTECTED>
21:23.38*** join/#asterisk [hC] (n=hardcore@66.119.176.4)
21:23.44Carp1[root@pbx asterisk]# asterisk -vvvgc
21:23.45Carp1bash: asterisk: command not found
21:23.46Carp1:(
21:23.53Carp1Do I need to type a path?
21:24.33NotJohnDavid/usr/sbin/asterisk
21:24.44NotJohnDavidthat's not in your path apparently
21:24.55Carp1yup
21:24.59Carp1it starts when I type that
21:25.37Carp1How do I make it so thats the path when I type just asterisk
21:26.59NotJohnDavidadd /usr/sbin to your path
21:27.27Carp1I dont know linux all that well
21:28.04findlayCarp1: I suggest you find a good command line tutorial
21:28.14*** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66)
21:28.51esculapio__Hola quien me puede ayudar con unos reporte en asterisk via web con una database
21:30.13esculapio__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)
21:35.48NotJohnDavidwhy 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
21:35.57NotJohnDavidI want to hurt someone
21:38.30*** join/#asterisk vo (n=saigon@pdpc/supporter/basic/vo)
21:38.48CrashHDcan 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)
21:42.22FuriousGeorgeCrashHD: then you MoH isnt working at all
21:42.24FuriousGeorgeis it?
21:42.47FuriousGeorgei can change the ringer via setting my sip header right?
21:43.32CrashHDFuriousGeorge: works fine during transfer and when call is placed on hold
21:43.43FuriousGeorgeCrashHD: thats kinda odd
21:43.48CrashHDI thought so as well
21:43.55FuriousGeorgewhat version of *?
21:44.00CrashHD1.2.9.1
21:44.05FuriousGeorgeim at a loss
21:44.42CrashHDI have [default] setup
21:44.43CrashHDworks ok
21:44.52CrashHDis there a special option that needs to be set in features or something
21:45.04CrashHDI am using mp3's
21:45.07CrashHDwith format_mp3
21:45.19CrashHDI do see the moh start and stop 3 or 4 times in the CLI
21:45.24FuriousGeorgeif it works on hold it should work for parking as far as i know
21:45.51X-Robfreepbx uses format_mp3 now, and moh in parking definately works
21:46.30FuriousGeorgeanyone using snoms?  isnt it possible to change the ring tone via the sip headder?
21:46.50X-RobFuriousGeorge, yeah. you can set alert info to be a URL
21:46.53X-Robit's on the snom wiki
21:47.10esculapio__Hello who can help with me it reports in asterisk in the Web with a database
21:47.22esculapio__Hola quien me puede ayudar con unos reporte en asterisk via web con una database
21:47.27jhbhi *, I would like to find out if a certain user (1234@sipgate.de) is available, maybe using something external for agi. Any hints?
21:47.42jbalcombno hablo espanol
21:47.50*** join/#asterisk Money5ack (i=moneysac@wer.will.spontanficken.de)
21:48.05jbalcombjhb: the AMI examples have something that may be helpful
21:48.20FuriousGeorgeX-Rob: can i just change the ringer to another one of the default ringers?
21:48.42jbalcombesculapio__: are you using a reporting package or are you custom coding something?
21:49.02X-RobFuriousGeorge, I think so. Check the snom wiki for howtos
21:49.21X-RobSet(SIP_HEADER(Alert-Info: ...))something like that
21:49.29jbalcombFuriousGeorge: I believe there are instructions on how to do that with Polycoms. I would think something similar is available for the SNOMs
21:49.34CrashHDhttp://pastebin.ca/74303 is the moh problem
21:50.45jbalcombX-Rob: I see that Command: SIPpeers has the IP and Extension. Any thoughts on perl or php code to pull them out efficiently?
21:51.03jhbjbalcomb: I am looking at http://www.voip-info.org/wiki/view/Asterisk+manager+Examples - did you mean that?
21:51.36jbalcombjhb: yes'm
21:51.39jhbjbalcomb: or asked the other way around - I don't have access to the foreigns ami
21:51.54jhbjbalcomb: thx btw
21:52.00jbalcombjhb: ah, now that's rather different
21:53.02jbalcombjhb: How about just trying to make a socket connection to that address?
21:53.38jbalcombjhb: perl, python, or php probably has some module for SIP support that would let you interact enough to get a status
21:54.17jhbjbalcomb: I would not know the clients address - just their registration number at provider. So basically I am asking the proxy/registrar/whatever <- me newbie
21:54.49jhbjbalcomb: I could use twisted or so. Any hints on how to ask for a status using SIP?
21:55.21jbalcombjhb: 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)
21:56.32jhbjbalcomb: thx. I will keep on searching. Maybe I find it, and can share with you
21:56.41ariel_I have a quick Polycom question.  How can I get the 2nd call inbound the phone to ring instead of flashing the light?
21:57.35jbalcombAnyone 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-Fenderariel_ : it only goes through the handset unles you're on speaker
21:57.41jbalcombYou can be listed as a contributor to ZIPP (Z IP Phone Provisioner). =)
21:57.43De_MonI've got some users whos ISP is blocking voip traffic, how can I get around it?
21:58.01[TK]D-Fenderariel_ : But you can change the CW "beep" sound in sip.cfg for a "ringing" wav if you like.
21:58.06jbalcombDe_Mon change the port and/or use a proxy
21:58.17ariel_ok
21:58.18[TK]D-Fenderjbalcomb : Change the name.. people will ask what it's worth ;)
21:58.38ariel_but that is only in the head set
21:59.05jbalcomb[TK]D-Fender Its more a measure of what I know about developing software but I'm open to suggestions on the name. =)
21:59.13jbalcombjhb: sounds good.
22:00.11[TK]D-Fenderariel_ : Yeah, thats just the way it is... then again... what phone really rings while you're on it?
22:00.23jhbjbalcomb: you would like to have a regex or what are you searching for?
22:00.28ariel_nortels do
22:00.57[TK]D-Fenderariel_ : Norhell.. *shudder*
22:01.09jbalcombjhb: 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-Fenderariel_ : I wonder if you CAN force it somehow... nothing that stood out to me...
22:01.18ariel_yes I know but it's what I am replacing
22:01.35[TK]D-Fenderariel_ : Congrats.... I did mine last Sep....
22:01.52ariel_I have switched about 10 setups already.
22:02.04ariel_and you have different questions as you go through the change
22:02.08jhbjbalcomb: The ipaddress lines start with IPaddress, how do I (human) find the extension?
22:02.34ariel_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)
22:03.50jbalcombjhb: it's ObjectName in the same section
22:04.26ManipuraWould asterisk run any better on a dual core processor?
22:04.30jbalcombjhb: I do wish it had the User Name in there as well
22:04.40*** join/#asterisk nortex (n=nortex@64.136.65.142)
22:04.48jbalcombManipura: it is recommended by digium.
22:05.04ManipuraDual Cores are recommended?
22:05.09ManipuraAwesome....
22:05.10jbalcombManipura: They also recommend against hyperthreading because of the interrupts it generates.
22:05.23ManipuraThank you.....
22:05.26jbalcombManipura: I have Dual Dual-Core 2.8 Ghz in mine
22:05.44ManipuraI'm looking for a blade server
22:06.31jbalcombDell has nice ones, as does HP
22:06.32*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)
22:06.50jbalcombI'm out. G'Night yall.
22:07.05CunningPikeNight, jb
22:07.14Kattylater guys
22:07.16*** part/#asterisk Katty (n=aisaacs@64.82.232.54)
22:08.11*** join/#asterisk rayvd (i=rayvd@arthur.bludgeon.org)
22:08.23rayvdanyone remember what size fans were on the old oem Duron 700's?
22:08.24rayvd45mm?
22:09.32NotJohnDavidthat sounds about right
22:11.45*** join/#asterisk saftsack (n=saftsack@p54A7DF23.dip.t-dialin.net)
22:12.07rayvdi think so too :)
22:12.15Dr-Linuxanybody knows about Cisco Voip conference phone?
22:12.41Dr-Linuxdoes it come with skinny protocol?
22:13.26Qwell[]Dr-Linux: yes
22:13.30Qwell[]7935/7936
22:14.35*** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net)
22:15.08Dr-LinuxQwell[]: so just i'll need same way to load SIP firmware on it as i do on my cisco 7940/60 ?
22:15.26pdtmobiledoes anybody know if there is a reason that queues don't have but one general option?
22:15.49CunningPikepdtmobile: ??
22:15.51pdtmobileit 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
22:16.39Manipurawould having a faster hard drive ie, 10,000 or 15,000 RPM make voip any better?
22:17.24CunningPikeManipura: Unlikely - other than voicemail and moh, there's very little disk activity involved
22:17.40ManipuraThats what I thought... Thank you....
22:17.59pdtmobileCunningPike: 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.
22:18.15CunningPikeManipura: There are 3 things that make it "better": Processor, processor and processor ;)
22:18.22FuriousGeorgeso people that interface a SER sip gateway with asterisk...  how does that work?  the clients register with SER, which is registered with asterisk?
22:18.35ManipuraCunningPike, Ram doesn't do much?
22:18.36CunningPikepdtmobile: What are you trying to do?
22:18.40pdtmobilequeues only have one option thats global persistentmembers, everything else must be set for every queue or you use the in the source code defaults
22:18.56pdtmobileuse defaults ;)
22:19.08pdtmobileglobals/general settings whatever you want to call them
22:19.09*** join/#asterisk Delta239 (n=delta_of@201.218.116.114)
22:19.09Dr-LinuxQwell[]: ....
22:19.20CunningPikeManipura: I don't think so....... I think that processor and bandwidth are the real constraints
22:19.36pdtmobilebut looking through the code, if it sees a general section, it handles persistentmembers and quits
22:19.38FuriousGeorge~ser
22:19.39jbotser is probably Sip Express Router - see http://www.iptel.org/ser/
22:20.04CunningPikepdtmobile: I see
22:20.09pdtmobilebut other applications handle oodles
22:20.10vader--how well do sip phones work over vpn?
22:20.12ManipuraCunningPike, awesome, bandwidth is not a problem for me anymore ;)
22:20.17pdtmobilejust wondering if that was intentional or lazy
22:20.21vader--like as far as QOS goes
22:20.36razuis there any way to modify playback volume in asterisk with extension Playback()
22:20.37razu?
22:21.41Delta239hello.. im having some problems trying to register my broadvoice account
22:21.43CunningPikevader--: I haven't tried SIP - IAX works well
22:22.05*** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au)
22:22.09vader--i have a cisco 3000 vpn concentrator at our office
22:22.10Qwell[]Dr-Linux: I don't know if the 793x can use sip
22:22.12pdtmobilei can understand the lazy, I was just wondering if there was a reason before I "fixed" it
22:22.19CunningPikevader--: So do we :)
22:22.30vader--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
22:22.32vader--from home
22:22.47vader--maybe some sort of linksys home device
22:22.47ManipuraSo 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?
22:22.50vader--that just does VPN shit
22:23.00Dr-LinuxQwell[]: hhm...
22:23.03CunningPikevader--: Actually, ours is a 3500
22:23.09vader--you got the bigger one
22:23.11vader--:)
22:23.18vader--you shwartz is bigger than mine
22:23.20vader--your
22:23.36CunningPikeManipura: Two ways - nfs to another server, or have another asterisk server for voicemail. We do nfs
22:23.45CunningPikevader--: Heh
22:24.15vader--i was pissed when we got ours and found out it doesn't support vlan's
22:24.33ManipuraCunningPike, whats nfs stand for?
22:24.37CunningPikevader--: I'm not sure if ours does
22:24.39vader--network file system
22:24.45CunningPike~nfs
22:24.46jbotTry using -o nolock,soft (soft will help stop the error cascades, especially over wireless)
22:24.55pdtmobilevader--: snom phones if i remember correctly have vpn capability built in
22:25.35vader--i have cisco 7940G
22:26.34pdtmobilewell looks like i was wrong must have been somebody else
22:26.49pdtmobilebut I looked at a phone at one point that had that built in so you wouldn't have to carry around two devices
22:27.05Qwell[]don't all the ciscos do vlan?
22:27.16pdtmobileand 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)
22:28.27jhbjbalcomb: http://pastebin.ca/74346
22:28.40jhbgood night *
22:28.46japerryCunningPike: so I talked to verizon, they won't do a PRI for under 800
22:29.09vader--hmm i don't think opening asterisk directly to the internet is a good idea either
22:29.14pdtmobilehell I was way off
22:29.16pdtmobileit's avaya
22:29.30CunningPikejaperry: That's for a full PRI, I sincerely hope
22:29.35japerryCunningpike: 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
22:29.35pdtmobilehttp://news.thomasnet.com/fullstory/476738
22:29.42vader--we pay 385 for our 23 channel pri
22:29.46japerryCunningPike: yup. full
22:29.52japerryvader--: lucky :-P
22:29.52pdtmobilevader--: thats a good price, where you at?
22:29.55vader--paetech
22:29.58CunningPikevader--: Not to everyone, but to specific IPs might be OK
22:30.12vader--we are in usa/de
22:30.15vader--delaware
22:30.25CunningPikejaperry: We get our partial PRI from Allstream
22:30.30vader--we hired a company to go out to all the different providers and get us pricing
22:30.44vader--and we picked which one we wante
22:30.45vader--d
22:31.11japerryCunningpike: now only if it was 100miles south of the border
22:31.45pdtmobilei can't remember what we pay, we only have 13-23 on ours and I am pretty sure we pay more than that
22:31.59CunningPikejaperry: ?
22:32.36japerryCunningpike: I'm not sure if they have T1 service out here?
22:33.00CunningPikejaperry: Allstream - I don't believe so......
22:33.13FuriousGeorgeis there a way to set cadence in the dialplan for zap channels
22:33.30japerryCunningpike: anywho, so do you think it'd make any difference if we switched it to groundstart or loopstart?
22:33.38CunningPikeFuriousGeorge: Don't think so - why would you want to set it per call?
22:34.00CunningPikejaperry: It can't hurt..... also confirm your wink settings with them
22:34.51kpettitI'm in macro hell
22:35.05japerryhehe, I wonder if its featd possibly
22:35.31CunningPikekpettit: Share your pain
22:36.12*** join/#asterisk Strom_C (n=strom@gateway.digium.com)
22:36.17kpettitcan 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)
22:37.16metfanhi, I need help with a TDM400, somebody helpe!!!
22:37.32CunningPikekpettit: 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
22:37.42Strom_Cmetfan: just ask your question
22:37.48Strom_Cmetfan: someone will answer
22:37.58CunningPikekpettit: So s is a placeholder for the extension
22:38.17kpettitbugger.
22:38.23CunningPikekpettit: If you need to refer to the extension as a variable, use ${MACRO_EXTEN}
22:38.54kpettitI'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
22:39.09CunningPikekpettit: Pastebin
22:40.10metfanOk, 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??
22:41.15Dr-Linuxanybody knows if Cisco 7935/7936 supports SIP ?
22:41.53kpettitCunningPike, pastebin.com is crawling...
22:42.00kpettitany other good pastebin's?
22:42.07De_Monpastebin.ca
22:42.35De_Mon??paste
22:42.39kpettitthanks
22:42.42De_Monwrong channel
22:42.45kpettitCunningPike, http://pastebin.ca/74355
22:43.02Strom_Cmetfan: your telco needs to provision far-end disconnect supervision on your analog lines
22:43.06kpettitthe [people] context is where I have all my extensions and where I call the macro
22:44.19kpettitCunningPike, with what I have there option 2 works, but the voicemail options don't
22:44.38CunningPikekpettit: OK - looking now....
22:44.45kpettitI 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.
22:44.46kpettitthanks
22:45.06metfanStrom_C: Do I need to ask my telco for that??? what if my telco does not support it?
22:45.19Strom_Cmetfan: your telco should support it
22:45.53Strom_Cmetfan: are you in north america?
22:46.07metfanStrom_C: yes, in Mexico
22:47.39Dr-Linuxanybody knows if Cisco 7935/7936 supports SIP ?
22:47.44metfanStrom_C: that service is provisioned per line??? I mean, is it a special feature to ask to the telcos??
22:49.13kpettitCunningPike, I've tried both {MACRO_EXTEN} and {EXTEN} with the same results
22:50.02CunningPikekpettit: Ya - neither will work. ${MACRO-EXTEN} will have died because you're no longer in a macro, and ${EXTEN} will be 1 or 2
22:50.43kpettitany suggestions?
22:51.24Strom_Cmetfan: it's per-line yes
22:51.33kpettitI was thinking of doing just a context rather than a macro but then I can't pass all the var's
22:51.33Strom_Cusually the telco should automatically provision it
22:51.42Strom_Ci only know of u.s. and canada though
22:51.44CunningPikekpettit: Use Set() to set a variable in to ${MACRO_EXTEN} in your macro (or to ${EXTEN} in your original context) and refer to that
22:52.32CunningPikekpettit: Set(foo=${MACRO_EXTEN})
22:53.05kpettitthen I just call ${foo} ??
22:53.06CunningPikekpettit: Then, exten => 1,1,VoiceMail(${foo}|u)
22:53.18kpettitah cool, let me try that.  thanks
22:53.53nortexIs it possible to build conference rooms dynamicly based on a extension that transfered the call to the macro/meetme application?
22:54.28CunningPikenortex: I believe so, but I have no direct experience - I think there's something in the wiki about it
22:54.33CunningPike~thewiki
22:54.34jbotthewiki is, like, at http://www.voip-info.org/wiki-Asterisk
22:55.05CunningPikenortex: Not very helpful, I'm afraid
22:55.22kpettitCunningPike, ohhh kick ass, that worked great.
22:55.29metfananybody knows if far-end disconnect supervision is a special service on most telcos or is a default service?
22:55.31kpettitCunningPike, so that's basically like setting up a global
22:55.55CunningPikekpettit: Not quite - a global would be same for all calls - the scope of that variable is per call
22:56.13kpettitgot ya.  that's pretty sweet
22:56.21CunningPikekpettit: Glad it worked out
22:56.34kpettitI'm going to go nuts with that, there are a couple places that will be really usefull for me
22:56.47CunningPikekpettit: Hee hee - go nuts ;)
22:56.50kpettitHey 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)
22:56.59kpettitI know I can do this in the source code, but I'd like to find a better way.
22:57.00nortexCunningPike, No problem, I had read through that, but it really dosen't do dynamic the way I thought it would.
22:57.02CunningPikekpettit: I can try.....
22:57.04shmaltzanybody here using an MSI AMD motherboard?
22:57.40kpettitCunningPike, 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
22:57.56kpettitI think it goes about 60 seconds and hang's up.  I'd like to be able to set that timeout.
22:58.17CunningPikekpettit: Let me check our setup.......
22:59.09kpettitwhen 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
23:00.22kpettitMy 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)
23:00.57kpettitI can do that with a 2 priority but i have to wait for that 60 seconds for that sip call back to fail
23:01.44CunningPikekpettit: OK - I have to wait for 120 seconds here :D
23:02.25shmaltzhow do I tell lilo to pass noapci to the kernel?
23:02.27kpettityou 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
23:03.08kpettitI 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
23:04.03Dr-Linuxanybody knows if Cisco 7935/7936 supports SIP ?
23:04.41CunningPikekpettit: 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)
23:05.00CunningPikekpettit: Not sure if that makes any sense at all lol
23:06.23kpettitI can see it doing the Dial, there is just no timeout set.
23:06.44kpettitAccording 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
23:07.04CunningPikekpettit: But is there a timeout on the Dial statement for that extension normally?
23:07.46kpettityou mean 700?
23:07.57kpettitor 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)
23:08.44mds2does 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)
23:08.56CunningPikekpettit: The one that got the call and is now having the call returned to it
23:09.17kpettityes
23:09.22kpettitthere is atimeout of 30 seconds
23:09.34kpettitthat'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)
23:10.11kpettitactually it's 20 seconds
23:10.27BeightoAsterisk on a cluster, is it possible?
23:10.57CunningPikekpettit: And it's executing that Dial() statement as far as you can tell?
23:10.59kpettitCunningPike, 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
23:11.06CunningPikekpettit: Ah OK
23:11.10kpettit<PROTECTED>
23:11.15kpettitthat's my excat dial statement
23:11.40CunningPikekpettit: What does your park extension look like?
23:11.52kpettitI just use the default from features.conf
23:12.08kpettitI park to 700, it gives me back 701-705 to where it parks it.
23:12.16ctaloiim 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??
23:12.50CunningPikekpettit: We have this in our extensions.conf:
23:12.54CunningPikeexten => callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1)
23:13.39CunningPikekpettit: The internal,${DIALEDPEERNUMBER},1 bit determines where the call goes on timeout, so you can make that whatever you want
23:13.41kpettitwow that's fancy
23:14.02*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
23:14.20CunningPikekpettit: It was the only way we could get the Polycom Park key to work - how are you doing it?
23:14.27kpettitwhere do you get "callpark" and "ParkAndAnnounce" from.  I haven't used those before
23:14.35kpettitI just transfer to 700
23:14.59kpettitit tells me where it's parked.  Then they page or whatever to call holler for who they want
23:15.23CunningPikekpettit: callpark is specific to Polycoms - it's what they dial when you press the Park key. ParkAndAnnouce is an asterisk application
23:15.48kpettitwhat model do you have?  I don't have a Park key on the 501
23:16.16CunningPikekpettit: It's a softkey - usually on More during a call
23:16.16kpettitwe're 90% polycom 501 and the rest 601's and 301's
23:16.23CunningPikekpettit: Same here :D
23:16.32kpettitits' not a default key is it
23:16.39CunningPikekpettit: Actually - we have no 301's
23:16.46kpettitthey suck
23:16.54CunningPikekpettit: It should be - unless you have the feature turned off....
23:17.03kpettithow do you get that soft key?  I don't have that on the default polycom image
23:17.16CunningPikekpettit: Even during a call?
23:18.07CunningPikekpettit: During a call, press 'More' - it should be there
23:18.13kpettityeah I'm on a call now and IU haved hold, endcall, transfer, confrence
23:18.22ctaloican I send all my ingress SIP calls to one context in extensions.conf and route the calls based on the call-ed number?
23:18.36kpettitI don't have a more button.  Just those 4 soft keys
23:18.52CunningPikekpettit: OK - you need to enable the feature - hang on a sec
23:19.16kpettitoh hell yeah, that'll be cool
23:20.17kpettitI figured out how to remove some keys, DND, call forwarding and that sort of hting. but that's as fancy as I've got
23:20.29CunningPikekpettit: Look in your sip.cfg for <feature and then for a feature with a name of "call-park"
23:20.42brijnexit
23:20.59kpettitok got it
23:21.06CunningPikekpettit: Set feature.x.enabled (can't remember what the x is) ="1"
23:21.10kpettitfeature.11.name="call-park" feature.11.enabled="0"
23:21.16kpettitjust change that to a 1 I guess
23:21.20CunningPikeYou got it!
23:21.31CunningPikekpettit: Then reboot your phone
23:22.26kpettitCunningPike, are you running 1.6.6?
23:22.32kpettitthat's what we've been using latgely
23:22.33CunningPikekpettit: Yes
23:22.46CunningPikekpettit: 1.6.6 with 3.1.3 bootrom
23:22.53kpettitsame here.
23:23.41kpettitI'm still anxious to figure out what your doing what that park line you gave me above, tath's really cool
23:24.36kpettitok I have the "park" option now.
23:24.44kpettitdonse'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)
23:25.12CunningPikekpettit: Basically, the Polycom will call an extension called callpark. The ParkAndAnnounce app is documented here: http://www.voip-info.org/wiki-Asterisk+cmd+ParkAndAnnounce
23:25.45kpettitah ok.
23:26.29CunningPikekpettit: The important thing for your purposes is return_context - you can send the returning call anywhere you like
23:27.38kpettit<PROTECTED>
23:27.38kpettitwhat's the differnce?
23:28.43kpettitoh this is a trip, man this is alot of fun toys to play with
23:30.42kpettitCunningPike, I think this is eactly what i need though, thanks a ton.
23:30.57kpettitCunningPike, don't happen to be in the Houston area do ya?
23:30.59CunningPikekpettit: 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
23:31.08CunningPikekpettit: No - Vancouver, BC
23:31.12kpettitbugger
23:31.18CunningPikekpettit: But we have the internet........ ;)
23:31.21kpettitTrying to hire some more people that do asterisk here
23:31.37CunningPikekpettit: ssh is your friend ;)
23:32.03kpettityeah i wish.  Most of the stuff we need help for is new installs, and that needs somebody onsite.
23:32.24CunningPikekpettit: Ah
23:37.14*** part/#asterisk Beighto (n=chatzill@64.160.113.130)
23:37.16CunningPikeDr-Linux: What's your question?
23:38.12Dr-LinuxCunningPike:
23:38.15Dr-Linuxanybody knows if Cisco 7935/7936 supports SIP ?
23:38.35Dr-Linuxit'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)
23:41.37CunningPikeDr-Linux: Looks like it's SCCP only, according to voipsupply
23:42.04*** join/#asterisk terrapen (n=cjs@166.70.183.108)
23:42.13terrapenbest music-on-hold evar:
23:42.18Dr-LinuxCunningPike: so can i use it with Asterisk?
23:42.22kpettitCisco's suck for SIP.  I hate them sooooo bad
23:42.31terrapen"Yakety Sax" (the Benny Hill theme music)
23:42.46CunningPikeDr-Linux: Only if you run chan_scp - do you feel lucky? ;)
23:43.00kpettitI've got a Cisco 7940 that briked becuase a sip upgrad didn't work
23:43.02CunningPikeDr-Linux: Why not get a SoundPoint IP4000?
23:43.23CunningPikekpettit: I broke a Blackberry in similar fashion......
23:43.24terrapenkpettit, switch to polycom...best decision i made
23:43.35Dr-LinuxCunningPike: bcoz just we recieved shipment from USA and got this phone
23:43.43Qwell[]Dr-Linux: No, it doesn't...
23:43.50kpettitI'm really happy with the Polycom's
23:44.01Qwell[]http://www.cisco.com/cgi-bin/tablebuild.pl/ip-7900ser
23:44.07CunningPikeDr-Linux: Guess you're stuck running SCCP then
23:44.21Dr-Linuxi have polycom too but that's not VOIP phone
23:44.33Dr-LinuxCunningPike: i don't know how to setup SCCP
23:44.39Qwell[]Dr-Linux: send it here...I'll get it working with chan_skinny
23:44.41Qwell[]:p
23:44.47CunningPikeDr-Linux: Cut your losses and get an ATA for your Polycom
23:45.12Qwell[]kpettit: send me the b0rked 7940 too :P
23:45.38Dr-LinuxCunningPike: i have ATA as well
23:45.48CunningPikeDr-Linux: You're all set
23:46.09CunningPikeQwell[]: Want my fscked Blackberry too?
23:46.11Dr-Linuxbut this new cisco conference phone is not set :(
23:46.13Qwell[]CunningPike: sure!
23:46.54drrayQwell, you sell phones?
23:46.58Qwell[]drray: no
23:47.07Qwell[]I want to make them work, heh
23:47.36drrayqwell, 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
23:48.06CunningPikedrray: No offense, but your users must be pretty stupid
23:48.11drrayyes
23:48.13drrayI get that
23:48.18CunningPikedrray: Have you tried Polycom?
23:48.20drrayit is only one site
23:48.26drrayno.. but I will
23:49.14CunningPikedrray: 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
23:49.25drray601?
23:49.44CunningPikeIf you lock a firefighter in a bare room with two steel ball bearings, he lose one and break the other
23:49.44drrayyou have to scroll to a second screen on the cisco
23:49.55CunningPikedrray: Really? wow
23:50.01CunningPikedrray: That bitrs
23:50.07CunningPikes/bitrs/bites/
23:50.15drrayor I don't have it configured right
23:50.28kpettitCunningPike, 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 ...
23:50.37drrayand I don't need park
23:50.40kpettit<PROTECTED>
23:50.47kpettitwhich dosen't seem to do thr trick
23:51.51CunningPikekpettit: Your 'internal,*2032,2' needs to be a Dial() style string - SIP/foo
23:52.06kpettitok
23:54.15*** join/#asterisk iq|mobile (n=iq@unaffiliated/iq)
23:55.02kpettithow are you pagin then?
23:55.14kpettitI'm using Page(Sip/XXXX) type of thing
23:55.26kpettitnot 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
23:57.09kpettitI 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
23:57.48CunningPikekpettit: We're not doing paging - we just call the parker back
23:58.04kpettitah.  I'm wanting to page just for the announce.
23:58.11kpettitthe call back does eactly whaqt I want though.
23:58.38CunningPikekpettit: Cool
23:59.09kpettitand I've set a couple short cuts so I can page through all the polycom phones
23:59.26kpettitso in my mind it would kick as to park a call and have it announces on all the polycom speakers.
23:59.38CunningPikekpettit: Neat - we haven't played with paging yet
23:59.52kpettitit's pretty cool, saved us alot in dumb paging equipment

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