| 00:09.20 | *** join/#asterisk mvand (n=mvand@CPE-65-28-181-127.neb.res.rr.com) |
| 00:21.08 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 00:30.38 | bluelinq2 | Mercestes around? |
| 00:32.48 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 00:33.44 | BZBW | anyone had tried sending fax to asterisk using t.38 and have it route to a particular extension that connected to a fax machine? |
| 00:34.08 | Mercestes | +sorta |
| 00:34.35 | JT | asterisk doesn't do t.38 endpoint |
| 00:35.13 | BZBW | I have an AudioCode and receive a PSTN incoming fax and sending it to Asterisk, and expect Asterisk to route to that particular fax line |
| 00:35.50 | JT | yeah it can't do that unless it's doing passthrough to another T.38 endpoint (1.4 only) |
| 00:35.53 | BZBW | Asterisk just need to detect this is a Fax call and route to that parcular fax extension |
| 00:36.02 | JT | that's not the problem |
| 00:36.10 | BZBW | I'm using 1.4.4 |
| 00:36.10 | JT | the problem is it does NOT do T.38 |
| 00:36.21 | JT | it doesn't do endpoint |
| 00:36.45 | BZBW | that's fine, all I need from * is to detect the fax tone and then route the call accordingly, the rest can be done between the two end points |
| 00:38.07 | JT | fax tone on a real phone line? |
| 00:39.33 | BZBW | no. fax tone is sent via voip audio, here is how it works: |
| 00:40.10 | BZBW | FAX--> PSTN--> Audiocodecs--> VoIP--> * ---> Fax Extension(sip) |
| 00:40.28 | JT | what's on the fax extension? |
| 00:42.30 | BZBW | a sip extension |
| 00:42.39 | JT | be more specific |
| 00:42.42 | JT | a phone |
| 00:42.43 | JT | or what |
| 00:43.19 | *** join/#asterisk netrat (n=agood@tlm-adsl77.konnect.net) |
| 00:43.20 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 00:43.23 | netrat | hello |
| 00:43.55 | netrat | i'm using DISA on asterisk and i'd like to forward the call to an extension if the DISA times out |
| 00:44.28 | BZBW | nope it's an ATA that connect to a FAX machine. |
| 00:44.29 | netrat | can anyone help? or maybe point me to some documentation? thanks |
| 00:44.55 | JT | BZBW: well it's a lot easier when you explain it like that |
| 00:45.02 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:45.03 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:45.04 | JT | BZBW: asterisk 1.4 is meant to passthrough T.38 |
| 00:47.27 | BZBW | I know, maybe I should explain it a bit more clear, all I need is initially when * connect to AudioCode, it detect that this is a FAX call rather than a audio call, then it will route the FAX call to the ATA for FAX, the rest can be done between AudioCode and the ATA |
| 00:48.22 | JT | i'm not sure if that's possible |
| 00:48.54 | netrat | i'm using DISA on asterisk and i'd like to forward the call to an extension if the DISA times out |
| 00:49.25 | BZBW | emm, that's what I'm trying to find out, I don't see any reason not:). |
| 00:50.00 | JT | the reason being how does it detect a fax tone on a sip call |
| 00:50.43 | BZBW | I just saw something from WIKI regarding Background() application that may detect the FAX call:) |
| 00:51.52 | JT | asterisk has some fax detection stuff, but it's not for T.38 |
| 00:51.53 | ez` | BZBW, you can play a fake ring and activate fax detec tone ... |
| 00:52.12 | ez` | why t.3 is not yet ready ... ;( |
| 00:52.19 | ez` | t.38 protocol |
| 00:52.34 | BZBW | ez`: how? |
| 00:52.35 | ez` | is it hard to do ? |
| 00:52.47 | JT | it just seems like it's not a priority for some, ez` |
| 00:53.25 | ez` | BZBW, its quit easy ... |
| 00:53.27 | BZBW | ez`: I don't need * to handle T.38, all it need is to detect this is a FAX call, then route the FAX call to a particular ATA |
| 00:53.49 | JT | it needs to handle T.38 passthrough |
| 00:53.55 | JT | and it needs to somehow hear the tone |
| 00:54.42 | ez` | we got audiocode a work x8 fxo port and i know they handle well fax ... |
| 00:54.53 | ez` | a = at |
| 00:55.14 | BZBW | all that can be handled correctly if * detect the tone. |
| 00:55.23 | JT | ez`: i think BZBW does not want to dedicate a line to fax, and needs to detect that, hence the problem |
| 00:55.31 | JT | i only use fax on numbers dedicated to fax |
| 00:56.25 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 00:56.44 | *** join/#asterisk JunK-Y (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 00:58.17 | BZBW | JT: u r not getting me, I just want to give out ONE single line to my customer, they use this for phone call or fax purpose:) |
| 00:58.37 | JT | BZBW: umm that's pretty much exactly what i just said |
| 00:58.45 | JT | BZBW: you don't want to dedicate a line to fax |
| 00:58.50 | JT | hence the potential difficulty |
| 00:59.16 | ez` | so you want to emule a ringtone for 4 sec ; if its detect fax beep ; it switch to fax ... right ? |
| 00:59.38 | *** join/#asterisk Fieldy (i=sxcHsC3M@gentoo/contributor/Fieldy) |
| 00:59.56 | JT | ez`: you're assuming the audiocodes box will send it in rtp ulaw or alaw instead of T.38 |
| 01:00.56 | BZBW | YES |
| 01:01.32 | ez` | i normaly receive or send fax using ulaw but to tell the true , on some network fax are incomple ; dunno why ... |
| 01:01.32 | JT | ez`: fax is not guaranteed to work at all over voip |
| 01:01.42 | JT | that's why you need to use FoIP, ie. T.38 |
| 01:02.21 | ez` | on some network its werk almost 99% ; but like today i never been able to send a fax even over a PRI .... |
| 01:03.25 | ez` | spanDSP make possible to handle fax ; is it same for FoIP ? |
| 01:03.47 | JT | it does FoIP |
| 01:03.50 | JT | T.38 |
| 01:03.55 | JT | but not with asterisk |
| 01:04.00 | JT | as far as i know |
| 01:04.03 | ez` | so its liek a adon ? |
| 01:04.19 | JT | T.38 only works with callweaver |
| 01:04.26 | ez` | k ; not very usefulll |
| 01:04.43 | JT | what do you mean? |
| 01:04.52 | JT | thing that is not useful is asterisk doesn't do it |
| 01:05.04 | ez` | right ... |
| 01:05.16 | JT | heaps of closed source stuff supports T.38 |
| 01:05.22 | JT | including ATAs |
| 01:06.05 | ez` | T.38 is a old shit ; and use everywhere .. |
| 01:06.20 | JT | ez`: i'm surry, but you're talking rubbish |
| 01:06.24 | JT | sorry |
| 01:06.41 | JT | T.38 is the only way to properly do realtime Fax over IP at the moment |
| 01:06.51 | ez` | i know |
| 01:07.12 | JT | then what are you saying? |
| 01:08.25 | ez` | just wondering if we could grab t.38 call and decde it with something else ; like addon < |
| 01:09.01 | JT | the problem is that asterisk needs to support it in the sip channel driver |
| 01:09.11 | JT | T.38 uses UDPTL not RTP for the payload |
| 01:09.31 | ez` | decde= decode it after ... ; after fax handshaking is done it look like its only one way sending ..... |
| 01:09.34 | ez` | k |
| 01:09.52 | JT | are you from america? |
| 01:09.55 | ez` | <PROTECTED> |
| 01:09.59 | ez` | oui |
| 01:10.10 | JT | i can hardly understand your english |
| 01:10.12 | ez` | yes ; east side ; quebc |
| 01:10.31 | JT | enough semicolon abuse already :P |
| 01:10.32 | ez` | i speak everyday english but never write i ;) |
| 01:10.37 | JT | that sounds like canada, not america |
| 01:11.11 | ez` | north america ;) |
| 01:12.06 | JunK-Y | what about i++? |
| 01:13.49 | ez` | i never write english; i learn it , speaking with people at work; |
| 01:15.54 | *** join/#asterisk jkimball4 (n=user@pc006629.mbsc.unomaha.edu) |
| 01:16.19 | jkimball4 | Is someone available to offer assistance with realtime? |
| 01:17.21 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:17.58 | [TK]D-Fender | Tabarnac, y-a trop de francophones! |
| 01:18.08 | JunK-Y | [TK]D-Fender: on le sait, on rock! |
| 01:18.16 | ez` | heheh |
| 01:18.29 | JunK-Y | [TK]D-Fender: are ya comin` tomorrow finally&? |
| 01:19.01 | [TK]D-Fender | JunK-Y, Sorry, Home theater gear is arriving, and its pool night :) |
| 01:19.24 | JunK-Y | have fun then :) |
| 01:19.39 | [TK]D-Fender | JunK-Y, 120" :D |
| 01:19.40 | ez` | my pool is 82 F ! ;) and i still next of my pool my gf in pool |
| 01:19.55 | [TK]D-Fender | ez`, je parles de billiard :) |
| 01:20.02 | ez` | hehe |
| 01:20.17 | ez` | petite diff |
| 01:20.18 | JunK-Y | on dit billard :) |
| 01:20.18 | ez` | hehe |
| 01:20.54 | [TK]D-Fender | "Je parles bilinge pour me sauver du temps ostie!" - Elvis Gratton |
| 01:21.06 | JunK-Y | on dit bilingue :P |
| 01:21.07 | JunK-Y | mouahha |
| 01:21.35 | [TK]D-Fender | JunK-Y, I can spell just fine... don't ask me to TYPE tonight ;) |
| 01:22.15 | [TK]D-Fender | JunK-Y, http://www.directdial.com/SC-PD-120.html |
| 01:23.25 | JunK-Y | nice, but i wont order more stuff from them. |
| 01:23.32 | ez` | what the max capacity conference participant asterisk allow ? ; till cpu melt ?? |
| 01:23.47 | JunK-Y | ez`: meetme? ya |
| 01:23.58 | ez` | yes meetme |
| 01:24.09 | [TK]D-Fender | JunK-Y, I'm buying from insight.ca.... was jsut a link |
| 01:24.20 | JunK-Y | so its in theory unlimited, just limited to ur machine. |
| 01:24.23 | *** join/#asterisk netrat (n=agood@tlm-adsl77.konnect.net) |
| 01:24.29 | JunK-Y | [TK]D-Fender: k |
| 01:24.35 | JunK-Y | [TK]D-Fender: julie says hi andrew. |
| 01:24.56 | BZBW | folks, I found that I can use exten ==> fax, 1, Dial(SIP/111), this seems to be the solution for * detecting an incoming fax call and route the fax call to an ATA extension, have you tried? |
| 01:25.15 | netrat | hello. i'd like to setup a dialplan that if a user doesn't dial anything they will be forwarded to an extension. i've tried the t and T extensions with no luck. |
| 01:25.23 | [TK]D-Fender | JunK-Y, Bebe don't hurt me! |
| 01:25.36 | ez` | BZBW, yo ucould do this : |
| 01:25.39 | BZBW | I have not test it, but I'm not sure if the FAX call has to go into a zaptel device, or a SIP incoming call will do:) |
| 01:26.22 | ez` | [context-incoming] |
| 01:26.22 | ez` | exten => s,1,Answer |
| 01:26.23 | ez` | exten => s,2,NVBackgroundDetect(welcome) |
| 01:26.23 | ez` | exten => s,3,Hangup |
| 01:26.27 | ez` | ; If this is a fax, dial fax line |
| 01:26.27 | ez` | exten => fax,1,Dial(SIP/5501) |
| 01:26.27 | ez` | exten => fax,2,Hangup |
| 01:26.44 | ez` | else send it to any extension |
| 01:27.20 | ez` | i am wrong ? |
| 01:27.21 | JT | ez`: i don't think that will work, but it's worth a try i guess |
| 01:27.47 | JT | the problem is the audiocodes will need to send in ulaw or alaw and then renegotiate to T.38 |
| 01:27.57 | ez` | ; If user is talking, send him to Debra |
| 01:27.57 | ez` | exten => talk,1,Dial(SIP/5502) |
| 01:27.57 | ez` | exten => talk,2,,Hangup |
| 01:28.21 | ez` | audiocode have many parameter to handle fax |
| 01:28.22 | JT | that would work if the call came in via zap |
| 01:28.32 | ez` | right .. |
| 01:28.45 | ez` | with xzapata.conn with fax xdeytect |
| 01:29.27 | ez` | i hate wype on my laptp ; letter key keep me crady `drive me crazy |
| 01:29.45 | netrat | hello. i'd like to setup a dialplan that if a user doesn't dial anything they will be forwarded to an extension. i've tried the t and T extensions with no luck. |
| 01:29.49 | ez` | oh well.. |
| 01:31.15 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 01:31.52 | ez` | netrat, whats exaclty not working ? its keep w8t for evr ? |
| 01:32.58 | netrat | ez`: if a user picks up a SIP phone and doesn't dial anything i'd like them to be forwarded to an extension, like an operator for example |
| 01:33.22 | netrat | ez`: right now it goes to a busy signal on timeout |
| 01:34.23 | ez` | define a timeout ... |
| 01:34.39 | netrat | using the t or T extension? |
| 01:35.20 | ez` | t and define timeout lenght ,,, |
| 01:35.40 | netrat | ez`: i'm using this now, exten => t,1,Dial(SIP/812) |
| 01:35.42 | JunK-Y | T is absolute Timeout |
| 01:35.44 | netrat | but it doesn't work |
| 01:35.57 | netrat | also i definied this exten => T,1,Dial(SIP/812) |
| 01:36.01 | *** join/#asterisk mrdigital-laptop (n=mrdigita@65-78-113-237.c3-0.tlg-ubr1.atw-tlg.pa.cable.rcn.com) |
| 01:36.11 | ez` | JunK-Y, know better * compare me ; listen him ;) |
| 01:37.35 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 01:37.35 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 01:39.24 | JunK-Y | netrat: thats the digitmap on the phone, nothing related to * |
| 01:39.39 | JunK-Y | do a sip debug, you will see, no sip messages will come to * |
| 01:39.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:40.29 | netrat | JunK-Y: okay. what about if the caller is coming in via DISA, is there a way to forward the caller if they don't dial any digits? |
| 01:41.19 | mrdigital-laptop | whats the best mod done to asterisk? |
| 01:41.34 | JunK-Y | im not workingso much with disa, try adding a priority after it. |
| 01:42.02 | netrat | JunK-Y: can you point me to some documentation? i've never used priorities before |
| 01:42.21 | JT | netrat: if you've used the dialplan, you've used priorities |
| 01:42.27 | JunK-Y | search for the extensions.conf |
| 01:42.31 | netrat | this is what i'm using now http://pastebin.ca/508520 |
| 01:43.21 | netrat | JunK-Y: oh okay, now i get it. yes i've tried adding a priority after the DISA to dial a SIP number, but no dice |
| 01:44.45 | *** join/#asterisk tzafrir_laptop (i=tzafrir@69-94-204-127.biltmorecomm.com) |
| 01:45.19 | JunK-Y | exten => i then |
| 01:46.25 | netrat | new dial plan http://pastebin.ca/508526 |
| 01:46.31 | netrat | the i works, but not for the timeout |
| 01:46.58 | netrat | if i try to dial an invalid extension, other than _8XXX, it forwards me |
| 01:47.05 | netrat | but it still doesn't work for the timeout issue |
| 01:47.40 | JunK-Y | take a look at app_disa.c then |
| 01:48.05 | JunK-Y | i dont think there's any handling for timeout directly in that app. |
| 01:48.27 | *** join/#asterisk DarylVOIP (n=daryl@c-71-224-42-97.hsd1.pa.comcast.net) |
| 01:48.34 | netrat | JunK-Y: is there any other way around DISA? |
| 01:48.44 | netrat | i'm not a programmer :-( |
| 01:49.39 | JunK-Y | you can handle it directly in the dialplan, no? |
| 01:50.48 | netrat | JunK-Y: i'm not for sure. i have a cisco box that forwards an external extension to the asterisk box. the asterisk box picks up on that extension and then users need to be able to dial internal extensions |
| 01:51.31 | JunK-Y | just use a background(choose-ur-extension-baby) and listen for ur internal extensions? |
| 01:51.42 | JunK-Y | t,T,i will all work. |
| 01:51.57 | netrat | okay i'll try that |
| 01:51.57 | netrat | thanks |
| 01:52.35 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 01:56.28 | netrat | JunK-Y: sorry to keep bothering you, but i'm a little confused with the example on voip-info http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Background |
| 01:56.47 | netrat | my users are coming in on extension 977, not the s extension |
| 01:57.07 | JunK-Y | so just replace s by 977 ;) |
| 01:57.48 | netrat | JunK-Y: i did and after it please the voice prompt it just hangs up |
| 01:58.11 | netrat | please=plays |
| 01:58.18 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 01:59.48 | netrat | let me try forwards 977 to s |
| 02:00.07 | JunK-Y | theres no goal to do that |
| 02:00.18 | JunK-Y | just fork background for that extension |
| 02:00.28 | JunK-Y | exten => 977,1,Background(ur_file) |
| 02:00.37 | netrat | JunK-Y: that's what i'm using |
| 02:00.41 | nick125_lappy | Once I transfer from one context to another, how do I get the name of the last context from the current context? is there even a way? |
| 02:00.57 | netrat | JunK-Y: after it plays the background file asterisk hangs up |
| 02:01.10 | *** join/#asterisk ThOr101 (n=bthorson@pool-71-126-163-76.washdc.fios.verizon.net) |
| 02:01.51 | JunK-Y | create an priority 2 |
| 02:02.04 | JunK-Y | exten => 977,2,Goto(1); |
| 02:02.28 | ThOr101 | I am setting up my TDM22B card with asterisk on fc 6 according to "the book" but I can't get the card to ring. Is there anyway I can debug at the card to make sure it is getting a "ring", I'm running asterisk with 6.02X10^23 v(s) and not getting anything. Typing in answer at the CLI replies with a "no one is calling" |
| 02:02.38 | netrat | JunK-Y: AH!! let me read some more on menus |
| 02:03.08 | JunK-Y | ThOr101: pastebin ur CLI output |
| 02:03.10 | JunK-Y | ~pb |
| 02:03.24 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 02:04.49 | ThOr101 | how many (v)s would you like with that? |
| 02:05.10 | JunK-Y | 4 |
| 02:05.13 | ez` | someone have succefuly register cisco softphone to * |
| 02:07.26 | ThOr101 | There are some ugly ANSI codes in the uploaded text. Are you ok with that, or is there a way to eliminate? |
| 02:08.08 | JunK-Y | thats oaky |
| 02:08.26 | JunK-Y | u may start asterisk -n to disable ansi colors too. |
| 02:08.43 | netrat | JunK-Y: is there anyway for WaitExten to wait for more than 1 digit? |
| 02:09.11 | JunK-Y | WaitExten waits for time, not for digits. |
| 02:09.29 | JunK-Y | just WaitExten(with a decent time here) |
| 02:09.38 | ThOr101 | http://paste.debian.net/28795 |
| 02:09.41 | ThOr101 | And TIA |
| 02:10.23 | JunK-Y | thats just a load, whats next? |
| 02:10.50 | ThOr101 | ?? That's all I get. Am I missing something? |
| 02:11.39 | ThOr101 | I can call in and type "answer" but all I get then is "no one is calling us" |
| 02:11.47 | pigpen | Hi All, question regarding asterisk manager interface (* ver 1.4.4), see pastebin: http://pastebin.ca/508580 |
| 02:12.03 | pigpen | I am trying to playback a recording directly to the speaker (intercom) |
| 02:12.15 | JunK-Y | ThOr101: thats not how u answer on zap channels, |
| 02:12.21 | pigpen | On polycom phones...but it just rings...plays back fine....but no auto answer...ideas? |
| 02:12.39 | JunK-Y | pigpen: alert-info |
| 02:13.02 | pigpen | ie: change case... |
| 02:13.03 | ThOr101 | There is obviously a large piece I am missing. I've been running from chapter one in that ORiely book. Can you point me elsewhere to the piece that I have missed? |
| 02:13.22 | JunK-Y | ThOr101: see the extensions.conf and application called answer |
| 02:13.27 | JunK-Y | application != cli command |
| 02:13.47 | pigpen | JunK-Y, changed case..no dice. |
| 02:14.15 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
| 02:14.31 | JunK-Y | homeworks time, tty tomorrow |
| 02:14.43 | ThOr101 | So I edited zapata.conf to set context=incoming |
| 02:14.52 | yidiyuehan | hi, any one knows how i can do incoming route to another server like most VoIP providers do? |
| 02:15.03 | ThOr101 | then I made an incoming in extensions.conf to answer it (I thought) |
| 02:15.17 | ThOr101 | Just like the oReily book said. |
| 02:16.07 | pigpen | yidiyuehan, just send it out like you would any other call to any other device. |
| 02:16.46 | yidiyuehan | yes, i have done this way but it always said circuit busy. |
| 02:17.01 | mrdigital-laptop | yidiyuehan: you trying to call out? |
| 02:17.05 | yidiyuehan | with remote registratioin for phone it works well. |
| 02:17.17 | yidiyuehan | yes, let's say i have two servers A, and B. |
| 02:17.56 | yidiyuehan | i create one SIP or IAX extension in server A, and cretea one SIP or IAX trunk in Server B, then i want to call from A to B by dialing this extension, is it possible? |
| 02:18.41 | ThOr101 | ok, so answer was the wrong thing to type. But I still can't get my TDM22 to answer the phone. Or asterisk actually. |
| 02:18.45 | yidiyuehan | i have achieved calls from B to A, which is trunk-to-extension call, but not extenion-to-trunk call. |
| 02:19.26 | [TK]D-Fender | ThOr101, pastebin your zaptel.conf, zapata.conf , and your channel's context from extensions.conf |
| 02:19.41 | ThOr101 | Thanks TK, I'm on it. |
| 02:19.43 | *** part/#asterisk jkimball4 (n=user@pc006629.mbsc.unomaha.edu) |
| 02:20.45 | pigpen | [TK]D-Fender, would you mind looking at my manager commands, as I am trying to playback a recording to a polycom speaker (auto answer): http://pastebin.ca/508580 |
| 02:21.24 | pigpen | If you don't have time, that's cool...I'll fiddle with it. |
| 02:21.44 | [TK]D-Fender | pigpen, Use a local channel on both ends, FORGET variables for the SIP header (SO deprecated...) |
| 02:21.59 | nick125_lappy | Anyone here have a guide for getting a PAP2-NA working with asterisk? (I can't get mine working for some reason..) |
| 02:22.23 | pigpen | [TK]D-Fender, local channel...ok...I will see what google has to offer.... thanks for the direction. |
| 02:22.37 | [TK]D-Fender | nick125_lappy, its rather dead-easy to do (only like 4 blanks to fill in), but go here if you're lost : www.voxilla.com and check out their forums |
| 02:23.01 | nick125_lappy | for some reason, I keep getting Not found from my asterisk system... |
| 02:23.05 | ThOr101 | http://paste.debian.net/28796 |
| 02:23.12 | [TK]D-Fender | pigpen, "Local/200@LocalPage" |
| 02:23.19 | [TK]D-Fender | pigpen, [LocalPage] |
| 02:23.20 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 02:23.31 | nick125_lappy | in my SIP config, I have a device named [acct-0001-01] with the username as 'nick125', I've tried both nick125 and acct-0001-01 as my userid |
| 02:23.38 | [TK]D-Fender | pigpen, exten => _XXX,1,SIPAddheader........ |
| 02:23.46 | [TK]D-Fender | pigpen, And I think you get the picture :) |
| 02:23.55 | pigpen | [TK]D-Fender, ah...using the existing page app I am using for normal intercom'ing.... |
| 02:24.15 | [TK]D-Fender | nick125_lappy, DIYCH the "username=", and set "acc-0001-01" as the user in your PAP-2 |
| 02:24.39 | anonymouz666 | when I use two 2 ast boxes through SIP, what I should use for type=? peer? friend? |
| 02:24.39 | [TK]D-Fender | pigpen, that's on the "what to do ONCE connected) context-exten-prio. |
| 02:25.24 | nick125_lappy | [TK]D-Fender: Weird, seems to work now.. |
| 02:25.32 | yidiyuehan | anonymouz666 type=peer for outgoing call, type =user for incoming call, type=friend for both |
| 02:25.33 | pigpen | right...essentially, I would use the manger to place a call to the existing intercom dialplan I already have....kinda.... |
| 02:25.50 | [TK]D-Fender | nick125_lappy, thats because the [] IS the user name... should be careful how you muck around with auth data |
| 02:26.10 | [TK]D-Fender | pigpen, Yeah, my sample was done as though you had never done anything at all :) |
| 02:26.14 | anonymouz666 | I think I heard someone saying the this config must be type=peer |
| 02:26.21 | anonymouz666 | but I don't remember why |
| 02:26.32 | ThOr101 | I found a google thing that said to recoming with #undef AUDIO_RINGCHECK, I did that and it didn't seem to make a difference so I'll go put that back to original/stock |
| 02:26.42 | ThOr101 | s/recoming/recompile |
| 02:26.45 | anonymouz666 | while Dial SIP from one box to another |
| 02:26.46 | pigpen | yeah...great..thanks yet again.... |
| 02:26.47 | [TK]D-Fender | anonymouz666, peer is used to PLACE calls, user to RECEIVE calls, and friend for BOTH |
| 02:26.53 | nick125_lappy | [TK]D-Fender: Thanks |
| 02:27.01 | [TK]D-Fender | nick125_lappy, Quite welcome |
| 02:28.27 | [TK]D-Fender | ThOr101, Your 1st channel on your TDM22 is the only 1 of 4 configured. in zapata you have it defined as FXO_LS, which would be for a PHONE, not a line. This is in disagreement with your zaptel.conf. |
| 02:28.58 | ThOr101 | Ok, let me go work on that. Thanks. |
| 02:28.59 | [TK]D-Fender | ThOr101, You also state Kewlstart in Zaptel, yet LoopStart in Zapata. |
| 02:29.14 | ThOr101 | Recommend KS everywhere? |
| 02:29.29 | [TK]D-Fender | ThOr101, Work on your consistancy and be CAREFUL. the wrong plug on the wrong port can fry things. |
| 02:29.36 | [TK]D-Fender | ThOr101, I recommend CONSISTANCY :) |
| 02:29.39 | nick125_lappy | For some reason, it seems that my asterisk install is lacking g722 support, how would I enable it? is it a ./configure time flag? |
| 02:29.56 | [TK]D-Fender | nick125_lappy, Shouldn't be.... I would think it'd be included by default. |
| 02:30.02 | [TK]D-Fender | nick125_lappy, Lemme check |
| 02:30.14 | nick125_lappy | Well, i'm having issues with 1.4.x and the g722 voices |
| 02:30.20 | nick125_lappy | [May 24 19:29:24] WARNING[444]: channel.c:2882 set_format: Unable to find a codec translation path from ulaw to g722 |
| 02:30.45 | ThOr101 | Lucky me they color coded the cards, so at least I know which ports to plug the lines with voltage into :-) |
| 02:30.50 | [TK]D-Fender | nick125_lappy, it works with stock * 1.4.2 + addons (not sure where it occurs) |
| 02:31.03 | [TK]D-Fender | ThOr101, And you know the order from top-bottom? :) |
| 02:31.12 | [TK]D-Fender | ThOr101, or perhaps the reverse? :) |
| 02:31.48 | *** join/#asterisk tonycr (n=tony@ip247-10.ct.co.cr) |
| 02:31.49 | nick125_lappy | [TK]D-Fender: Hrm... |
| 02:32.44 | ThOr101 | Top to bottom, 1-4 and the modules 1-4 from left to right. Red Red Green Green (FXO FXO FXS FXS) and the configs are the opposite of what the module is. |
| 02:32.47 | [TK]D-Fender | nick125_lappy, I see it in "show codecs" but I'm not sure where it comes from. |
| 02:32.55 | [TK]D-Fender | ThOr101, as long as YOUR sure... |
| 02:33.01 | [TK]D-Fender | YOU'RE* |
| 02:33.05 | ThOr101 | and not color blind ;-) |
| 02:33.09 | nick125_lappy | <PROTECTED> |
| 02:33.21 | nick125_lappy | It shows it here too, I guess it can't convert g722 to ulaw |
| 02:33.23 | JT | err |
| 02:33.33 | JT | asterisk doesn't support wideband audio transcoding |
| 02:33.36 | JT | ie. g.722 |
| 02:33.50 | [TK]D-Fender | JT : OMG so retarded.... |
| 02:34.01 | JT | ? |
| 02:34.03 | [TK]D-Fender | JT : Don't tell me its friggen patented... |
| 02:34.08 | JT | no |
| 02:34.19 | JT | the infrastructure for asterisk is based on 8kHz audio |
| 02:34.25 | [TK]D-Fender | JT : just that nobody wrote an interface for it? |
| 02:34.38 | [TK]D-Fender | JT : Transcode should still be possible |
| 02:34.42 | JT | [TK]D-Fender: asterisk internals need rewriting is what i've last heard |
| 02:34.48 | JT | "we're working on it" |
| 02:34.51 | nick125_lappy | Will alaw work fine (I want higher-quality than GSM)? |
| 02:34.52 | [TK]D-Fender | JT : Ick... |
| 02:35.01 | [TK]D-Fender | nick125_lappy, Hell yeah |
| 02:35.03 | JT | yes |
| 02:35.10 | nick125_lappy | Or better yet, I'll just download the ulaw ones |
| 02:35.20 | JT | nick125_lappy: what country are you in? |
| 02:35.22 | [TK]D-Fender | nick125_lappy, There isn't really a point to anything higher than G.711 |
| 02:35.31 | [TK]D-Fender | (from a reality POV) |
| 02:35.41 | nick125_lappy | JT: US |
| 02:35.50 | JT | ulaw then |
| 02:36.03 | [TK]D-Fender | indeed |
| 02:36.09 | JT | alaw for most other countries outside north america and japan |
| 02:36.24 | [TK]D-Fender | When in Rome .... (watch out for Brutus...) |
| 02:36.50 | *** join/#asterisk fbffff (n=fbffff@24-148-35-123.grn-bsr1.chi-grn.il.cable.rcn.com) |
| 02:36.57 | nick125_lappy | It really shows that it has been a very long time since I've had to work on my asterisk system, doesn't it? |
| 02:37.23 | *** join/#asterisk n00dle (n=ccraft@ip-249-27.springsips.com) |
| 02:37.56 | tonycr | I need help. I need to configure a PRI T1 using euroisdn. Because i need to connect an asterisk to a meridian pbx using euroisdn and T1. I can not use national or other protocols. |
| 02:38.23 | JT | hah |
| 02:38.43 | yidiyuehan | i create one SIP or IAX extension in server A, and cretea one SIP or IAX trunk in Server B, then i want to call from A to B by dialing this extension, is it possible? |
| 02:39.04 | *** join/#asterisk ssokol (n=ssokol@65-182-39-203.cre.bil.biltmorecommunications.net) |
| 02:39.17 | ThOr101 | so in my zapata.conf I change my signalling from fxo_ls to fxs_ks to match my zaptel.conf. Restarting *, and still no answer. Is there another move? channel => 1 so it should answer when I have a POTS plugged into the top port. |
| 02:39.17 | JT | yidiyuehan: yes |
| 02:39.22 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:39.49 | yidiyuehan | JT, what i need to do in order to get incoming calls routed to another server? |
| 02:39.51 | shido6 | yes, yidiyuehan, it is very possible |
| 02:40.02 | JT | yidiyuehan: an extension match and a Dial command |
| 02:40.19 | pigpen | [TK]D-Fender, right on the money...thanks. |
| 02:40.22 | yidiyuehan | i tried to call and it always said circuit busy. but remote extension works well |
| 02:40.54 | yidiyuehan | JT, can you explain me a bit detailed? |
| 02:41.33 | n00dle | ThOr101, no worries, we were all there at one time... just make sure that the zapata.conf has a context it's sending the call to, and you have an s extension there (at the simplest) to catch the incoming call. |
| 02:41.33 | yidiyuehan | now i have extension in server A, and trunk in server B registered to server A by this extension. and now i can make calls from server B to server A, but not reverse. |
| 02:42.53 | ThOr101 | yep, I have a context (context=incoming) and the I define that in the extensions.conf with two s extensions under the [incoming[ context |
| 02:42.57 | *** join/#asterisk unspin (n=unspin@24.82.161.85) |
| 02:43.06 | JT | tonycr: umm so have you tried settin it up yet? |
| 02:43.27 | yidiyuehan | shido6, i pm you as well, did you receive it? |
| 02:43.37 | ThOr101 | this should answer right away right? I'm not shooting myself in the foot by hanging up after 4 rings right? |
| 02:44.52 | shido6 | http://www.pastebin.ca/508620 |
| 02:44.58 | shido6 | yidiyuehan , http://www.pastebin.ca/508620 |
| 02:45.27 | tonycr | yes but it seems like asterisk reserves the channel 16 for the D channel. It thinks that is a E1 |
| 02:45.56 | Juggie | tonycr, that sounds like a misconfiguration |
| 02:46.17 | JT | tonycr: is the D channel specified in zaptel.conf? |
| 02:46.43 | tonycr | JT: yes it is |
| 02:46.53 | Juggie | tonycr, pastebin your zaptel.conf |
| 02:46.59 | Juggie | and your console output which you think is wrong. |
| 02:47.12 | JT | and zapata.conf |
| 02:48.52 | n00dle | ThOr101, Can you show me the s extensions under incoming in you extensions.conf? |
| 02:49.49 | ThOr101 | [incoming] (CRLF) exten => s,1,Answer( ) (CRLF) exten => s,2,Echo( ) |
| 02:50.28 | JT | why do you have brackets with whitespace in the middle? |
| 02:50.31 | *** join/#asterisk kavit (n=kavit@ppp167-236-231.static.internode.on.net) |
| 02:50.37 | nick125_lappy | This is weird, sometimes right after I hang up a call, my PAP2 won't allow me to dial another call out for a few seconds |
| 02:50.41 | JT | hello kavit |
| 02:50.48 | kavit | hey JT |
| 02:50.52 | kavit | hows it going? |
| 02:50.57 | JT | not too bad |
| 02:50.57 | ThOr101 | that's what it had in the book, I'll go remove and retry |
| 02:50.59 | JT | you? |
| 02:51.34 | n00dle | ThOr101, So far, so good... what does the CLI show when you call the * box? (Try invoking "asterisk -vvvvvvvr" if you need a CLI) |
| 02:51.36 | ThOr101 | still no answer |
| 02:51.38 | kavit | btw before I say anything else... asterisknow webui is buggy as hell... doesnt work with Konqueror, Opera or Seamonkey |
| 02:51.41 | JT | ThOr101: either () or no brackets at all is fine |
| 02:51.49 | kavit | JT: trying to find a ADSL tail provider |
| 02:52.01 | JT | hrm |
| 02:52.13 | JT | as in someone who owns DSLAMs? |
| 02:52.19 | kavit | JT: looking to bundle TLAN or SHDSL service with out VoIP offerings to customer... sick of sending calls over the intenet |
| 02:52.26 | JT | nice |
| 02:52.31 | kavit | JT: layer 2 |
| 02:52.53 | ThOr101 | Ok, I tried with () and with nothing. When I call the asterisk box, there is no movement on the command line using many V(s) |
| 02:53.04 | JT | but you want people who own DSLAMs, right, not resellers? |
| 02:53.13 | yidiyuehan | shido6, in this case your Server B has the extension and Server A has the trunk right? |
| 02:53.19 | yidiyuehan | and also with your config i can make calls from server A to server B am i right? |
| 02:53.26 | n00dle | ThOr101, Hang on, timer on the oven's going... brb |
| 02:54.41 | ThOr101 | :-) |
| 02:54.41 | yidiyuehan | <PROTECTED> |
| 02:54.41 | kavit | JT: either or.... as long as someone can guarantee Layer 2 connectivity and offer CBR tails |
| 02:54.41 | JT | fair enough |
| 02:54.41 | kavit | JT: and traffic to our network is unmetered :) |
| 02:54.42 | JT | heh |
| 02:54.42 | blitzrage | HI!!!!11one! |
| 02:54.42 | JT | kavit: cross connect would make the most sense there |
| 02:54.42 | ThOr101 | So there should be something on the command line when I ring the box then. |
| 02:54.59 | kavit | JT: yeah... i spoke to someone a few weeks ago... they would run and maintain a private network for us |
| 02:55.05 | kavit | but they were a bit pricey |
| 02:55.07 | *** part/#asterisk tonycr (n=tony@ip247-10.ct.co.cr) |
| 02:55.11 | JT | hrm |
| 02:55.41 | JT | all you need for a cross connect really is a piece of cat 5, 6 or fibre in globalswitch or equinix going between their rack and yours :) |
| 02:56.13 | kavit | JT: yeah thats what I told the guy and he started blabbering about layer 2 and routers and blah blah blah |
| 02:56.27 | kavit | JT: obviously looking to push his products |
| 02:56.31 | JT | right |
| 02:56.47 | kavit | JT: wanted me to buy a 4 meg - 4meg link off him |
| 02:56.49 | n00dle | ThOr101, Yep. So it seems that the channel is either not being seen by * or the ringing isn't being seen by your FXO. |
| 02:57.27 | ThOr101 | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 02:57.39 | ThOr101 | And I have the POTS line plugged into this, so that's cool right? |
| 02:57.42 | kavit | the bloody thing locks up my computer... pagefaults and what not |
| 02:57.53 | n00dle | So far. |
| 02:58.08 | kavit | JT: do you know any tail providers? |
| 02:58.10 | JT | mind you, if you're not in the same room, that'd be uber expensive, that piece of cable in globalswitch |
| 02:58.25 | JT | internode might do it |
| 02:58.27 | JT | nextep too |
| 02:58.50 | n00dle | ThOr101, what does "zap show channels" give you? |
| 02:58.52 | ThOr101 | ok, so let me try the second FXO card, by setting channel => 2 right? |
| 02:59.01 | [TK]D-Fender | ThOr101, I'd also double check your module order and the order from top-bottom that the jacks use. Confirm by configuring all 4 ports, and plug in only the PHONE to confirm the extremity |
| 02:59.32 | ThOr101 | that's the weird thing... |
| 02:59.32 | ThOr101 | *CLI> zap show channels |
| 02:59.32 | ThOr101 | No such command 'zap' (type 'help' for help) |
| 02:59.34 | kavit | JT: you are right... cross floors is a night mare but it saves a hell of a lot on cabling costs and per month bandwidth costs |
| 02:59.57 | kavit | i will see if I can get in touch with someone from Nextep.... Internode/Agile = expensive |
| 02:59.59 | n00dle | ThOr101, did you compile and install zaptel before you compiled and installed *? |
| 03:00.17 | [TK]D-Fender | oh boy.... |
| 03:00.21 | ThOr101 | yes, I've compiled it 4 times. |
| 03:00.27 | JT | kavit: equinix is far more reasonable |
| 03:00.47 | ThOr101 | when I rmmod the modules, the lights go off, when I modprobe the lights go on. the zt... tools all work |
| 03:00.49 | JT | kavit: cross connects are free from monthly charges, just a nominal cabling charge once off |
| 03:00.57 | [TK]D-Fender | ThOr101, have you recompiled * AFERWARDS? |
| 03:01.02 | JT | only connections to the telco cage have monthly charges |
| 03:01.58 | ThOr101 | hmmmm. I can't recall. I had at one point the pre-compiled zaptel, then I ripped that out and compiled my own. I can't recall if I recompiled * after that. Let me go do that. |
| 03:01.58 | kavit | JT: yeah I had a look at that... the issue is our upstream provider is in global switch |
| 03:02.10 | JT | heh |
| 03:02.29 | kavit | JT: we have a link into their machine seeing as we are on the same floor. |
| 03:02.35 | JT | yeah oh well |
| 03:02.39 | JT | 2? |
| 03:02.46 | kavit | JT: yeah |
| 03:02.52 | JT | everyone's there |
| 03:02.57 | JT | except telstra and the govt |
| 03:02.59 | JT | :) |
| 03:03.02 | kavit | :) |
| 03:03.03 | n00dle | ThOr101, Don't feel bad - I've been running * for years and got it out of order myself today... major "DUH" moment... |
| 03:03.23 | kavit | JT: i would love to go in there with a fully charged one farad capacitor and a metal bar |
| 03:03.34 | kavit | JT: would be worth all the trouble |
| 03:03.43 | JT | equinix is much more "user friendly" but their redundancy isn't quite equinix level |
| 03:03.47 | JT | especially fire supression |
| 03:03.51 | JT | err |
| 03:03.57 | JT | equinix is much more "user friendly" but their redundancy isn't quite globalswitch level |
| 03:04.01 | JT | i meant |
| 03:04.02 | ThOr101 | :-) It's ok I am running 1.4.2.1 (zaptel) with 1.2.18 (*) right? |
| 03:04.03 | kavit | yeah |
| 03:04.14 | [TK]D-Fender | ThOr101, NO |
| 03:04.21 | JT | kavit: telstra has a dedicated security guard |
| 03:04.23 | kavit | optus have offered me a tour of their rosebery and their harris street datacentre |
| 03:04.24 | JT | at gs |
| 03:04.37 | JT | where is their harris st datacentre? |
| 03:05.07 | kavit | JT: I have no idea... i just read it in an email... i presume it might be space at globalswitch |
| 03:05.11 | ThOr101 | should I downgrade zaptel, or upgrade * ? |
| 03:05.12 | JT | ah right |
| 03:05.20 | [TK]D-Fender | ThOr101, ... YES :) |
| 03:05.21 | JT | you should see their space at equinix, paranoid.... |
| 03:05.29 | FastFeet | <[TK]D-Fender>: Thanks for your help earlier... It turns out my Asterisk Box is not resolving Hostnames. |
| 03:05.40 | JT | both telstra and optus have suites on the datacentre floor, solid walls and ceilings |
| 03:05.41 | wunderkin | ThOr101, [TK]D-Fender... actually maybe.. as long as things haven't changed.. digium actually suggested i did that at one point |
| 03:05.48 | [TK]D-Fender | FastFeet, /etc/resolv.conf = your friend |
| 03:05.59 | JT | optus has CCTV cameras mounted on the cable trays pointing at the vents on the ceiling of their suite |
| 03:06.07 | blitzrage | I gotta play with SLA>.. |
| 03:06.22 | FastFeet | ya, I will play with it tommorow.. I am tired out now... |
| 03:06.23 | FastFeet | Thanks |
| 03:06.26 | FastFeet | again |
| 03:07.05 | [TK]D-Fender | FastFeet, np |
| 03:07.19 | JT | kavit: i thought the cctv camers for vents was over the top |
| 03:07.36 | n00dle | blitzrage, I got the things rebuilt and now there are applications showing in the core for SLAStation and SLATrunk, but I'm not at the office now... I won't get to play with it until tomorrow. |
| 03:07.37 | nick125_lappy | If you go from one context to another, is there a way to get that original context from the new context? |
| 03:08.00 | n00dle | nick125_lappy, Set a variable before you Goto? |
| 03:08.02 | blitzrage | n00dle: ya, not much I can do to help anyways since I've never sued it :) |
| 03:08.14 | nick125_lappy | n00dle: It's not a goto, it's simply a include |
| 03:08.15 | nick125_lappy | *an |
| 03:08.54 | n00dle | nick125_lappy, ah... that makes it appear as if all the "exten =>"s were in that context... I think... |
| 03:09.06 | nick125_lappy | n00dle: Really? |
| 03:09.19 | nick125_lappy | Verbose() has helped me so much today :p |
| 03:09.34 | nick125_lappy | n00dle: Yup, you are right. |
| 03:09.39 | n00dle | nick125_lappy, If you used a Goto, then you could set ${OLDCONTEXT} beforehand and know where you came from. |
| 03:09.53 | n00dle | ...but that may not suit. |
| 03:10.22 | ThOr101 | so zaptel 1.4.2.1 and * 1.4.4 are ok, right? |
| 03:10.28 | n00dle | blitzrage, Yeah, well, sounds like we're learning together. |
| 03:10.43 | blitzrage | n00dle: I haven't even started :) |
| 03:10.52 | [TK]D-Fender | ThOr101, yes, both are current |
| 03:10.54 | blitzrage | more interested in making chan_mobile work actually at home |
| 03:11.41 | nick125_lappy | Now here's what I need to do: strip some numbers off of the ${EXTEN} before using a Goto to go back to the original context (I wrote a context that changes a few variables based on what is passed in the number, but, when I go back to the original context, I just need a part of the number) |
| 03:11.54 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 03:12.31 | nick125_lappy | so it will process it like the part of the original number, not the actually dialed number, if that makes any sense |
| 03:13.02 | ThOr101 | does 1.4.4 do a good job of overwriting 1.2.* ? Or do I need to do a surgical removal? |
| 03:13.12 | ThOr101 | I wish there was a make uninstall |
| 03:13.14 | [TK]D-Fender | nick125_lappy, Yeah sure, not overly difficult |
| 03:13.31 | n00dle | nick125_lappy, Yeah, it makes sense... sounds more like a job for a macro maybe? Macro(dialmynum|${EXTEN:4}...) ? |
| 03:13.54 | [TK]D-Fender | ThOr101, flush out /usr/lib/asterisk/modules and recompile everything |
| 03:13.57 | nick125_lappy | example of what I'm doing: okay, I dial *343*3251625, context 1 transfers it to context 2, context 2 changes a few variables, does a few things. Now I need to only pass 3251625 back to context 1 |
| 03:14.38 | ThOr101 | [TK]D-Fender Deleted, thanks. |
| 03:15.07 | [TK]D-Fender | nick125_lappy, Goto(context1,${EXTEN:5},1) |
| 03:16.24 | n00dle | That's the ticket, [TK]D-Fender :) |
| 03:17.14 | *** join/#asterisk nullvariable (n=nullvari@66-169-41-250.dhcp.gnvl.sc.charter.com) |
| 03:17.33 | nick125_lappy | [TK]D-Fender: Thank you so much again! |
| 03:18.09 | nick125_lappy | Ugh, it does no good to make a configuration change, try it, and start yelling when it doesn't work, then realize that you forgot to reload |
| 03:19.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:22.11 | n00dle | nick125_lappy, I do that too... too easy to think cisco IOSese where it's almost all immediate. |
| 03:22.17 | [TK]D-Fender | nick125_lappy, Please report to the closest Soylent Green kiosk for further instructions :) |
| 03:22.24 | ThOr101 | Well now, that looks a little more lively |
| 03:22.34 | ThOr101 | *CLI> -- Starting simple switch on 'Zap/2-1' |
| 03:22.34 | ThOr101 | [May 24 23:21:21] NOTICE[27965]: chan_zap.c:6351 ss_thread: Got event 18 (Ring Begin)... |
| 03:22.34 | ThOr101 | [May 24 23:21:23] NOTICE[27965]: chan_zap.c:6351 ss_thread: Got event 2 (Ring/Answered)... |
| 03:22.34 | ThOr101 | <PROTECTED> |
| 03:22.34 | ThOr101 | <PROTECTED> |
| 03:22.41 | ThOr101 | sorry for the spam. Rock on! |
| 03:22.46 | [TK]D-Fender | ThOr101, YAY... though next time ... PASTEBIN |
| 03:22.53 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:22.54 | ThOr101 | Ahh right. |
| 03:23.08 | ThOr101 | So that echo thing didn't work that great, but I can tinker with that at another time. |
| 03:23.09 | n00dle | Rebuilding my * I got an error: chan_zap.c:9271: structure has no member named `call' |
| 03:23.29 | *** join/#asterisk Pegasus_RPG (n=pegasus@adsl-75-13-19-93.dsl.snantx.sbcglobal.net) |
| 03:23.30 | n00dle | I googled it, but no one else seems to have reported it anywhere. |
| 03:23.33 | [TK]D-Fender | ThOr101, Yes... you actually have a CARD as for as * is concerned now :) |
| 03:23.53 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:23.54 | [TK]D-Fender | ThOr101, Odds are....... Echo clashed with your Echo Cancellation ;) |
| 03:24.17 | Pegasus_RPG | Hi there. I'm in the USA and suddenly Caller ID is not working. That is, it shows "" <> for the caller ID in the CLI. |
| 03:24.44 | Pegasus_RPG | It used to work sometimes, but I added a ring group in FreePBX and now it doesn't ever work. |
| 03:24.49 | ThOr101 | Heh heh. Sounds like a BOFH move. machine against machine. Thank you all for your help. I'm glad I asked what I thought was the dumbest question of all (versioning) |
| 03:25.45 | [TK]D-Fender | ~freepbx |
| 03:25.56 | jbot | methinks freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 03:26.33 | Pegasus_RPG | I sympathize with that! It sure is! |
| 03:27.06 | Pegasus_RPG | Thanks for your time. Sorry to bother you. |
| 03:27.42 | n00dle | ...and I'm beginning to wonder if this old RHL9 is too old to work with. |
| 03:27.58 | nick125_lappy | It might be...what version of GCC and so on? |
| 03:28.17 | ThOr101 | oh this demo is cooooool |
| 03:28.24 | *** part/#asterisk Pegasus_RPG (n=pegasus@adsl-75-13-19-93.dsl.snantx.sbcglobal.net) |
| 03:28.38 | n00dle | Oh gods, gcc 3.2.2... that might need help... |
| 03:28.55 | Juggie | fedora core had a update project for RH9 |
| 03:29.03 | Juggie | it was shut down, but you can find some repositories still up if you look |
| 03:29.16 | Juggie | and it likely has a ton of updates for your RH9 box. |
| 03:29.32 | *** join/#asterisk markgreene (n=markgree@71-12-183-104.dhcp.leds.al.charter.com) |
| 03:30.00 | n00dle | Juggie, Thanks for the info, but I may just stick ubuntu 7.04 on instead... may be quicker! |
| 03:30.13 | nick125_lappy | n00dle: May I ask why are you using RH9.0? |
| 03:30.30 | markgreene | Hey everyone. I don't know where else to ask this and google is not turning up enough. How in gods name do I setup a Polycom 301? It does not have ANY web interface even though it's getitng an IP from the DHCP server and all the setup does is allow me to specify a tftp or dhcp setup. which I don't knwo who to do |
| 03:30.43 | n00dle | nick125_lappy, It was handy when I had to rebuild my crashed * server the last time... :-/ |
| 03:31.26 | Juggie | n00dle, http://fedoralegacy.org/ |
| 03:31.26 | n00dle | ...and went up to * version 1.2.6 |
| 03:31.26 | Juggie | you can find some working mirrors still |
| 03:31.32 | ThOr101 | Whoa, IAX. This is sooo cool. |
| 03:31.52 | n00dle | ThOr101, Yep. IAX2 beats the pants off SIP when you're going through NAT. |
| 03:32.17 | [TK]D-Fender | NAT rarely poses a real issue for * |
| 03:32.36 | JT | markgreene: polycom.com, read the sip administrator's guide? |
| 03:32.56 | markgreene | JT: it didn't make any sense to me |
| 03:33.00 | nick125_lappy | Anyone here have enum working correctly (e164.org and e164.arpa)? |
| 03:33.09 | JT | markgreene: err ok |
| 03:33.46 | [TK]D-Fender | markgreene, voipspeak.net has an easy to follow article and flash guide to setting them up and there is a good WIKI section on in on voip-info.org |
| 03:35.10 | nick125_lappy | Well, anyone using DUNDi/e164? |
| 03:35.35 | markgreene | [TK]D-Fender: thanks |
| 03:36.30 | JT | [TK]D-Fender: is it common for polycom 501s to freeze on one of the boot screens... say if there is no config server available? |
| 03:36.54 | [TK]D-Fender | JT : Nope, without a boot server it should run from the last load |
| 03:37.00 | n00dle | Juggie, no updates found for gcc on RH9... thanks for the pointer though. |
| 03:37.08 | JT | i have a 501 i got off ebay, but it just freezes at the "your IP is blah, please wait a few seconds" screen |
| 03:37.16 | markgreene | JT: [TK]D-Fender: It has NEVER been used, if that helps |
| 03:37.26 | JT | web interface is not accessible either, can't remember if it can be pinged |
| 03:37.26 | [TK]D-Fender | JT : reflash the whole thing for the latest BR/SIP |
| 03:37.34 | JT | planning to do that |
| 03:37.52 | Hmmhesays | <PROTECTED> |
| 03:38.02 | [TK]D-Fender | markgreene, if the web interface doesn't come up then that means its disabled. That can only be done in provisioning, meaning it HAS been used. |
| 03:38.07 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 03:38.18 | BSD_Tech | http://pastebin.ca/508699 have fun |
| 03:38.26 | *** join/#asterisk markgreene (n=markgree@71-12-183-104.dhcp.leds.al.charter.com) |
| 03:38.42 | markgreene | sorry - internet problems |
| 03:38.44 | [TK]D-Fender | BSD_Tech, You should host that. |
| 03:38.49 | [TK]D-Fender | markgreene, if the web interface doesn't come up then that means its disabled. That can only be done in provisioning, meaning it HAS been used. |
| 03:38.56 | BSD_Tech | I need help with it |
| 03:39.05 | BSD_Tech | its still in dev |
| 03:39.05 | JT | [TK]D-Fender: also, factory resets didn't help at all |
| 03:39.22 | [TK]D-Fender | JT : that won't clear a firmware issue. Reflash it |
| 03:39.33 | JT | i hope it's not a dud |
| 03:39.37 | markgreene | [TK]D-Fender: So would I be missing something when I try and go to the web-interface. I am just doing http://[PHONE_IP] |
| 03:40.30 | [TK]D-Fender | markgreene, Thats all you should need to do. If you don't get it then either something else is whacked, or its been disabled in prior provisioning. |
| 03:40.45 | [TK]D-Fender | markgreene, What do you see on your softkeys with it having finished booting? |
| 03:41.23 | markgreene | Setup - Start - About |
| 03:41.35 | markgreene | [TK]D-Fender: Setup - Start - About |
| 03:41.42 | markgreene | [TK]D-Fender: It is in a cycle |
| 03:41.45 | [TK]D-Fender | BSD_Tech, You're using priority jumping like NUTS in there and need to learn the term "Macro" BADLY :) |
| 03:42.01 | [TK]D-Fender | markgreene, Going nowhere mighty slow? |
| 03:42.13 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 03:42.51 | markgreene | [TK]D-Fender: It just keeps rebooting looking for a config file |
| 03:43.44 | [TK]D-Fender | markgreene, Ok well get the latest SIP & BootROM from your vendor, download the admin guide, and get cracking on those guides I've referred you to. |
| 03:43.50 | BSD_Tech | thats why I am putting it out to get help . this project neeeds to get off the ground and I am giving a start |
| 03:44.19 | markgreene | [TK]D-Fender: I ordered the phone from voip-supply. I don't know if they have the software. |
| 03:44.31 | markgreene | [TK]D-Fender: And polycom seems to make it a real pain in the ass to get from them |
| 03:44.33 | [TK]D-Fender | markgreene, They do. askt hem for it. |
| 03:45.47 | ThOr101 | So I am diving into distinctive ring (that I found docs for). My question is, if I don't want * to do anything on a normal ring, do I give it a non-existant context for the default? context=ThisDoesNotExist and a dring1context=demo ? |
| 03:46.18 | markgreene | [TK]D-Fender: How big should the file I get from them be? 14 MB? |
| 03:46.20 | markgreene | ~ |
| 03:47.02 | markgreene | [TK]D-Fender: What were the sites you sent me again? My session closed |
| 03:47.03 | [TK]D-Fender | markgreene, around there, yeah |
| 03:47.15 | [TK]D-Fender | markgreene, voipspeak.net has an easy to follow article and flash guide to setting them up and there is a good WIKI section on in on voip-info.org |
| 03:47.17 | BSD_Tech | the 611 needs work I agree on that one |
| 03:47.41 | tengulre | hi,all |
| 03:47.56 | tengulre | where have document of agi command |
| 03:48.14 | [TK]D-Fender | ~book |
| 03:48.20 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:48.22 | [TK]D-Fender | ^^^^^^ |
| 03:48.41 | ThOr101 | watch out, that book puts spaces in between its () |
| 03:49.05 | JT | ThOr101: what page number? |
| 03:49.19 | ThOr101 | hmm, I suppose I could set the default context to answer the phone after 15 rings, which would be way more than the answering machine |
| 03:49.26 | ThOr101 | Page 83 |
| 03:49.32 | *** join/#asterisk Pegasus_RPG (n=pegasus@adsl-75-13-19-93.dsl.snantx.sbcglobal.net) |
| 03:49.56 | ThOr101 | 2 lines exten => s,1,Answer( ) Copy and pasting gives you the previous, when it should be exten => s,1,Answer() |
| 03:50.17 | ThOr101 | It should also probably mention that the echo program won't work with echo cancelling enabled. |
| 03:50.18 | *** join/#asterisk VJFROMGT (n=vijay_0@190.80.51.16) |
| 03:50.57 | JT | it won't? i've never tried |
| 03:51.56 | ThOr101 | It doesn't on my system, and I think it was TK who lead me to believe that echo wouldn't work because of the cancelling. It indeed doesn't work with voice. I punched in a tone digit, and it just replayed it forever forcing me to hangup |
| 03:52.15 | JT | ThOr101: i just checked the book, there is no space there in the brackets |
| 03:52.40 | *** part/#asterisk Pegasus_RPG (n=pegasus@adsl-75-13-19-93.dsl.snantx.sbcglobal.net) |
| 03:52.45 | ThOr101 | copy it from the PDF and paste it to something. Maybe it is my PDF reader, but I doubt it. |
| 03:52.55 | ThOr101 | Evince 0.6.0 |
| 03:53.02 | JT | i did copy it somewhere |
| 03:53.12 | JT | also highlighting it demonstrates it |
| 03:53.27 | ThOr101 | weird, must be a PDF redering issue. |
| 03:53.44 | JT | may be the font, but i guess it could be your reader |
| 03:53.45 | ThOr101 | I can actually highlight the line, and see the highlight jump in that area |
| 03:55.05 | *** join/#asterisk DaveCanoe (n=Dave@adsl-70-235-73-216.dsl.mrdnct.sbcglobal.net) |
| 03:55.20 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:55.52 | *** join/#asterisk TheCops (n=henri@got.securebinary.com) |
| 03:59.23 | ThOr101 | hmm distintive ring doesn't seem to be all that dependable yet. |
| 03:59.56 | n00dle | ThOr101, Indeed, with GNOME Evince 0.8.1 it looks ok, but copy/paste puts in a space. (I'm running Ubuntu Feisty Fawn) |
| 04:00.28 | n00dle | ...but indeed there appears to be a space in the PDF. |
| 04:01.02 | ThOr101 | weird for 2 reasons. 1, why the space with evince, and not whatever JT was using. Why the ^%** is FC6 so darn old when it comes to something as simple as a PDF reader |
| 04:01.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:02.21 | JT | Adobe Acrobat |
| 04:02.36 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 04:02.46 | ThOr101 | Adobe. Hah what do they know about... Oh ... nevermind :-) |
| 04:03.27 | *** join/#asterisk markgreene (n=markgree@71-12-183-104.dhcp.leds.al.charter.com) |
| 04:03.34 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 04:03.40 | TheCops | I have an important problem with the Polycom phone, all of my phone have a lag. Im pressing keys to dial a number and sometime it get stuck and the | stop flashing..after 1 or 2 sec all is back and you can dial..it is causing some delay too with the ring and stuff like that. Seem to be a CPU load but dont know why. Im using default config file from polycom. |
| 04:04.25 | n00dle | ThOr101, That's one reason I like Ubuntu... the software updater. |
| 04:04.38 | markgreene | [TK]D-Fender: I found some older firmware, SoundPoint IP SIP 2.0.3 Rev B, and I threw it into a tftp server and had the phone boot to it. The phone has been on "Checking applicaiton..." for going on five minutes. Is that normal? |
| 04:05.01 | [TK]D-Fender | markgreene, Did it show you downloading & all that? |
| 04:05.21 | ThOr101 | Yeah, I think I'm headed in that direction. I really think it is going to bump FC out of the mainstream. It just seems overall better. |
| 04:06.12 | markgreene | [TK]D-Fender: My log file shows that the phone started a transfer but no details. I fired up wireshark to look at the communications between the phone and my comp - I am seeing a lot of packets saying "UDP CHECKSUM INCORRECT" coming from the phone |
| 04:06.53 | [TK]D-Fender | markgreene, Switch to FTP and retry. |
| 04:07.20 | markgreene | hm - what about http? I don't know how to setup an ftp server on my comp quickly |
| 04:07.36 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:07.43 | n00dle | Well, have a wonderful evening all... I'm off to sleep. |
| 04:09.15 | *** join/#asterisk Swat2 (n=bler@218-215-192-135.people.net.au) |
| 04:09.24 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 04:11.06 | ThOr101 | Incoming distinctive ring isn't in the OReily book. Got any other pointers? The docs I've seen say the command line should spit out the cadence values, but it doesn't. |
| 04:13.30 | Swat2 | Is there a way to debug a dial plan for an incomming call to see what contexts etc a call is getting routed through? |
| 04:14.20 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 04:14.24 | ThOr101 | can't you see that in the debug output at the CLI? |
| 04:14.56 | Swat2 | im a bit of a newb |
| 04:15.05 | ThOr101 | no worries, me too. |
| 04:15.21 | ThOr101 | asterisk -rvvvvvvvvvvvvv |
| 04:15.32 | ThOr101 | or start it up with c instead of r |
| 04:27.34 | *** join/#asterisk vAd0r (n=IceChat7@65.67.210.121) |
| 04:27.53 | vAd0r | can someone help me w/ the config of cisco ata 186 |
| 04:28.11 | vAd0r | sip show peers doesn't show it online |
| 04:29.08 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 04:35.56 | *** join/#asterisk mkl1525 (n=qwertz@i59F7720F.versanet.de) |
| 04:38.59 | BSD_Tech | http://pastebin.ca/508807 TK does this look ok to you I shrank the queues. |
| 04:39.26 | BSD_Tech | I am still working on the 611 to shrink it |
| 04:39.48 | *** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net) |
| 04:39.58 | mkl1525 | Hi, (* 1.2.13) I'm always getting english prompts although the german prompts are in subdirectory de and in misdn.conf there's a [general] language=de setting - do I have to set the language on another place too? |
| 04:40.16 | BSD_Tech | 1.2.13 is old |
| 04:40.22 | BSD_Tech | wow |
| 04:40.51 | JT | why would misdn.conf change your prompts setting? |
| 04:40.54 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
| 04:42.32 | BSD_Tech | man thinking in dialplan mode messes with your brain |
| 04:42.49 | *** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 04:42.59 | BSD_Tech | JT do you think that idea will work right |
| 04:43.13 | vAd0r | is tehre any thing i need to do to make this register? |
| 04:43.15 | JT | what idea? |
| 04:43.21 | BSD_Tech | http://pastebin.ca/508807 |
| 04:43.23 | vAd0r | i added the username and password to the device and teh asterisk ip |
| 04:43.53 | BSD_Tech | or shoule exten =s be changes to exten ${EXTEN} |
| 04:44.48 | BSD_Tech | insted of havding 3 full setups I wanted to shrink it |
| 04:45.17 | vAd0r | will the cisco ata register before i actually plug a phone into it |
| 04:45.25 | JT | yes |
| 04:45.28 | vAd0r | k |
| 04:45.32 | JT | it has no idea if a phone is plugged in |
| 04:45.35 | vAd0r | do i need to add something to the config |
| 04:45.38 | vAd0r | in asterisk |
| 04:45.46 | vAd0r | besides the extension |
| 04:45.49 | JT | relevant stuff in sip.conf yes |
| 04:45.57 | vAd0r | like what |
| 04:46.26 | vAd0r | it doesn't even show in my sip debug trying to register |
| 04:46.54 | JT | user/peer/friend entry |
| 04:47.02 | JT | friend would be easiest |
| 04:47.17 | vAd0r | got that |
| 04:47.39 | BSD_Tech | I have another way to to do it |
| 04:47.47 | BSD_Tech | if that wont work |
| 04:49.01 | *** join/#asterisk ssokol (n=ssokol@65-182-39-203.cre.bil.biltmorecommunications.net) |
| 04:49.25 | vAd0r | BSD im all ears |
| 04:51.22 | BSD_Tech | vador hold on |
| 04:51.26 | vAd0r | k |
| 04:51.30 | BSD_Tech | I wan tot repaste both |
| 04:51.37 | BSD_Tech | and see what you think |
| 04:52.11 | *** join/#asterisk `Sean (i=Un1x@CPE000c248d137c-CM00111ae601f8.cpe.net.cable.rogers.com) |
| 04:53.53 | BSD_Tech | http://pastebin.ca/508824 vador there |
| 04:54.26 | vAd0r | i set mine up yesterday |
| 04:54.31 | vAd0r | so kinda ocnfused w/ that |
| 04:54.56 | BSD_Tech | the idea is to make 1 macro to cover them all |
| 04:55.18 | vAd0r | its for my house |
| 04:55.23 | vAd0r | dont know what you mean cover them all |
| 04:55.35 | BSD_Tech | the goto on the 2nd one is bad hold on |
| 04:55.56 | vAd0r | what file is this in |
| 04:56.44 | vAd0r | i thought you could just plug your info in these things and it would connect like a softphone. Am i mistakken |
| 04:56.46 | vAd0r | mistaken |
| 04:57.17 | BSD_Tech | http://pastebin.ca/508827 |
| 04:57.21 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 04:57.40 | vAd0r | prob |
| 04:57.45 | vAd0r | i just setup my box yesterday |
| 04:57.52 | BSD_Tech | I am working on dial plan stuff |
| 04:57.53 | vAd0r | and want to plug my cisco ata in so i can have a phone |
| 04:57.57 | vAd0r | lol not me |
| 04:58.04 | vAd0r | i am like kindergarden |
| 04:58.09 | vAd0r | your are college |
| 04:58.11 | vAd0r | lol |
| 04:58.12 | BSD_Tech | you would need a phone to set it up |
| 04:58.20 | vAd0r | i have an analog phone |
| 04:58.26 | BSD_Tech | so you know the ip to get to the gui |
| 04:58.31 | BSD_Tech | to configure it |
| 04:58.33 | vAd0r | i was gonna plug into it after it shows registerd |
| 04:58.36 | vAd0r | yeah |
| 04:58.38 | vAd0r | on the ata |
| 04:58.40 | vAd0r | im in it |
| 04:58.44 | BSD_Tech | yeah |
| 04:58.53 | BSD_Tech | what ata make model |
| 04:59.01 | BSD_Tech | is it linksys or cisco |
| 04:59.05 | vAd0r | Cisco ATA 186 SIP |
| 04:59.09 | vAd0r | i flashed it already |
| 04:59.47 | *** join/#asterisk alexzz (n=chatzill@122.166.0.71) |
| 05:00.07 | BSD_Tech | man I have not setup one of those in aages |
| 05:00.09 | BSD_Tech | wow |
| 05:00.14 | BSD_Tech | those are old |
| 05:00.19 | vAd0r | lol i got it for free |
| 05:00.29 | vAd0r | im sure i got it ocnfigured right |
| 05:00.37 | vAd0r | is there anything special i do in shell |
| 05:01.31 | *** join/#asterisk Penggu (i=foobar@220-245-200-87.static.tpgi.com.au) |
| 05:01.52 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 05:02.11 | Penggu | hi all. with snom phones, if I try to ring one when it's on DND, i get a "temporarily unavailable" (i think 408) message |
| 05:02.12 | BSD_Tech | is it getting a ip dynamiocly |
| 05:02.18 | Penggu | is there a way to make it go to voicemail |
| 05:02.22 | vAd0r | yes |
| 05:02.24 | Penggu | or at least say the phone's on dnd ? |
| 05:02.25 | vAd0r | it is dhcp |
| 05:02.33 | vAd0r | ? |
| 05:02.40 | BSD_Tech | then you have to have a phone to get the ip |
| 05:02.55 | vAd0r | the cisco ata gets the ip |
| 05:03.02 | BSD_Tech | the phone will not say its on dnd |
| 05:03.08 | vAd0r | i will plug just a plain telephone into it |
| 05:03.09 | vAd0r | k |
| 05:03.14 | vAd0r | should i see registered though |
| 05:03.17 | BSD_Tech | you have to setup your dialplan to roll over to vm |
| 05:03.18 | vAd0r | i see status unknown |
| 05:03.29 | BSD_Tech | sip show peers |
| 05:03.34 | vAd0r | that is what i did |
| 05:03.40 | BSD_Tech | it should say registerd |
| 05:03.45 | vAd0r | 5000 and 5001 show Unknown |
| 05:03.59 | BSD_Tech | then its not registering |
| 05:04.02 | vAd0r | http://www.voip-info.org/wiki/view/Cisco+ATA+186+SIP+and+Asterisk+-+HowTo |
| 05:05.01 | JT | BSD_Tech: dude, try addressing people, it's very confusing with you trying to help 2 people at once :) |
| 05:05.01 | vAd0r | i seen this link but didn't understand everything |
| 05:05.02 | BSD_Tech | sorry JT |
| 05:06.37 | BSD_Tech | Vador ar eyou using asterisk-now or just asterisk |
| 05:06.46 | vAd0r | asterisk -r |
| 05:07.02 | vAd0r | im logged into it |
| 05:07.07 | vAd0r | not sure what you mean |
| 05:07.30 | BSD_Tech | wait your on 1.2.x right |
| 05:07.36 | BSD_Tech | then there is no gui |
| 05:07.39 | BSD_Tech | ok |
| 05:07.49 | vAd0r | 1.2 |
| 05:08.36 | BSD_Tech | ok |
| 05:08.38 | BSD_Tech | brb |
| 05:12.25 | Penggu | BSD_Tech> the phone will not say its on dnd <-- is that sought of a 'mask' so that people don't know you're trying to ignore them? (then they think your phone's stuffed or something?) |
| 05:13.13 | vAd0r | got it |
| 05:13.22 | vAd0r | let me test call one sec |
| 05:13.28 | BSD_Tech | ok depending on the phone and how you set dnd depends on what works |
| 05:13.32 | BSD_Tech | Peg |
| 05:13.48 | BSD_Tech | polycoms have dnd built into them |
| 05:13.58 | BSD_Tech | and some other phones do also |
| 05:17.11 | Penggu | i got snom 320s |
| 05:17.11 | vAd0r | can you call |
| 05:17.11 | Penggu | they got a dnd button |
| 05:17.11 | vAd0r | 8000@trix.kanetworks.com |
| 05:17.11 | Penggu | when it's on, and you call the phones, they give you 408 |
| 05:17.11 | vAd0r | to see if i can communicate |
| 05:17.11 | vAd0r | anyone |
| 05:17.11 | Penggu | the btn can be set to do other things |
| 05:17.11 | BSD_Tech | Peng you have to write you dialplan to check to see if dnd is enabled and then direct it to either play a file stating that the user is not accepting calls and then forward them to voicemail or send them to another user |
| 05:17.12 | Penggu | hmm |
| 05:17.12 | BSD_Tech | you have to write it |
| 05:17.12 | Penggu | how would you pre-check? AGI ? |
| 05:17.12 | vAd0r | can someone make a test call to me |
| 05:17.12 | BSD_Tech | not sure never done it |
| 05:17.12 | BSD_Tech | Peng what phones |
| 05:17.12 | BSD_Tech | and how ar eyou setting dnd |
| 05:17.13 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 05:17.13 | Penggu | snom320 |
| 05:17.13 | Penggu | press the "dnd" button on the phone |
| 05:17.13 | BSD_Tech | ok then asterisk will not know what to do |
| 05:17.13 | JT | BSD_Tech: just a little tip, try hitting <tab> after typing "peng" |
| 05:17.32 | vAd0r | BSD can you call my meetme |
| 05:17.32 | vAd0r | to see if my phone works |
| 05:17.32 | BSD_Tech | since your using the built in DND |
| 05:17.34 | Penggu | JT: not everyone's using mirc |
| 05:17.45 | JT | Penggu: i think most people here are NOT using mirc |
| 05:17.46 | Penggu | (or has tab-completion enabled) |
| 05:17.48 | JT | mirc is trash |
| 05:17.53 | BSD_Tech | Penggu, if you use the asterisk DND then you can set it to do other things. |
| 05:17.54 | JT | almost everything supports tab complete |
| 05:18.06 | BSD_Tech | I use XChat |
| 05:18.11 | JT | Penggu: BSD_Tech uses xchat, which supports tab complete |
| 05:18.16 | Penggu | BSD_Tech: that would mean the phone would not show "DND" on the screen when it's set? |
| 05:18.20 | JT | only crap irc clients don't do tab complete |
| 05:18.23 | BSD_Tech | I just never used it |
| 05:18.28 | Penggu | i could map the dnd button to *<number> or something |
| 05:18.38 | BSD_Tech | Penggu, Correct |
| 05:18.41 | Penggu | but with that shortcoming |
| 05:18.48 | vAd0r | lol anyone for a test call to my conference |
| 05:18.59 | BSD_Tech | Penggu, check the config file for th ephone |
| 05:19.16 | BSD_Tech | Vador setup a echotest |
| 05:19.19 | BSD_Tech | and call it |
| 05:19.23 | vAd0r | um |
| 05:19.28 | vAd0r | no idea how |
| 05:19.33 | vAd0r | 8000@trix.kanetworks.com |
| 05:19.34 | JT | BSD_Tech: tab complete saves so much time :) |
| 05:19.39 | BSD_Tech | Then you need to go read |
| 05:19.49 | BSD_Tech | JT yes it does |
| 05:19.53 | vAd0r | a simple call from someone would suffice too |
| 05:20.06 | BSD_Tech | Vador alot of people dont want to |
| 05:20.14 | vAd0r | i see that |
| 05:20.17 | Penggu | action_dnd_on_url .. hmm |
| 05:20.20 | BSD_Tech | that why you need to learn to do dial plan |
| 05:20.26 | Penggu | and an off one as well |
| 05:20.37 | Penggu | have a db with dnd people... |
| 05:21.11 | JT | i wonder if asterisk can simply handle the 408 |
| 05:21.14 | JT | that'd be much easier |
| 05:21.26 | Penggu | i guess action urls can be sip commands? i've neber gone down to raw sip stuff |
| 05:21.57 | vAd0r | *45 does nothing |
| 05:22.09 | Penggu | whats *45 ? |
| 05:22.25 | vAd0r | Make a call from your phone. (try *45 this is a local echo test) |
| 05:22.32 | BSD_Tech | Then you might have not installed all the needed sound files |
| 05:22.45 | Penggu | is that a trixbox thing? |
| 05:22.55 | vAd0r | i just google echo test asterisk |
| 05:23.00 | BSD_Tech | cool my idea for the queues worked |
| 05:23.04 | BSD_Tech | yes |
| 05:23.15 | BSD_Tech | that will now save me alot of work |
| 05:23.23 | BSD_Tech | now to fix the 611 exten |
| 05:23.34 | Penggu | JT: i think 408 is probably a side-effect of snom's dnd, but 408 could mean a whole lot of other things why a phone is temporarily unavailable |
| 05:24.23 | BSD_Tech | That means its no longer registerd with the server |
| 05:24.23 | Penggu | i'd have to use those action urls on/off to tell asterisk that dnd is being switched, store the values, and when any ext is ringed, check, if a person is on dnd then voicemail or tell the caller whats going on |
| 05:24.26 | vAd0r | 5000@trix.kanetworks.com |
| 05:24.43 | BSD_Tech | back to work on my dial plan crap |
| 05:24.48 | JT | right, but not much may normally give a 408, so it may be safe to handle a 408 from a snom as meaning the phone is DND if possible |
| 05:24.51 | vAd0r | gl on that lol |
| 05:25.02 | Penggu | can asterisk intercept 408s ? |
| 05:25.09 | JT | not sure |
| 05:25.13 | Penggu | is it one of those s-STUFF things? |
| 05:25.15 | Penggu | hmm |
| 05:25.21 | JT | i think it will convert it to a DIALSTATUS |
| 05:25.27 | JT | which might not be fine grained enough |
| 05:26.10 | Penggu | not much on google.. |
| 05:27.30 | Penggu | speaking about snoms.. i love the auto-config/update abilities tied in to dhcp |
| 05:27.51 | Penggu | all they need now is a management config utility, similar to that node management thing packaged into trixbox |
| 05:27.52 | JT | you mean the abilities that pretty much all phones have? :) |
| 05:28.11 | Penggu | well ive never played with anythign besides snoms... |
| 05:28.44 | JT | imho the snoms don't seem that spectacular |
| 05:29.22 | BSD_Tech | is iaxtel down |
| 05:30.06 | Penggu | might need to look into alternative handset then, to get a 'feel' |
| 05:30.21 | Penggu | the body of the snom isnt too good - handset falls off easily, the hang up button too sensitive |
| 05:30.23 | JT | i reckon they could be cheaper, heh |
| 05:30.29 | JT | hmm |
| 05:30.46 | JT | i think their design isn't that snazy either, personally |
| 05:30.56 | Penggu | i agree |
| 05:31.09 | Penggu | thankfully theyre doign their jobs |
| 05:31.14 | JT | polycom are the most loved around here |
| 05:31.36 | Penggu | if we eevr need more phones ill hand down my snom and get myself a polycom |
| 05:31.43 | Penggu | or something different |
| 05:31.48 | JT | ~phones |
| 05:32.01 | jbot | phones is, like, http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 05:32.02 | Penggu | /s// |
| 05:32.20 | Penggu | hmm, snoms last.. |
| 05:32.32 | ltdwk | all personal opinion |
| 05:32.59 | JT | only europeans and australians who haven't seen better seem to like snoms, heh |
| 05:33.04 | Penggu | anyone here have any linksys switches? im trying to find which web page on its server has the mac addresses per port |
| 05:33.14 | Penggu | JT: im from AU |
| 05:33.23 | kavit | Astra > Snom ? |
| 05:33.30 | JT | Penggu: be thankful it's last on the suggestable list, and it's not a grandstream :P |
| 05:33.36 | JT | kavit: yeah |
| 05:33.43 | kavit | never used an Aastra |
| 05:33.48 | kavit | might have to give it a shot |
| 05:33.55 | JT | Penggu: most vendors here in .au overprice on polycom |
| 05:33.58 | JT | by about $100 |
| 05:34.01 | Penggu | really |
| 05:34.03 | Penggu | pfft |
| 05:34.08 | Penggu | prolly why we ddint go down that path |
| 05:34.10 | JT | you can get them for a proper price if you know where to go |
| 05:34.17 | JT | that's why they're unpopular |
| 05:34.25 | JT | buy them from Westan |
| 05:34.28 | Penggu | why are they overpriced? taxes? or.. ? |
| 05:34.36 | JT | about the same as the cost of self importing from USA |
| 05:34.43 | JT | nah, retailers are just greedy |
| 05:34.44 | ltdwk | they're made by yanks, they most be worth more |
| 05:34.46 | ltdwk | right? |
| 05:34.47 | kavit | hopefully Aastra has BLF and SLA support |
| 05:35.03 | JT | looks like retailers stick $100+ profit on them |
| 05:35.08 | Strom_M | ltdwk: I'm sorry, but Californians don't count as "yanks" |
| 05:35.09 | kavit | JT: have you dealt with Unitel ? |
| 05:35.12 | JT | Westan is the official NSW distributor |
| 05:35.20 | JT | and they sell in quanitites of 1 up |
| 05:35.25 | kavit | Strom_M: *.us = yanks to the world |
| 05:35.26 | JT | for non stupid prices |
| 05:35.31 | ltdwk | Strom_M: Whatever you say, yank! |
| 05:35.37 | JT | kavit: nup |
| 05:35.39 | Strom_M | I say "boners" |
| 05:36.05 | kavit | JT: Unitel are like tier one polycom supplier.... We get our stuff for corp exp |
| 05:36.32 | JT | AUD $189 ex GST for IP301, $240 for IP430, $281 for IP501 |
| 05:36.44 | JT | kavit: yeah i think they refered me to westan |
| 05:36.57 | ltdwk | how good's the mass deployment of polycom? |
| 05:37.29 | JT | good |
| 05:37.43 | JT | they've been doing phones for longer than most other voip phone makers too |
| 05:37.47 | JT | and conferencing |
| 05:38.01 | ltdwk | got any specifics? |
| 05:38.06 | JT | well |
| 05:38.25 | JT | pretty much any movie you see with a star trek looking conferencing station from the last 20 years, is a polycom |
| 05:38.37 | ltdwk | on mass deployment |
| 05:38.41 | JT | oh |
| 05:39.00 | JT | you can provision them with ftp, tftp, ftps, http, https |
| 05:39.13 | JT | and you can set options per mac as well as in general |
| 05:39.17 | JT | using config files |
| 05:39.41 | ltdwk | dhcp boot file support too ? |
| 05:39.51 | JT | yes, naturally :) |
| 05:40.05 | JT | my favourite usability thing really is the audio quality |
| 05:40.10 | kavit | JT: what polycoms do you get off westan? maybe we can beat their prices? |
| 05:40.10 | JT | blows all softphones out of the water |
| 05:40.43 | JT | kavit: most of the ones i've got for myself have been cheaper than westan |
| 05:40.43 | ltdwk | kavit: i started using iVox for some termination... they're not too bad |
| 05:40.48 | JT | surplus deals |
| 05:40.53 | JT | and ebay specials |
| 05:40.55 | JT | and what not |
| 05:41.16 | kavit | ltdwk: good stuff.... i hope they appreciate the business i am throwing them |
| 05:41.17 | kavit | !! |
| 05:41.21 | JT | is ivox wholesale only? |
| 05:41.26 | kavit | JT: yes |
| 05:41.31 | JT | t.38? |
| 05:41.40 | kavit | JT: please not one more competitor in the market :( |
| 05:41.45 | kavit | JT: dont think so |
| 05:41.53 | ltdwk | kavit: i threw your name michaels way when i was in talks |
| 05:41.58 | JT | kavit: what competitor? |
| 05:42.02 | kavit | ltdwk: noice! |
| 05:42.20 | ltdwk | their pricing is not quite as competetive as soul |
| 05:42.22 | kavit | ltdwk: Michael is a champ |
| 05:42.34 | kavit | ltdwk: yeah but they are a lot more reliable |
| 05:42.47 | kavit | JT: i thought you were going to start a vsp :| |
| 05:42.51 | ltdwk | i've not had any reliability issues with soul, just the voice quality sucks |
| 05:42.56 | JT | who says i haven't :P |
| 05:43.01 | kavit | JT: :( |
| 05:43.11 | JT | not interested in cutprice stuff so much, no profit margin there |
| 05:43.15 | kavit | luckily i do businesses only |
| 05:43.26 | JT | kavit: i don't know really what you do, maybe it's best i don't :PO |
| 05:43.42 | Penggu | i gtg ppl |
| 05:43.44 | Penggu | thansk |
| 05:43.46 | kavit | JT: i am going to start with reselling E1s in July optus |
| 05:43.50 | Penggu | bye |
| 05:43.53 | kavit | that will be good |
| 05:44.01 | JT | i think a lot of ITSPs will go bust in the next 2 years |
| 05:44.08 | JT | 10c national will be the death of them |
| 05:44.17 | kavit | yeah JT, i think so too |
| 05:44.32 | kavit | JT: we are moving away from being VoIP only |
| 05:44.38 | ltdwk | yeah |
| 05:44.46 | JT | i hope engin either goes bankrupt, or invests a hell of a lot more into their crappy infrastructure |
| 05:44.50 | JT | kavit: nice |
| 05:44.55 | kavit | ltdwk: do you guys do ADSL tails? |
| 05:45.04 | ltdwk | i can't see engin surviving |
| 05:45.10 | JT | kavit: optus E1s, you mean you connect direct to clients with E1, or just for voip termination? |
| 05:45.18 | JT | ltdwk: they probably will, they have the numbers |
| 05:45.20 | ltdwk | kavit: yep |
| 05:45.23 | kavit | JT: clients to E1 |
| 05:45.37 | JT | kavit: how do you do that, do you have to be a telco? |
| 05:45.39 | ltdwk | kavit: i refuse to do Voip over ADSL though |
| 05:45.49 | kavit | ltdwk: who do you go through? we want to bundle ADSL services |
| 05:46.00 | kavit | JT: nah optus manage it for you |
| 05:46.04 | JT | engin gets 1000 new customers a week last i heard |
| 05:46.20 | JT | kavit: you just terminate it to a pri card? |
| 05:46.25 | kavit | JT: you need minimum spend and hosting with them... |
| 05:46.28 | BSD_Tech | grr did iaxtel die |
| 05:46.28 | JT | kavit: i assume it's a virtualised service |
| 05:46.28 | kavit | JT: yeah |
| 05:46.49 | kavit | JT: they bill you, you add a fraction on top and bill customer |
| 05:46.56 | JT | so do you actually terminate customer calls or do optus do it? |
| 05:46.57 | JT | hmm ok |
| 05:47.09 | kavit | JT: optus terminate them on to your network |
| 05:47.11 | ltdwk | kavit: I use SPT for ADSL too now, just switched from AAPT |
| 05:47.21 | kavit | hrm |
| 05:47.25 | JT | Customer > Optus E1 > Kavit server > Optus E1 ? |
| 05:47.40 | kavit | JT: thats one model |
| 05:47.48 | JT | right |
| 05:47.59 | ltdwk | I use a TW X.163 model |
| 05:48.10 | kavit | JT: the other is Optus (But really Optus) > Client > Kavit (But really Optus) > Optus |
| 05:48.14 | kavit | err |
| 05:48.15 | kavit | shit |
| 05:48.18 | kavit | i made a mess of that |
| 05:48.30 | kavit | JT: the other is Optus > Kavit (But really Optus) > Client > Kavit (But really Optus) > Optus |
| 05:49.02 | kavit | I dont touch the E1 infrastructure... i just white label it and act as an integrator |
| 05:49.11 | kavit | JT: i dont see optus going bust anytime soon |
| 05:49.22 | JT | indeed |
| 05:49.24 | JT | so umm |
| 05:49.36 | JT | calls really go from customer to optus but you bill them? |
| 05:49.42 | kavit | aye |
| 05:49.47 | JT | makes sense |
| 05:50.01 | kavit | we manage the service for them |
| 05:50.07 | JT | not sure if they're offer you a direct E1 to a customer site for a reasonable cost |
| 05:50.41 | kavit | JT: it is meant to be bulk... we are getting 5 thrunks as backup in our datacentre |
| 05:50.58 | JT | backup for voip termination? |
| 05:51.29 | kavit | JT: the model we are moving to is... Client > Dark Fibre > US (VoIP ... E1s here with other servers) > Optus |
| 05:51.43 | kavit | yeah JT back up |
| 05:51.47 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 05:51.55 | kavit | eventually we want to host clients E1 for them off site |
| 05:51.55 | JT | well they'd have to be large customers to afford dark fibre |
| 05:51.58 | ltdwk | Dark Fibre... is very expensive |
| 05:52.10 | kavit | ltdwk: tell me about it :( |
| 05:52.28 | ltdwk | it's amazing what they try to charge you just to use a small fraction of a fiber |
| 05:52.29 | kavit | JT: we also have plans to have MPLS links |
| 05:52.41 | JT | nice |
| 05:53.30 | BSD_Tech | ok round 1 ver 2 |
| 05:53.36 | BSD_Tech | is almost done |
| 05:53.36 | JT | heh, if the fibre went to the usa, i could understand the cost :P |
| 05:54.18 | ltdwk | indeed |
| 05:54.44 | JT | they're upgrading SCC to 1.2Tbit/s :D |
| 05:55.00 | ltdwk | yeah not bad for 4 fiber pairs |
| 05:55.10 | JT | 3 |
| 05:55.27 | JT | the technology has been able to do a lot higher speeds for 10 years now |
| 05:55.39 | JT | but maybe there are limitations with their transmission network |
| 05:55.47 | kavit | i hate australian telcos |
| 05:55.50 | JT | 1.6Tbit/s per pair was possible in 1998 |
| 05:56.29 | ltdwk | yeah they only have 4 pairs us<->hawaii |
| 05:56.43 | JT | well |
| 05:56.47 | JT | they claim 3 pairs |
| 05:57.08 | ltdwk | the cost of the DWDM equipment is pretty damn full on |
| 05:57.40 | JT | heh |
| 05:57.50 | JT | it's only terminating that really costs |
| 05:57.54 | JT | ie the transceivers |
| 05:58.09 | JT | the rest of dwdm is just mostly piles of Y couplers and bragg gratings |
| 05:58.52 | ltdwk | still it's not cheap |
| 05:59.28 | JT | sure, it's chicken feed in the case of southern cross cable though |
| 05:59.52 | JT | laying a 13000km+ undersea cable in dual loop topology isn't cheap |
| 06:04.15 | ltdwk | indeed |
| 06:04.26 | ltdwk | they'll be using any technology necessary to make their money back |
| 06:04.48 | JT | i can't believe Telecom NZ had enough money to be 50% shareholders :P |
| 06:05.09 | kavit | PowerTel got bought over by AAPT |
| 06:05.17 | kavit | too little too late |
| 06:05.31 | ltdwk | yeah |
| 06:05.47 | ltdwk | AAPT really went downhill...I used to get all my Data and DSL through them |
| 06:06.10 | kavit | AAPT = Telecom NZ |
| 06:06.19 | kavit | totally useless |
| 06:06.20 | kavit | :( |
| 06:08.15 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:12.39 | kavit | anyway i wish customers were less stingy.... just got an email for this guy haggling over $500, he has a gazillion dollars in his bank account |
| 06:12.52 | JT | heh |
| 06:13.19 | ltdwk | haggling over what? |
| 06:14.04 | kavit | ltdwk: wants a gsm gateway installed but doesnt want to pay me installation costs for doing it... well not as much as I quoted anyway |
| 06:14.20 | ltdwk | kavit: heh |
| 06:15.13 | kavit | really wish it was legal to punch people |
| 06:15.35 | *** join/#asterisk pabs3 (i=daemon@60-242-186-48.tpgi.com.au) |
| 06:15.43 | JT | those things cost the earth |
| 06:16.11 | pabs3 | is 1.4 or 1.2 the stable asterisk series? |
| 06:16.24 | JT | both, but 1.2 is considered more stable |
| 06:17.06 | carrar | 1.4 is sexier |
| 06:18.10 | pabs3 | hmm, ok. is it much work to convert 1.2 configs to 1.4 configs? |
| 06:18.48 | JT | depends on your configs |
| 06:18.58 | JT | stick with 1.2 unless you actually need 1.4 |
| 06:19.29 | Pagautas | hi all |
| 06:19.35 | Pagautas | i've a problem |
| 06:19.45 | Pagautas | i have extension like this |
| 06:19.45 | Pagautas | exten => _12XX,1,Dial(SIP/${EXTEN}) |
| 06:19.45 | Pagautas | exten => _12XX,2,AGI(/path/to/agi) |
| 06:19.46 | Pagautas | exten => _12XX,3,Hangup |
| 06:20.02 | Pagautas | i'd like agi to be executed after a call |
| 06:20.09 | Pagautas | but after a call |
| 06:20.18 | Pagautas | asterisk just hangup the call |
| 06:20.28 | Pagautas | ant doesn't execute agi |
| 06:20.42 | Pagautas | how could this be fixed? |
| 06:21.41 | kavit | DeadAGI() ? |
| 06:22.01 | kavit | use the h operator as well... as well |
| 06:22.12 | kavit | h operator or context or something like that |
| 06:22.18 | kavit | i forget the asterisk jargon |
| 06:22.26 | kavit | correct me here people |
| 06:22.41 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:22.43 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 06:31.54 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:36.17 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 06:44.01 | BSD_Tech | http://pastebin.ca/508937 ok there is ver 2 of my dial plan more tomarrow |
| 06:44.28 | BSD_Tech | if you hane time to make changes and then repost and point me to it . it would help. |
| 06:44.32 | BSD_Tech | night guys |
| 06:44.59 | *** join/#asterisk alexzz (n=chatzill@122.166.0.71) |
| 06:49.31 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
| 06:49.40 | drrt | hello |
| 06:55.26 | *** join/#asterisk friedrich| (n=friedric@e177246114.adsl.alicedsl.de) |
| 06:59.30 | *** join/#asterisk co-stick (n=kostya@81.95.32.154) |
| 06:59.31 | *** join/#asterisk hermuli (n=Eladamri@a88-112-255-26.elisa-laajakaista.fi) |
| 07:01.02 | co-stick | hello |
| 07:02.28 | *** join/#asterisk mkl1525 (n=qwertz@i59F720AB.versanet.de) |
| 07:02.55 | mkl1525 | Hi, (* 1.2) is there a way to login agents automatically when asterisk starts? |
| 07:03.35 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 07:04.06 | co-stick | i've got a problem with 1.4.4 & asterisk-addons 1.4.1 with ooh323... segfault when even one peer\friend exists... |
| 07:04.51 | Pagautas | co-stick: i've got a problem with this too |
| 07:05.06 | Pagautas | when there are no calls |
| 07:05.10 | Pagautas | then it works |
| 07:05.26 | Pagautas | but when there is even a few calls |
| 07:05.31 | Pagautas | asterisk crashes |
| 07:06.07 | co-stick | just after setting default context to default |
| 07:08.15 | Pagautas | if context is not default everything works fine? |
| 07:11.10 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:11.34 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 07:14.21 | co-stick | nop |
| 07:14.29 | co-stick | any context |
| 07:16.17 | Pagautas | 10:05 < co-stick> just after setting default context to default |
| 07:16.28 | Pagautas | what does then this means? |
| 07:18.06 | kavit | ltdwk: do you send your traffic to ivox over public ip? |
| 07:19.20 | co-stick | <PROTECTED> |
| 07:19.20 | co-stick | <PROTECTED> |
| 07:19.26 | co-stick | when core dump |
| 07:19.30 | co-stick | *then |
| 07:21.01 | ltdwk | kavit: yeah |
| 07:21.38 | kavit | ah ok |
| 07:22.05 | *** join/#asterisk zol_ (n=z@AClermont-Ferrand-156-1-2-34.w81-251.abo.wanadoo.fr) |
| 07:22.27 | ltdwk | my soul termination traverses only their network though |
| 07:22.38 | *** join/#asterisk matsk (n=mk@194.68.102.173) |
| 07:23.59 | ltdwk | geographically the connection is not to bad to ivox via public ip though |
| 07:24.02 | ltdwk | 64 bytes from ns1.ivox.net.au (202.83.183.44): icmp_seq=2 ttl=121 time=2.21 ms |
| 07:25.25 | kavit | thats good |
| 07:25.39 | ltdwk | gswitch -> eqx |
| 07:25.39 | JT | --- www.ivox.com.au ping statistics --- |
| 07:25.40 | JT | 4 packets transmitted, 4 received, 0% packet loss, time 2999ms |
| 07:25.40 | JT | rtt min/avg/max/mdev = 2.583/2.971/3.465/0.395 ms |
| 07:25.47 | JT | ping from mascot |
| 07:26.12 | ltdwk | yah, i'm in mascot too |
| 07:26.32 | JT | heh |
| 07:27.12 | kavit | well I live in mascot |
| 07:27.17 | ltdwk | haha |
| 07:27.21 | ltdwk | you lie |
| 07:27.25 | kavit | rosebery |
| 07:27.31 | kavit | 2 minutes from mascot |
| 07:27.46 | ltdwk | that is funny |
| 07:28.05 | kavit | if i threw a stone it would probably hit your server |
| 07:28.09 | ltdwk | JT: which provider is that IP with ? |
| 07:28.14 | co-stick | hm |
| 07:28.20 | JT | someone inside eqx |
| 07:28.22 | co-stick | afaik got it |
| 07:28.32 | ltdwk | that really narrows it down |
| 07:28.59 | co-stick | int friend_type = strcasecmp(utype, "friend"); |
| 07:29.15 | co-stick | utype is uninitializes |
| 07:29.16 | JT | indeed |
| 07:29.24 | JT | message me if you're really interested |
| 07:35.53 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.133.227) |
| 07:43.52 | walhala | hi |
| 07:43.55 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
| 07:47.39 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 07:52.51 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 07:55.56 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 07:55.58 | tengulre | hi,all |
| 07:56.44 | tengulre | I download the asterisk-gui, but I don't know how to type the url in exopler? |
| 07:58.55 | Mavvie | you do it with the same keyboard you use to enter text in IRC. |
| 07:59.33 | *** join/#asterisk drega (n=drega@mail.amcat.co.uk) |
| 08:03.21 | tengulre | Mavvie: do u answer me? |
| 08:03.36 | Mavvie | I wouldn't dare. |
| 08:04.00 | sevard | <Mavvie> I wouldn't dare. |
| 08:04.32 | sevard | i've never laughed so hard at 4a.m. |
| 08:06.46 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 08:07.40 | co-stick | ))) |
| 08:08.54 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
| 08:12.17 | mkl1525 | Hi, is there something like "autostart" that is run once on startup to run some configure scripts/commands? |
| 08:14.10 | drrt | mkl1525, hi. are u talking about * or about system in all ? |
| 08:14.24 | mkl1525 | drrt, i mean asterisk |
| 08:14.44 | *** join/#asterisk Ravi1974 (n=I@ool-18b80982.dyn.optonline.net) |
| 08:14.59 | *** join/#asterisk af_ (n=getsmart@81-174-46-93.f5.ngi.it) |
| 08:15.26 | drrt | mkl1525, you can put some comms into bash script before the asterisk starts |
| 08:17.30 | mkl1525 | drrt, I'd need it after * is started, so maybe I could start * and then run "asterisk -x" |
| 08:21.24 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 08:21.25 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) |
| 08:27.11 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) |
| 08:27.48 | drrt | mkl1525, u need to execute it in asterisk cli or in your system ? |
| 08:29.22 | mkl1525 | drrt, I'd like something to login agents set some values in * db |
| 08:31.36 | *** join/#asterisk punani (n=m@87.127.7.210) |
| 08:33.46 | drrt | mkl1525, there is no place to exec any scripts at asterisk start in my opinion. then you should start *, wait aprox time and exec comms by remote -rx |
| 08:36.25 | *** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk) |
| 08:36.36 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 08:38.00 | *** join/#asterisk ghenry (n=ghenry@fw2-inet.semantico.net) |
| 08:40.58 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 08:47.07 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 09:10.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:11.26 | *** join/#asterisk gardo (n=gardo@61.28.160.218) |
| 09:23.09 | alexzz | can somebody tell me how to setup iax.conf for 2 users in the same network with 2 different extensions |
| 09:25.21 | drrt | ~paste |
| 09:25.32 | jbot | paste is, like, http://rafb.net/paste/ |
| 09:25.49 | drrt | alexzz, ~paste your iax.conf |
| 09:28.16 | *** part/#asterisk sebastian|foo (n=sebastia@61.151.249.123) |
| 09:28.57 | *** join/#asterisk friedrich| (n=friedric@e177248030.adsl.alicedsl.de) |
| 09:34.49 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 09:35.34 | jmls | Does anyone know which sound file is used when playing the "enter sound" when someone enters a meetme conference ? |
| 09:38.11 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 09:41.41 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 09:41.54 | Zeeek | muhaha |
| 09:53.50 | Zeeek | ominous silence |
| 10:03.53 | *** part/#asterisk alexzz (n=chatzill@122.166.0.71) |
| 10:10.17 | Zeeek | human life detected |
| 10:10.39 | Siya | bleep |
| 10:11.46 | Zeeek | robot life detected |
| 10:11.56 | Zeeek | up mars. |
| 10:12.06 | Siya | ~mars |
| 10:12.17 | jbot | Java-based network services status monitor. URL: http://www.altara.org/mars.html |
| 10:12.18 | Zeeek | up Jupiter |
| 10:12.21 | Zeeek | up Uranus |
| 10:12.33 | Siya | there you go |
| 10:12.36 | Zeeek | ~asterisk |
| 10:12.46 | jbot | rumour has it, asterisk is the best free PBX in the world |
| 10:13.03 | Siya | I think jbot is biased |
| 10:13.04 | Zeeek | ~trixbox |
| 10:13.06 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 10:13.10 | sevard | holy crap, i've never seen asterisk take so long to compile |
| 10:13.29 | Zeeek | remove the pr0n dialup and it'll go faster |
| 10:13.35 | sevard | oh snap that worked |
| 10:13.48 | Siya | ~asterisknow |
| 10:13.56 | Zeeek | ~windows |
| 10:13.58 | jbot | rumour has it, windows is a 32 bit hack on a 16 bit operating system, originally designed for an 8 bit CPU, with a 4 bit system bus, made by a 2 bit company that can't stand 1 bit of competition... or the World of Warcraft bootloader, or the most important collection of bugs |
| 10:14.23 | *** join/#asterisk Snake-Eyes (n=blog@70.55.220.203.static.comindico.com.au) |
| 10:14.34 | Siya | hehe jbot doesn't know asterisknow! |
| 10:14.34 | sevard | Has anyone ever used pyastre? |
| 10:14.51 | Siya | never even heard of it |
| 10:14.57 | Zeeek | python + asterisk? |
| 10:15.04 | sevard | python agi interface for asterisk |
| 10:15.06 | sevard | ja |
| 10:15.12 | Zeeek | never heard of it |
| 10:16.24 | Zeeek | if we had less customers, there'd be room for my car in the parking lot |
| 10:16.53 | Zeeek | of course, this is simplified by the fact that we have no parking lot and I have no car... but still. |
| 10:17.33 | sevard | so.. you have zero customers? |
| 10:17.51 | Zeeek | nah, we have customers, just no car or parking |
| 10:18.28 | sevard | it follows that if you had less customers parking in your parking lot you'd have room for your car, which doesn't exist to park in the lot that the customers, which don't exist, are occupying |
| 10:18.41 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:18.53 | Zeeek | it's so nice out the entire company is going to have lunch in the park. All 2 of us. |
| 10:19.40 | Zeeek | as Mark Spencer says, "the bugtracker never takes a day off" |
| 10:19.46 | sevard | heh |
| 10:20.26 | sevard | WARNING[712]: loader.c:500 load_modules: Loading module /usr/lib/python2.4/site-packages/_pyastre.so failed! |
| 10:20.34 | sevard | yeah, I don't see how this #*$#ing thing is supposed to work. |
| 10:20.34 | Zeeek | which is why he's so bad tempered usually |
| 10:20.45 | sevard | I don't know the guy. |
| 10:21.04 | Zeeek | who began the obnoxious idea of using punctuation in error and informational messages, anyway |
| 10:21.13 | sevard | no idea, it's pretty annoying. |
| 10:21.22 | Zeeek | not Mark, the bugtracker |
| 10:21.43 | Zeeek | SomeFool is UNREACHABLE!!!!! |
| 10:21.50 | Zeeek | SomeFool is now REACHABLE!!!!! |
| 10:21.52 | sevard | seriously |
| 10:22.07 | sevard | the last thing I need is some stupid software yelling at me. |
| 10:22.50 | sevard | almost as annoying as "The Wiz" |
| 10:22.51 | Zeeek | it should be more like: "hmmmmm. SomeFool didn't answer my OPTIONS." |
| 10:23.30 | sevard | I don't know. I've always been equally irked about the VARIABLES being ALLINCAPS |
| 10:23.59 | Zeeek | how about obnoxious update reminders. "Why are you still running 1.2? Afraid to update? Get some cojones!" |
| 10:24.29 | sevard | mostly because I can't stop my brain from screaming things that are all in caps and in an hour's time working with dialplans I have a headache |
| 10:24.38 | sevard | What reminders? |
| 10:24.41 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 10:26.03 | Zeeek | the ones in my imagination. LUNCH |
| 10:26.03 | eeos | hi everybody |
| 10:26.46 | eeos | I am finding it hard to configure Asterisk so that our voip lines (provided from our provider, SIP protocol) feed into our local asterisk |
| 10:26.59 | eeos | I am using thebook, but still .... |
| 10:28.15 | eeos | Is there a case study we could use as reference, as far as you now? |
| 10:28.28 | walhala | does anyone know if hint works with skinny ? |
| 10:34.47 | *** join/#asterisk Snake-Eyes (n=blog@70.55.220.203.static.comindico.com.au) |
| 10:35.24 | eeos | also, is there a GUI based configurator for Asterisk? |
| 10:36.22 | punani | #freepbx |
| 10:36.56 | e-ddie | is there a feepbx too? |
| 10:38.10 | punani | google is thy friend :P |
| 10:38.13 | Siya | asterisknow |
| 10:38.19 | punani | freepbx / trixbox / etc |
| 10:38.50 | Mavvie | these cisco icons for networks are getting worse and worse. |
| 10:39.14 | Mavvie | when they started it was easy: this is a router, this is a switch and this is an framerelay device. |
| 10:39.42 | Mavvie | these days the orientation, number of arrows, shape of the arrows and if it is a circle or a square have all to be taken into account. |
| 10:42.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:42.36 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 10:43.57 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:44.55 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 10:45.04 | Siya | Mavvie: hehe, you don't have to use all of them |
| 10:46.32 | Mavvie | Siya: I know, but I can't find the ones I'm after neither :-) |
| 10:52.55 | Zeeek | eeos what are you attempting to accomplish? |
| 10:53.13 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 10:56.58 | Zeeek | lunch in the sun was very agreeable by the way |
| 10:57.23 | *** join/#asterisk Snake-Eyes (n=blog@70.55.220.203.static.comindico.com.au) |
| 10:58.06 | Siya | Mavvie: /msg... |
| 10:59.06 | eeos | Zeeek: sorry, sip phone ringing :D |
| 10:59.20 | eeos | Zeeek: we have some voip lines provided by a provider |
| 10:59.21 | Zeeek | what is you problem you want to solve? |
| 10:59.29 | Zeeek | SIP ? |
| 10:59.34 | eeos | Zeeek: we want to feed them into our local asterisk server we have just installed |
| 10:59.46 | Zeeek | what do you mean by "feed" |
| 10:59.49 | eeos | Zeeek: and use Asterisk server as a local pbx |
| 10:59.55 | eeos | Zeeek: yes, SIP |
| 11:00.04 | Zeeek | you want asterisk to register with your SIP provider? |
| 11:00.21 | eeos | Zeeek: instead of having every user using his / her own voip phone |
| 11:00.51 | Zeeek | so providerX calls asterisk but wants a specific extension? |
| 11:01.04 | eeos | Zeeek: we want the local sterisk server to distribute the calls, answer automatically when necessary (like "Zeeek is on a meeting, call back later") |
| 11:01.06 | eeos | and os on |
| 11:01.25 | Zeeek | and the dialplan isn't clear to you? |
| 11:01.28 | eeos | Zeeek: yes, it should register |
| 11:01.46 | eeos | Zeeek: not too much. :( |
| 11:01.53 | Zeeek | you have a peer entry for the provider in sip.conf and a register line |
| 11:02.04 | eeos | Zeeek: I am using this document called AsteriskTFOT |
| 11:02.11 | Zeeek | the call comes in and goes to a context (given in sip.conf peer entry) |
| 11:02.12 | eeos | Zeeek: that OReilly made |
| 11:02.20 | Zeeek | yes it's pretty clear |
| 11:02.35 | Zeeek | at what point do you stop understanding it? |
| 11:03.05 | eeos | Installtion went well, I am now at Chapter 4, ad using Appendix A and D |
| 11:04.10 | Zeeek | the call comes to "contextX" and you write extension in that context to hadle your users |
| 11:04.10 | Zeeek | handle the routing to your users phones |
| 11:04.36 | Zeeek | if your users have extensions like 2000-2100 you can use macros or wildcards to accompish this in a few lines |
| 11:04.58 | Zeeek | how many users/phones? |
| 11:05.22 | eeos | at the moment 2 |
| 11:05.29 | eeos | 2 softphones |
| 11:05.31 | Zeeek | not horribly complicated, then |
| 11:05.43 | eeos | no, just a good project to learn :D |
| 11:05.52 | Zeeek | and they can call each other now thru asterisk ? |
| 11:05.54 | eeos | people use headsets |
| 11:06.13 | *** join/#asterisk A[s]H (n=TnT@host117-192-static.53-88-b.business.telecomitalia.it) |
| 11:06.15 | eeos | Zeeek: not even tried, because they work in the same office so we did not need it |
| 11:06.15 | A[s]H | help me please |
| 11:06.21 | Zeeek | no law against headsets. Yet. |
| 11:06.30 | A[s]H | what file i mst download for g729 codec for intel celeron? |
| 11:06.32 | eeos | Zeeek: do you think I should start from there? |
| 11:06.41 | A[s]H | from digium ? |
| 11:06.49 | Zeeek | eeos no matter whether you need to or not, this is fun and learning, right? |
| 11:06.54 | eeos | Zeeek: yes |
| 11:07.03 | Zeeek | sure. Learn how to call one phone from the other |
| 11:07.11 | eeos | Zeeek: ok |
| 11:07.26 | Zeeek | <PROTECTED> |
| 11:07.28 | jbot | g729 is probably an ITU-standard voice codec which operates at 8kbps and offers quality very similar to GSM. G.729 is patent-encumbered; those wishing to use it with Asterisk must buy a license from Digium. |
| 11:07.32 | *** join/#asterisk VijayG (i=VijayG@202.131.145.247) |
| 11:07.35 | VijayG | Hello |
| 11:07.38 | Zeeek | not too helpful |
| 11:07.52 | eeos | Zeeek: why did you say g729? |
| 11:07.59 | Zeeek | eeos make up extensions thta each phone can be reached at |
| 11:08.01 | eeos | do we ned to use that codec? |
| 11:08.03 | VijayG | is there any way, i can hangup all calls going on asterisk server, without restarting the asterisk |
| 11:08.06 | A[s]H | from here http://ftp.digium.com/pub/telephony/codec_g729/asterisk-1.2/x86-32/ |
| 11:08.18 | Zeeek | no for the other person with the ridiculous pseaudo that can't be typed |
| 11:08.33 | eeos | Zeeek: so, I can assign an extesion on Asterisk that is not visible from outside, only from inside the organisation |
| 11:08.41 | Zeeek | eeos of course |
| 11:08.56 | Zeeek | Look at the chapter called "The Dialplan is the heart of asterisk" |
| 11:08.57 | A[s]H | please help me one second? |
| 11:09.11 | VijayG | any command, using which i can disconnect all the calls going on asterisk server |
| 11:09.27 | eeos | Zeeek: I meant this is the exercise you are telling me to perform. |
| 11:09.30 | Zeeek | for g729 if you bought a license there should be instructions |
| 11:09.34 | VijayG | something like soft hangup but all the calls should get disconnected at once |
| 11:09.35 | A[s]H | yes |
| 11:09.44 | Zeeek | eeos yes, the Zeeel lesson plan is thus |
| 11:09.46 | A[s]H | but i want to know for celeron |
| 11:09.52 | A[s]H | which file i must donwload |
| 11:09.55 | Zeeek | why not ask digium? |
| 11:10.11 | A[s]H | on digium concact center i have found this way |
| 11:10.13 | A[s]H | ask here |
| 11:10.14 | A[s]H | :) |
| 11:10.15 | Zeeek | or wait until more people are here in say 5 hours |
| 11:10.21 | A[s]H | haah |
| 11:10.30 | Zeeek | it's very early in the US |
| 11:10.39 | A[s]H | grrr |
| 11:10.46 | A[s]H | and you where are u from ? |
| 11:10.46 | eeos | Zeeek: thanks |
| 11:10.58 | VijayG | any command, using which i can disconnect all the calls going on asterisk server |
| 11:11.11 | Zeeek | VijayG stop the server? |
| 11:11.26 | Zeeek | like restart it |
| 11:11.37 | VijayG | is there any command in asterisk CLI |
| 11:11.43 | VijayG | which i can use to hangup all the calls |
| 11:11.50 | Zeeek | there is, see show applications |
| 11:12.04 | Zeeek | all the calls, not sure other than a restart |
| 11:12.15 | A[s]H | soft hangup |
| 11:12.44 | Zeeek | hangs up all the calls? |
| 11:12.48 | Zeeek | not so sure |
| 11:13.15 | A[s]H | uhm channel |
| 11:13.19 | A[s]H | but try |
| 11:13.37 | A[s]H | make a scritp for alla channel |
| 11:13.38 | A[s]H | :) |
| 11:14.23 | VijayG | that was good, but not a solution |
| 11:14.36 | VJFROMGT | is there a way to have a sip client which get authenticate by ip and not by userid / password? |
| 11:14.36 | A[s]H | grrr |
| 11:14.41 | Zeeek | restart gracefully |
| 11:14.51 | A[s]H | no |
| 11:15.12 | Zeeek | stop now |
| 11:15.16 | Zeeek | restart now |
| 11:15.17 | VijayG | actually what is happening is, my asterisk server is not getting proper disconnection signals from VOIP vendor |
| 11:15.26 | A[s]H | authentication = user + pass :) |
| 11:15.29 | Zeeek | contact vendor! |
| 11:15.51 | VijayG | so, after every one hour, i see that there is a log of about 400 calls on my CLI |
| 11:16.12 | Zeeek | bad vendor, bad |
| 11:16.13 | VijayG | so, restarting the server or stop now everytime will not work |
| 11:16.24 | VijayG | he is working on signalling part |
| 11:16.43 | Zeeek | stopping the server will kill all chanels |
| 11:16.48 | VijayG | but somehow, for the time being, i thought if i can find out a solution that if i can clear that log |
| 11:16.57 | VijayG | thats right, stopping will kill the channels |
| 11:17.11 | VijayG | but stopping and restarting every few minutes is not a good solution |
| 11:17.19 | Zeeek | not the best, no :) |
| 11:17.33 | Zeeek | getting a vendor that isn't broken is one |
| 11:18.01 | VijayG | for the time being, that cannot be done |
| 11:18.14 | VijayG | as we are taking a special service from him, which no other vendor gives |
| 11:18.25 | Zeeek | pr0n dialup? |
| 11:18.42 | VijayG | like we need to dial on UK special digit numbers, so for 90% of the providers its blocked |
| 11:18.52 | VijayG | very few vendors offer that service |
| 11:21.46 | VJFROMGT | i am looking for a way to authenticate based on ip alone |
| 11:26.34 | Siya | VijayG: you UK based? |
| 11:26.47 | Siya | Any advice on UK voip providers? |
| 11:27.28 | Siya | Any NL voipbuster users around? |
| 11:27.50 | *** join/#asterisk krdian_ (n=krdian@killer.radom.net) |
| 11:27.58 | krdian_ | hi |
| 11:28.52 | Zeeek | voiptalk |
| 11:28.57 | Zeeek | but a bit pricey |
| 11:29.43 | VijayG | Siya, i'm based in india |
| 11:29.54 | VijayG | and what into you want? |
| 11:32.35 | krdian_ | i have installed voicechanger in asterisk 1.2.13, but when i speaking through i can hear only noise. Any ideas ? |
| 11:33.22 | *** join/#asterisk jmacz (n=jmacz@201.244.170.3) |
| 11:35.51 | *** join/#asterisk berktr (n=canberk@teknopet.com) |
| 11:36.14 | berktr | how can i create monthly bills to my customers with asterisk2billing without using the ivr interface of a2billing |
| 11:36.42 | Zeeek | krdian_ what does that tell you about your voice? |
| 11:40.16 | krdian_ | Zeeek: do you know this software ? |
| 11:40.35 | Zeeek | I'm afraid not |
| 11:40.46 | krdian_ | :( |
| 11:40.55 | Zeeek | I know it works for some people |
| 11:41.11 | Zeeek | but why do you need it, to call pr0n vocal servers? |
| 11:41.24 | Zeeek | use a handkerchief |
| 11:42.14 | krdian_ | Zeeek: hehe, no i'm trying to develop funny application for mobile operator |
| 11:43.48 | krdian_ | Zeeek: i'm affraid this software is working only with asterisk 1.2.0 |
| 11:43.53 | VJFROMGT | <PROTECTED> |
| 11:45.40 | krdian_ | VJFROMGT: as i know, yes |
| 11:46.15 | VJFROMGT | krdian ,, please tell me how |
| 11:46.16 | Zeeek | krdian_ could be! I wonder what the percentages of adoption are between 1.2 and .4 |
| 11:48.41 | krdian_ | VJFROMGT: just put host=x.x.x.x withoust username/secret into configuration about your device |
| 11:50.46 | Siya | VijayG: UK voip providers for consumers are rare (or insanely priced) |
| 11:50.50 | VJFROMGT | and sip.conf? |
| 11:52.03 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 11:53.37 | VijayG | specially for premium numbers, right |
| 11:53.52 | krdian_ | VJFROMGT: yap |
| 11:54.43 | VJFROMGT | ? |
| 11:55.15 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 11:59.35 | VJFROMGT | krdian,, the reason i ask is because when i set host=myip my client does not register |
| 12:01.59 | *** join/#asterisk Corydon76-home (i=eight@pdpc/supporter/sustaining/Corydon76-home) |
| 12:01.59 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 12:02.29 | krdian_ | VJFROMGT: what a myip means ? |
| 12:02.38 | krdian_ | VJFROMGT: ip of your client ? |
| 12:05.19 | *** join/#asterisk skyhawker (n=info@a62-216-22-13.adsl.cistron.nl) |
| 12:05.42 | skyhawker | hey guys // got a problem with my DTMF .. I use inband all works except voicemail |
| 12:05.44 | skyhawker | any ideas / |
| 12:06.01 | Zeeek | ... |
| 12:06.39 | LeddyHM | Anyone use/recommend a pay as you go provider in/close to Houston |
| 12:07.02 | LeddyHM | looking for someone a little closer to us for some of our outbound calls |
| 12:07.22 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 12:07.34 | Zeeek | ... |
| 12:07.39 | VJFROMGT | kdrian yes,, the ip of my client |
| 12:08.26 | krdian_ | skyhawker: maybe try to change codec, what dabug says ? |
| 12:10.28 | skyhawker | krdian_ http://pastebin.ca/509329 thats the debug |
| 12:10.37 | skyhawker | i using alaw here in holland |
| 12:11.03 | Zeeek | can you smoke alaw? |
| 12:11.03 | berktr | i love asterisk |
| 12:11.05 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 12:11.27 | jmls | anyone know of meetme limitations ? How many meetme rooms etc ? |
| 12:11.39 | *** join/#asterisk Winkie (n=urmom@87-194-8-125.bethere.co.uk) |
| 12:13.19 | Siya | skyhawker: being in NL I assume you're aware of FSK/DTMF issues? |
| 12:13.34 | skyhawker | FSK/DTMF issues nope not yet |
| 12:14.02 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:15.03 | skyhawker | Siya: what FSK issues are those .. is that the reason why either my DTMF works on voicemail or it works on external calls |
| 12:15.06 | krdian_ | skyhawker: May 25 14:08:51 DEBUG[17570]: rtp.c:1361 ast_rtp_write: Ooh, format changed from unknown to alaw |
| 12:15.08 | skyhawker | but not both |
| 12:15.35 | Siya | NL phone system is FSK |
| 12:16.01 | Siya | asterisk = DTMF |
| 12:16.03 | skyhawker | krdian_ yeah dont get that there is no other codecs besides alaw |
| 12:16.48 | krdian_ | VJFROMGT: look at sip show peer <peer_name> |
| 12:17.05 | skyhawker | Siya: interesting .. is there a way to change asterisk to FSK then ? it might be easier then changing KPN to FSK LOL |
| 12:17.19 | Siya | skyhawker: so it's unrelated if your phones do DTMF and the vmail is on the asterisk server |
| 12:17.42 | Siya | KPN is FSK, changing them to DTMF would be the real challenge indeed |
| 12:18.04 | Siya | I only do voip on my *now box so I have no issues |
| 12:18.36 | Siya | afaik you can set it for your providers so asterisk will do FSK when connecting to the PSTN |
| 12:19.02 | Siya | though I'm not sure how it would relay DTMF to FSK |
| 12:19.28 | Siya | I'm no expert on this matter I just know that NL is FSK while most of the world is DTMF |
| 12:19.57 | skyhawker | Siya: great thanks .. gonna check out manuals google see what i can find |
| 12:21.38 | Siya | skyhawker: which providers do you use? |
| 12:22.13 | skyhawker | Siya: we have 3 providers KPN / Esprit telecom and xs4all (not usually used) |
| 12:22.29 | Siya | business use ic |
| 12:22.30 | skyhawker | Esprit are our SIP trunk provider |
| 12:22.33 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 12:22.45 | skyhawker | yeah .. it part business and part hobby stuff |
| 12:22.59 | Siya | ahh ic |
| 12:23.13 | skyhawker | using thomson 2030 phones .. trying to see if they support FSK now |
| 12:25.41 | *** join/#asterisk pejo_ (n=peter@192.36.127.64) |
| 12:25.54 | *** join/#asterisk Delvar (n=Delvar@217.14.137.34) |
| 12:26.23 | *** join/#asterisk TheCops (i=henri@206-248-146-28.dsl.teksavvy.com) |
| 12:27.48 | TheCops | Hi, I currently have a problem with polycom Soundpoint IP with firmware 2.x. When i'm dialing a number, i've got a delay sometime. it will hang 1 or 2 second before I can continue to dial. This is anoying problem, someone had this problem before ? I dont have this problem with 1.6.7 but I have more important problem with this release. |
| 12:28.36 | [TK]D-Fender | TheCops: So you have a forced "wait" in the middle of dialing? |
| 12:28.58 | TheCops | [TK]D-Fender in my dialplan you mean ? |
| 12:29.30 | TheCops | [TK]D-Fender, nop, this is doing this too when doing any task on the phone |
| 12:29.34 | TheCops | (menu and stuff like that) |
| 12:30.04 | TheCops | the scheduled log command "showcpuload" show me that the phone have 91% 70% of CPU average...I think this is not normal |
| 12:30.05 | TheCops | :) |
| 12:31.40 | [TK]D-Fender | TheCops: Oh, so just generally slow... |
| 12:32.05 | TheCops | [TK]D-Fender, I did a debug log on the polycom, I was dailing 1111111 and when the load come in, I press 555555 so I can see between this dialing what happenned |
| 12:32.29 | TheCops | [TK]D-Fender, ce n'est pas comprehensible les logs (sorry dont know how to say that in english) |
| 12:33.28 | TheCops | [TK]D-Fender, you have a field in logs file that show "how many event missed due to CPU load" and it show 6506 or 1200 or any high number... |
| 12:33.59 | TheCops | [TK]D-Fender I toke the original config file with only the SIP account modified, nothing more the problem is again there |
| 12:34.28 | [TK]D-Fender | TheCops: Jamais eu des problemes avec moi-meme. Mes copains ici ont 2.0.3.B sans trouble, et j'ai 2.1.1 au maison mais pas utilise |
| 12:35.15 | *** part/#asterisk Ravi1974 (n=I@ool-18b80982.dyn.optonline.net) |
| 12:35.17 | [TK]D-Fender | maybe too much "qualify" load? we know they are SLOW to respond so maybe the backlog is gtting to them... |
| 12:36.19 | TheCops | Ho! |
| 12:36.21 | TheCops | Qualify |
| 12:36.26 | TheCops | they are all on qualify |
| 12:36.30 | TheCops | Good idea dude |
| 12:37.37 | TheCops | [TK]D-Fender thank you let me try that :) |
| 12:39.19 | TheCops | Cpu load is 17.3, and the average is 46.9 |
| 12:39.23 | TheCops | ouch |
| 12:39.48 | TheCops | I'll test it a couple of day thank you a lot |
| 12:41.16 | *** join/#asterisk VJFROMGT (n=vijay_0@190.80.51.16) |
| 12:41.16 | [TK]D-Fender | np |
| 12:43.40 | *** join/#asterisk MindTheGap (n=iote@201.80.202.249) |
| 12:44.23 | sevard | So, can anyone tell me why I can get SAY NUMBER working in a python AGI but not STREAM FILE? |
| 12:46.15 | *** join/#asterisk DarylVOIP (n=daryl@c-71-224-42-97.hsd1.pa.comcast.net) |
| 12:47.35 | sevard | print "SAY NUMBER 192837465"," \"\"" gives me AGI Rx << SAY NUMBER 192837465 "" and print "STREAM FILE","tt-monkeys"," \"\"" gives me AGI Rx << STREAM FILE tt-monkeys "" yet only the former produces results. |
| 12:48.11 | sevard | is there something totally simple I'm missing here |
| 12:51.39 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 12:52.18 | FuriousGeorge | i dont suppose anyone in here is very familiar with troubleshooting a sangoma a200 are they. mine was working yesterday, is not today, and now we have replaced out asterisk pbx with a two line 2 station cordless phone |
| 12:56.28 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 12:59.51 | cpm | I think a good measure of the quality of a vendor is whether or not they have brag-vertising on their MOH for tech support |
| 13:00.24 | cpm | The *LAST* thing I want to hear when holding for tech support is happy girlies bragging about how great the service/product is |
| 13:00.47 | tzanger | cpm: heh |
| 13:01.03 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 13:01.16 | tzanger | FuriousGeorge: what changed between yesterday and today |
| 13:01.25 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 13:03.05 | [TK]D-Fender | tzanger: his entire PBX! ;) |
| 13:03.12 | cpm | heh |
| 13:03.49 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 13:03.49 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 13:04.14 | tzanger | [TK]D-Fender: :-) |
| 13:04.24 | [TK]D-Fender | pwned |
| 13:04.49 | sevard | [TK]D-Fender: long time no see. |
| 13:06.00 | [TK]D-Fender | sevard: well.... I've been here all along :) |
| 13:06.00 | sevard | heh |
| 13:06.00 | sevard | any ideas on my question? :) |
| 13:08.58 | [TK]D-Fender | sevard: Don't do real programming or AGI, so nope... |
| 13:11.12 | sevard | heh |
| 13:11.15 | sevard | "real programming" |
| 13:11.29 | sevard | my life just got a whole lot prettier. |
| 13:14.39 | tzanger | sevard: ? |
| 13:15.43 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:15.52 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 13:16.38 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 13:16.58 | sevard | tzanger: ? |
| 13:17.17 | tzanger | your life just got prettier |
| 13:17.31 | tzanger | who moved in or walked by? |
| 13:18.22 | *** join/#asterisk jtexter3 (n=jtexter3@69-94-197-4.biltmorecomm.com) |
| 13:18.35 | sevard | you, baby! |
| 13:18.41 | sevard | any insight to my question? :) |
| 13:18.53 | [TK]D-Fender | sevard: Let me put it this way. I was a DOS God in my day. I wrote telecom software, file managers, and all sorts of stuff. Then Windows came and crashed my party and my skills have atrophied. I do a bit of PHP these days, and thats it (as far as acknowledged languages go). |
| 13:19.15 | tzanger | I'm getting nothing but 100% quality levels in my unlimitel reports now.. woo. :-) |
| 13:19.18 | tzanger | sevard: eep |
| 13:19.21 | tzanger | sevard: what question |
| 13:19.31 | sevard | [TK]D-Fender: DOS god FTW |
| 13:19.37 | tzanger | I used to be a dos god too |
| 13:19.44 | sevard | tzanger: it's a couple lines up, i'll repost |
| 13:19.47 | tzanger | i386 assembly was like a second language |
| 13:19.57 | tzanger | reverse engineering, soft-ice was my tool of choice |
| 13:20.04 | sevard | So, can anyone tell me why I can get SAY NUMBER working in a python AGI but not STREAM FILE? |
| 13:20.06 | tzanger | but yeah windows came and I just lost interest |
| 13:20.07 | sevard | print "SAY NUMBER 192837465"," \"\"" gives me AGI Rx << SAY NUMBER 192837465 "" and print "STREAM FILE","tt-monkeys"," \"\"" gives me AGI Rx << STREAM FILE tt-monkeys "" yet only the former produces results. |
| 13:20.19 | tzanger | now I develop embedded shit with linux |
| 13:21.18 | sevard | I'm trying to stay away from "official AGI python libraries" since they are all depreciated and have broken classes, kind of worthless. |
| 13:21.32 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:22.09 | tzanger | sevard: hmm, that is beyond me |
| 13:22.13 | tzanger | I too stay the hell away from agi |
| 13:22.35 | sevard | crud |
| 13:22.37 | [TK]D-Fender | I need to start up with PHP-AGI |
| 13:23.51 | mocker | sevard: Could it need the extension? |
| 13:24.14 | mocker | I know usually asterisk autodetects that.. |
| 13:24.20 | *** join/#asterisk Gringo_ (n=N3TW4LK3@37.200-245-81.adsl-dyn.isp.belgacom.be) |
| 13:25.06 | Gringo_ | hey |
| 13:25.29 | Gringo_ | is it normal that it takes zaptel a few seconds to detect an incoming ring (on PSTN) |
| 13:25.37 | Gringo_ | regular trunk |
| 13:26.07 | Gringo_ | 'cause it happens here every so often than someone dials a call at the exact moment an incoming PSTN rings |
| 13:26.16 | *** join/#asterisk jtexter3 (n=jtexter3@69-94-197-4.biltmorecomm.com) |
| 13:26.23 | sevard | mocker: ? |
| 13:26.56 | Gringo_ | asterisk then picks up the line before detecting the ring, and connects the incoming call to the outgoing |
| 13:27.03 | Gringo_ | causing confusion :) |
| 13:27.13 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 13:27.30 | Gringo_ | is it normal for zaptel to take a second or 2 to detect an incoming call? |
| 13:27.46 | [TK]D-Fender | Gringo_: Happens all the time and it DOES have to actually ring for a 1 sec or so before * can confirm its a ring. |
| 13:28.02 | [TK]D-Fender | Gringo_: So you'll bel dialing out on a freshly answered call... |
| 13:28.21 | Gringo_ | [TK]D-Fender: nice :( so you're saying you have this too, and there not much you can do about it? |
| 13:28.32 | [TK]D-Fender | Gringo_: Welcome to the wonrderful world of analog. This happened to me several times on a regular phone at home ages back. |
| 13:28.48 | mocker | sevard: Can you get agi-test.agi working? |
| 13:28.52 | [TK]D-Fender | Gringo_: thats the downside of the entire tech. |
| 13:28.57 | mocker | sevard: It has a STREAM FILE part in it |
| 13:29.00 | [TK]D-Fender | Gringo_: Live with it or go digital. |
| 13:29.42 | *** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:29.44 | Gringo_ | okay then, thanks for the info, [TK]D-Fender |
| 13:29.54 | [TK]D-Fender | Gringo_: np |
| 13:30.43 | sevard | mocker: yes, but mine spits out the exact same thing and doesn't work. |
| 13:32.19 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
| 13:32.56 | mocker | sevard: Hmm, that's strange. |
| 13:33.02 | mocker | sevard: Do you grab a result at the end?' |
| 13:35.18 | *** join/#asterisk myiagy (i=myiagy@201.31.20.47) |
| 13:35.29 | mocker | sevard: http://pastebin.ca/509433 |
| 13:35.32 | sevard | there's some god damned weirdness goin on here |
| 13:35.50 | mocker | I had to do that last readline to get anything to go.. |
| 13:36.11 | mocker | (obviously this is w/ SAY DIGITS, but it may apply?) |
| 13:36.44 | sevard | try it with STREAM FILE, i had tried something similar with raw_input() |
| 13:37.10 | [TK]D-Fender | Qwell : that sounds strangely perverse.... |
| 13:37.10 | sevard | mmmmmsexy qwell |
| 13:37.56 | sevard | mocker: no go. |
| 13:39.18 | *** join/#asterisk shinao1 (n=shinao1@dial-pool1.lagos.starcomms.net) |
| 13:41.03 | mocker | sevard: Hmm, dunno then. :( |
| 13:41.05 | *** join/#asterisk bbryant (n=brett@69-94-196-94.biltmorecomm.com) |
| 13:41.19 | mocker | sevard: Have you tried connecting directly to the asterisk session and using agi debug? |
| 13:42.21 | sevard | that's what i'm using |
| 13:43.33 | mocker | sevard: But the agi-test works? |
| 13:43.39 | mocker | for the stream file part.. |
| 13:44.28 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
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| 13:49.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:50.29 | *** join/#asterisk jcaceres (n=jcaceres@190.41.82.1) |
| 13:52.07 | jcaceres | hello i have a problem for serting a sib outbound route using grandstream GXW4104 with the last firmware |
| 13:52.26 | jcaceres | it just don send the calls |
| 13:53.04 | *** join/#asterisk MrChicken (n=Dorphals@200.71.58.39) |
| 13:53.06 | MrChicken | Hello |
| 13:53.14 | MrChicken | I was reading the wiki page on queues.conf |
| 13:53.21 | jcaceres | i made a trunk with a trixbox but it dials grandstream, but the gateway does nothing |
| 13:53.25 | MrChicken | and I saw the weight option |
| 13:53.39 | MrChicken | and also that it generates a deadlock |
| 13:53.44 | jcaceres | any idea? |
| 13:53.53 | MrChicken | I was kinda wondering if latest asterisk version is patched |
| 13:54.28 | MrChicken | (also read the bugtracker which states that another solution must be found) |
| 13:54.53 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:54.58 | Katty | morning :< |
| 13:55.09 | *** join/#asterisk putnopvut (n=putnopvu@69-94-197-46.biltmorecomm.com) |
| 13:55.11 | MrChicken | so my Q is... will this genearate a deadlock in a high call volume environment? |
| 13:55.12 | Katty | [TK]D-Fender: i have serious problem. |
| 13:55.18 | Katty | [TK]D-Fender: and it's my boss. |
| 13:55.19 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:55.26 | Katty | anonymouz666: (= |
| 13:55.41 | MrChicken | Katty ... why dont you 'Take care of him' ? |
| 13:55.43 | MrChicken | :P |
| 13:55.46 | anonymouz666 | Katty!!! |
| 13:55.50 | *** join/#asterisk HYPN0TEK (n=Hypn0tek@41.226.241.50) |
| 13:56.14 | Katty | MrChicken: not that easy... |
| 13:56.23 | Katty | MrChicken: multimillionares don't go down easily. |
| 13:56.35 | Katty | we're renting out the upstairs, and provided phone lines, internets etc. |
| 13:56.44 | Katty | but they want to bill the upsetairs for the longdistance |
| 13:56.50 | Katty | and do so by giving them a pin number |
| 13:56.52 | *** join/#asterisk heliosj (n=jeff@pdpc/supporter/active/xheliox) |
| 13:57.02 | *** part/#asterisk heliosj (n=jeff@pdpc/supporter/active/xheliox) |
| 13:57.04 | Katty | i don't know how on earth i'm going to make the upstairs context long distance take a pin number |
| 13:57.11 | Katty | much less what to do with it once i've got it |
| 13:57.18 | Katty | so i need advice |
| 13:57.22 | tzanger | Katty: you can do it |
| 13:57.24 | tzanger | just break it down |
| 13:57.26 | Katty | if you have clients that you're trying to bill for long distance |
| 13:57.29 | Qwell | they dont need a pin...just use CDR |
| 13:57.36 | tzanger | they dial a LD number, you want to present them with a "enter the damn code" prompt |
| 13:57.45 | tzanger | I do agree with qwell though |
| 13:57.45 | Katty | i can do that tzanger |
| 13:57.51 | Katty | there's even a vm prompt wave for that |
| 13:57.53 | tzanger | you get the biling infos |
| 13:57.56 | tzanger | Katty: yep |
| 13:58.07 | Katty | they're going to be moving offices tho |
| 13:58.19 | Katty | so i can't just charge via the extension they're on |
| 13:58.30 | Katty | pin numbers do seem logical |
| 13:58.34 | Katty | but once i get the pin number... |
| 13:58.46 | Katty | i presume i need some sort of this is the number you dialed, this is how long you were on the call |
| 13:58.49 | MrChicken | Katty... it all depends on how good looking you are... if u're reaaally hot, then bringing a multimillionare down is a piece of cake :P |
| 13:58.51 | tzanger | just assign all their sip accounts the same accountcode |
| 13:59.16 | Katty | tzanger: i don't understand what you're getting at. |
| 13:59.18 | tzanger | Katty: cdr already provides all of that |
| 13:59.21 | MrChicken | kaldemar ... CDR |
| 13:59.21 | Katty | tzanger: all the phones are logged in. |
| 13:59.28 | Katty | tzanger: i can't change their login information |
| 13:59.30 | tzanger | Katty: how are they connected, SIP phones, another asterisk box, what |
| 13:59.34 | Katty | tzanger: nor would i want to... |
| 13:59.39 | Katty | tzanger: they're using our upstairs phones |
| 13:59.40 | tzanger | no you don't have to |
| 13:59.43 | Katty | tzanger: random ones. |
| 13:59.49 | Katty | tzanger: but they're all in the Upstairs context |
| 13:59.49 | tzanger | upstairs phones = what, Zap channels, SIP phones, what |
| 13:59.57 | Katty | zap chanenls, sip phones |
| 14:00.04 | tzanger | Katty: shit, if they're all in the upstairs context |
| 14:00.04 | Katty | physically one on server |
| 14:00.07 | tzanger | just start them out there |
| 14:00.10 | tzanger | [upstairs] |
| 14:00.13 | Katty | yeah |
| 14:00.15 | *** join/#asterisk tmcpr (n=tmcpr@85-189-92-116.btlnet.managedbroadband.co.uk) |
| 14:00.17 | Katty | but what do i do after i have the pin number |
| 14:00.24 | Katty | can i make a new column in the cdr? |
| 14:00.24 | tzanger | exten => s,1,Set(ACCOUNTCODE=them) |
| 14:00.33 | Katty | no |
| 14:00.34 | tzanger | exten => _NXXNXXXXXX,1,Set(ACCOUNTCODE=them) |
| 14:00.34 | tzanger | etc |
| 14:00.34 | *** part/#asterisk tmcpr (n=tmcpr@85-189-92-116.btlnet.managedbroadband.co.uk) |
| 14:00.40 | Katty | there will be different pin numbers |
| 14:00.41 | tzanger | you do not need a new column |
| 14:00.42 | Katty | they're not all the same |
| 14:00.47 | tzanger | accountcode is already there |
| 14:00.48 | *** join/#asterisk Dovid (n=Dovid@bzq-88-153-15-22.red.bezeqint.net) |
| 14:00.49 | Katty | i must not understand. |
| 14:00.52 | Katty | what is accountcode? |
| 14:00.57 | tzanger | just a variable in the cdr |
| 14:01.02 | tzanger | it's already there and generally unused |
| 14:01.03 | Katty | a predefined field specifically for pin numbers? |
| 14:01.05 | tzanger | so use it :-) |
| 14:01.06 | tzanger | no |
| 14:01.10 | MrChicken | accountcode is the account of the phones |
| 14:01.16 | MrChicken | so you know who's calling in the CDR |
| 14:01.22 | MrChicken | and the PINS... |
| 14:01.23 | MrChicken | welll |
| 14:01.28 | MrChicken | you can do it thry MySQL |
| 14:01.35 | Dovid | is relaxeddtmf=X only used for zaptel ? or will it help with SIP as well ? I am having an issue with a2billing where when a caller enters a number to call some times the system receives double digits |
| 14:01.35 | Katty | but the only way we'll know is by their pin number |
| 14:01.40 | Katty | we have no way of knowing which phone they'll be using |
| 14:01.43 | tzanger | Katty: the upstairs company is treated as one entitty right, you're not interested in billing each individual phone differently are you? |
| 14:01.49 | Katty | tzanger: no |
| 14:01.50 | tzanger | who cares what phone they're using |
| 14:01.56 | Katty | tzanger: they are offices rented out to multiple businesses |
| 14:01.59 | tzanger | you said they ALLLLLLLL end up (ro start out rather) in the upstairs context |
| 14:02.04 | tzanger | aha |
| 14:02.09 | Katty | tzanger: yes, they are all in the upstairs context |
| 14:02.10 | tzanger | starting point 1 |
| 14:02.15 | Katty | tzanger: and based on that context, it will ask for a pin number |
| 14:02.18 | tzanger | you have numerous clients upstairs |
| 14:02.23 | Katty | tzanger: but once they input that pin number, i don't know how to get it into the cdr |
| 14:02.28 | *** join/#asterisk ThOr101 (n=bthorson@pool-71-126-163-76.washdc.fios.verizon.net) |
| 14:02.30 | [TK]D-Fender | Katty: Mew. |
| 14:02.35 | Katty | [TK]D-Fender: mew. kill my boss. |
| 14:02.51 | Katty | tzanger: let's take this to private |
| 14:04.48 | TheCops | [TK]D-Fender havent solved the problem but still help I guess...but happen again, weird. I sent an email to polycom with debug logs and stuff like that should get an answer |
| 14:05.00 | ThOr101 | The book doesn't mention distinctive ring. I've tried google. A suggestion for documentation on "Routing incoming analog calls based on distinctive ring"? |
| 14:05.11 | *** join/#asterisk tmcpr (n=tmcpr@85-189-92-116.btlnet.managedbroadband.co.uk) |
| 14:05.25 | *** join/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
| 14:05.56 | [TK]D-Fender | Katty: What is it now? |
| 14:08.34 | *** join/#asterisk tzafrir_laptop (i=tzafrir@69-94-197-125.biltmorecomm.com) |
| 14:08.49 | Katty | [TK]D-Fender: 100 offices up stairs. |
| 14:08.56 | Katty | [TK]D-Fender: someone goes upstairs and picks an office. |
| 14:09.00 | Katty | [TK]D-Fender: they pick a new one tomorrow |
| 14:09.09 | ThOr101 | Hmm, if * isn't even detecting the distinctive ring, and only detecting a ring, it could be my signalling? |
| 14:09.13 | Katty | [TK]D-Fender: they want to assign a pin number, to a company, so they can bill them their long distance |
| 14:11.03 | [TK]D-Fender | Katty: Been done.... not terribly difficult |
| 14:11.19 | Katty | [TK]D-Fender: tzanger's holdin my hand right now |
| 14:11.20 | berktr | i have a question |
| 14:11.24 | Katty | [TK]D-Fender: i'm gonna get through it ^_^ |
| 14:11.33 | berktr | let's say i have a fxo gateway that sends all the pstn calls to 1002 extension |
| 14:11.59 | berktr | and i want my users to be able to pick up call from 1002 when the 1002 rings |
| 14:12.04 | berktr | is it possible |
| 14:12.32 | [TK]D-Fender | berktr: Yes. Go lookup "pickupgroup" on the WIKI |
| 14:12.37 | *** join/#asterisk Cresl1n (n=matt@69-94-196-9.biltmorecomm.com) |
| 14:12.37 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:12.50 | berktr | thanks |
| 14:13.20 | *** join/#asterisk y7n (n=na@81-179-112-35.dsl.pipex.com) |
| 14:13.26 | [TK]D-Fender | ThOr101: http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels |
| 14:14.10 | TheCops | [TK]D-Fender have you tried the SLA support of * ? |
| 14:15.07 | ThOr101 | Thanks TK, I've been parked on that page for a while. "Asterisk will apparently report the three numbers it sees as representing the ring tone it heard, and you can use these numbers in a dring" isn't true. I just get the standart "Answer / Ring" messages in the console / CLI. |
| 14:15.26 | y7n | I want to receive a variable from a user via the keypad for example a password or account number. Is the ${EXTEN} variable used for this purpose or should I be using a specific function? |
| 14:15.34 | [TK]D-Fender | TheCops: It isn't "real" and applies at best to those with IP 501's or higher with multiple ANALOG lines |
| 14:15.39 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 14:16.06 | TheCops | [TK]D-Fender this is working well ?! |
| 14:16.23 | [TK]D-Fender | <PROTECTED> |
| 14:16.24 | [TK]D-Fender | <PROTECTED> |
| 14:16.25 | ThOr101 | What is interesting is that I set usedistinctiveringdetection=yes in zapata.conf, but I don't see it in the debug output, which I would have thought I would have. |
| 14:16.26 | [TK]D-Fender | <PROTECTED> |
| 14:16.42 | ThOr101 | Yeah, I have all that in there. |
| 14:16.59 | [TK]D-Fender | TheCops: It "works" *sorta*, but its SO not real and only useful for those with very few analog lines as to be useless to me. |
| 14:17.11 | ThOr101 | I tried all the different values for dring1, 2 and 3, all while stopping and restarting * |
| 14:17.51 | [TK]D-Fender | ThOr101: maybe your telco is using a funky pattern... |
| 14:18.21 | pigpen | I am having issues with a perl script that telnets into the asterisk manger. It ran great then unexpectedly stopped working. |
| 14:18.22 | Nugget | telnet is eeeeeeevil! |
| 14:18.26 | pigpen | http://pastebin.ca/509498 |
| 14:18.29 | ThOr101 | That's my theory. I was hoping the documentation would be correct that it would display on the CLI, but it doesn't. I have a growing suspicion that distinctive ring isn't fully "engaged" |
| 14:18.55 | pigpen | Thanks to [TK], manually the output works, auto answering at the polycom extension. |
| 14:19.09 | pigpen | now, it just opens the manager and closes it....no action. |
| 14:20.09 | pigpen | but the script does successfully logon and off. |
| 14:22.06 | berktr | ok guys. i set the features.conf to pickup the calls with *3 hotkey however when i call somebody and try to pick up the call from another phone, i get the error 'nothing to pickup' why? |
| 14:22.17 | [TK]D-Fender | $exten = $ARGV[0]; <- is ARGV supposed to be in caps? |
| 14:22.54 | pigpen | if I hardcode the variable to "200", it acts the same. |
| 14:23.00 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:23.17 | pigpen | I will replace it again..just to be sure. |
| 14:23.29 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 14:23.48 | *** join/#asterisk myiagy (i=myiagy@201.31.20.47) |
| 14:24.14 | pigpen | yeah..no dice. |
| 14:24.51 | pigpen | I have a output in there that dumps out the exten, which is reporting the correct extension. |
| 14:25.07 | pigpen | And I see: |
| 14:25.08 | pigpen | [May 25 09:11:42] VERBOSE[628] logger.c: == Manager 'admin' logged on from 127.0.0.1 |
| 14:25.08 | pigpen | [May 25 09:11:42] VERBOSE[628] logger.c: == Manager 'admin' logged off from 127.0.0.1 |
| 14:25.12 | [TK]D-Fender | pigpen: one too many "\n"'s after your secret |
| 14:25.33 | [TK]D-Fender | pigpen: and |
| 14:25.35 | [TK]D-Fender | $telnet->print("Events: off\n"); |
| 14:25.36 | [TK]D-Fender | $telnet->print("Application: Playback\n"); |
| 14:25.38 | [TK]D-Fender | $telnet->print("Data: tt-monkeysintro\n"); |
| 14:25.39 | [TK]D-Fender | is useless, remove |
| 14:25.53 | *** join/#asterisk iulius_ (n=iulius@mail1.technologieshq.com) |
| 14:26.18 | pigpen | how is it to know what to playback? |
| 14:26.26 | [TK]D-Fender | pigpen: I use "originate" as all lower case... not sure if that matters. |
| 14:26.34 | pigpen | ie: tt-monkeysintro was a test audio file. |
| 14:26.35 | *** join/#asterisk Fieldy (i=4VYtH2w6@gentoo/contributor/Fieldy) |
| 14:27.02 | berktr | i figured it out thanks anyway |
| 14:27.02 | y7n | I wish to store a store a number entered on a users phone then store this in a temporary variable. What function would i use? |
| 14:27.03 | [TK]D-Fender | pigpen: You do your page with the local channel, and the context,exte,prio you DUMP THEM INTO is supposed to do the playback. |
| 14:27.18 | [TK]D-Fender | y7n: "show application read" |
| 14:27.26 | y7n | thanks |
| 14:27.31 | pigpen | oh...so like a meetme room. |
| 14:27.55 | [TK]D-Fender | pigpen: No. $telnet->print("Channel: Local/$exten*\@from-sip\n"); |
| 14:27.58 | *** join/#asterisk Arsenick-TC2L (n=tc2l@modemcable026.33-70-69.static.videotron.ca) |
| 14:28.04 | [TK]D-Fender | pigpen: That is the person you are calling. |
| 14:28.36 | [TK]D-Fender | pigpen: When they answer (forcibly or willing depending on your dialplan), they get dumped where you told it to go. in THERE you Answer, playback, hangup |
| 14:29.10 | [TK]D-Fender | pigpen: You don't try to issue this from the manager, you let the dialplan do its work. |
| 14:29.18 | pigpen | ah..so you don't have the system do the playback....but a section of the dialplan. |
| 14:29.22 | pigpen | k..got it. |
| 14:29.25 | [TK]D-Fender | pigpen: Correct |
| 14:29.32 | pigpen | makes sense. |
| 14:30.27 | pigpen | So in short, have the "phone" call the exten in the dialplan that does the playback that I wish. |
| 14:30.54 | [TK]D-Fender | pigpen: Yup, you're just "pushing" the "caller" into the process |
| 14:31.15 | pigpen | k.. |
| 14:31.26 | pigpen | [TK]D-Fender, thanks yet again. |
| 14:34.05 | [TK]D-Fender | pigpen: Np, let me know how it turns out |
| 14:37.11 | berktr | i have four different pstn lines, how can i make asterisk to choose a random one when calling out? |
| 14:37.18 | berktr | i have 4 different sip accounts for them |
| 14:37.26 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 14:37.38 | TheCops | berktr make a group of these line and call trought a group (g0) or something like that |
| 14:38.18 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:38.22 | eeos | do I always have to have a zap channel configured or are there situations where it is not necessary? |
| 14:38.26 | berktr | can you explain it a little bit |
| 14:38.59 | eeos | We do not have any phisical card or phone, only sip sofphone and even our lies travel over the broadband (SIP) |
| 14:39.36 | eeos | sorry, We do not have any physical card or phone, only sip softphone and even our lines travel over the broadband (SIP) |
| 14:39.39 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 14:39.43 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:39.47 | eeos | so I thought we did not need a zap channel |
| 14:41.14 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 14:42.22 | Arsenick-TC2L | Hi guy's, I would like to make a callbacl system. When I call from my cell to the asterisk, he hangup and call me back on my cell, then I would like to be able to dial out in the PSTN someone know how can I do that ? I've already done the callback part but I'm completely missed with the dialout... |
| 14:42.56 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:44.40 | tmcpr | Hi - Is there a way to know which gzipped release relates to which subversion version? |
| 14:45.04 | [TK]D-Fender | eeos: you do not HAVE zap channels. |
| 14:45.12 | Ryushin | Well, I'm setting up another server for a new customer. They are going to be using polycom phones and one of their requirements is that polycom phones must pull their configuration using an encrypted method. So no tftp or ftp. That leaves me implicit FTPS and HTTPS. |
| 14:45.48 | pigpen | [TK]D-Fender, Ok. This is what I have: http://pastebin.ca/509528 |
| 14:45.55 | eeos | [TK]D-Fender: what do you mean? |
| 14:45.57 | berktr | is it possible to block some extensions to make calls to pstn? |
| 14:46.10 | pigpen | I run the script and it doesn't work. |
| 14:46.12 | [TK]D-Fender | eeos: If you don't have any TDM cards they you have no zaptel channels |
| 14:46.19 | Ryushin | I'm having a heck of a time finding a linux based ftp server that does implicit ftps. Does the HTTPS portion support webdav so the phone can push up it's logs using a username and password? |
| 14:46.29 | pigpen | however, if I past the output from /tmp/output.log, it works fine. |
| 14:46.48 | [TK]D-Fender | eeos: Zaptel = driver for PCI interfaces (and ZTDUMMY for timing, but thats not what we're talking about here) |
| 14:46.49 | berktr | is it possible to block some extensions to make calls to pstn? |
| 14:47.18 | pigpen | berktr, yes..it is all about your dialplan. |
| 14:47.37 | [TK]D-Fender | berktr: Yes, its your dialplan. Don't let the devices you want to block have access to extens going where you don't want them to go. |
| 14:47.52 | [TK]D-Fender | pigpen: does it WORK? |
| 14:47.55 | eeos | [TK]D-Fender: yes, that is what I meant! because I have no hardware of this type I do not need to set up anything. Is that right? |
| 14:47.56 | pigpen | manually. |
| 14:48.05 | eeos | I only needed to cofirm, I am an absolute newbie |
| 14:48.13 | berktr | [TK]D-Fender, can you explain a little bit how can I do it? |
| 14:48.16 | eeos | going though thebook right now |
| 14:48.38 | pigpen | it almost seems like net::telnet is not sending everything....because the output file is correct. |
| 14:48.42 | [TK]D-Fender | berktr: make a new context. Point the phone at it. DON'T inlucde the extens that use your PSTN access. The end. |
| 14:48.48 | eeos | [TK]D-Fender: so I am a bit concerned about jumping important stages or not understanding :) (chapter 4 now) |
| 14:48.53 | Dovid | is there any way I can change the H part of the dial command ? |
| 14:49.06 | Dovid | I want the user to press ** and not just * to end a call |
| 14:49.09 | pigpen | berktr, yeah..get to know dialplan logic and all about contexts |
| 14:49.09 | [TK]D-Fender | eeos: Move along... nothing to see here :0 |
| 14:49.27 | berktr | [TK]D-Fender, let's say i've done that, will the other context users be able to dial those numbers? |
| 14:49.33 | berktr | who are blocked |
| 14:49.36 | *** join/#asterisk DaveCanoe (n=Dave@ool-45789009.dyn.optonline.net) |
| 14:49.50 | eeos | [TK]D-Fender: thanks for your help! |
| 14:49.56 | [TK]D-Fender | berktr: A context does what you tell it to. A phone has access to the context you tell it to use. Nothing more to say |
| 14:50.58 | pigpen | berktr, think of a context like a fat kid. Unless he is included, he will not play with anyone. |
| 14:51.18 | berktr | pigpen, you confused me lol :) |
| 14:51.32 | Arsenick-TC2L | If someone had already did this kind of thing let me know.. here's the callback part http://pastebin.ca/509533 |
| 14:52.04 | [TK]D-Fender | Arsenick-TC2L: that SO doesn't work. |
| 14:52.05 | *** join/#asterisk vAd0r (n=IceChat7@216-201-139-51.res.logixcom.net) |
| 14:52.16 | eeos | [TK]D-Fender: do you know of a tutorial for setting asterisk up so that people can use it on the internal network? |
| 14:52.32 | [TK]D-Fender | Arsenick-TC2L: Go read up on ".call" files and AMI Originate to have * generate outbound calls for "callback" |
| 14:52.47 | berktr | pigpen, can you help me a lil bit |
| 14:53.05 | [TK]D-Fender | eeos: That is a fabulously vague statement. basically says "I know nothing about Asterisk, how do I set it up?". For that... |
| 14:53.07 | [TK]D-Fender | ~book |
| 14:53.19 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:53.19 | Ryushin | Is anyone provisioning their polycom phones in a encrypted method? |
| 14:53.20 | pigpen | yeah..give me a few... |
| 14:53.20 | pigpen | [TK]D-Fender, I will bug you later with my woes. |
| 14:53.30 | [TK]D-Fender | Ryushin: FTPS should be easy enough. |
| 14:53.37 | Arsenick-TC2L | You mean I'll not be able call back and dialout with this kind of callback ? cuz it's working now If I reach the ext. 206 he hangups and call me back with another zap |
| 14:54.13 | Ryushin | Well, it's not true FTPS. It's implicit ftps. Explicit ftp is the RFC and the polycom phones don't support explicit yet. |
| 14:54.32 | Ryushin | Yea, yea, I thought it would be easy, but it's turning out to be a pain. |
| 14:54.43 | [TK]D-Fender | Arsenick-TC2L: that is not a callback (the way you've writtien it". You are calling YOURSELF from the same # you call in from but would be getting a call-waiting beep and you do not get a 2nd dialtone or anything else. |
| 14:54.51 | [TK]D-Fender | Arsenick-TC2L: Try and see. |
| 14:54.51 | vAd0r | how do i reset some ports of people connected? |
| 14:54.55 | Ryushin | Do you know if the phones will push their logs and such through https using webdav? |
| 14:55.01 | eeos | [TK]D-Fender: not really. Anynway. |
| 14:55.02 | [TK]D-Fender | Arsenick-TC2L: and then read up on the topics I provided to you. |
| 14:55.14 | Arsenick-TC2L | ok thx |
| 14:55.56 | [TK]D-Fender | Ryushin: No clue, not sure about how it would deposit them.... never did any excryption with mine. |
| 14:56.17 | [TK]D-Fender | Ryushin: I presume it'd use a standard means. |
| 14:57.53 | *** join/#asterisk dimas (n=ds@81.18.135.125) |
| 14:57.59 | vAd0r | how can i kill a sip client that is connect through asterisk -r |
| 14:58.18 | [TK]D-Fender | vAd0r: SIP does not connect to that * CLI. |
| 14:58.34 | Mercestes | Ryushin, As I recall b11d managed to get the encryptor program from Polycom, maybe he wil share it. |
| 15:00.21 | Ryushin | Thanks [TK]D-Fender. Time to set up webdave and https and see if it works. |
| 15:00.21 | Mercestes | Ryushin, Polycom's stance on it is "Oh shit, that wasn't even supposed to be in the manual" in remarkable Polycom fashion. |
| 15:00.21 | Ryushin | Mercestes: That's classic. |
| 15:00.21 | Mercestes | Or you could try to bug Polycom for it. I might still have Tim Sisneros # around somewhere. |
| 15:00.26 | vAd0r | can i only do one connection through a cisco ata at a remote site or can they get both lines |
| 15:01.06 | Ryushin | From what I've read, that keeps the files on the phone encrypted. But that does not set up encryption for provisioning. Or does that program encrypt communications between that asterisk server and the phone. That would be cool. |
| 15:01.35 | [TK]D-Fender | vAd0r: Are you asking if you can use both ports on an ATA 186 as seperate SIP devices with *? |
| 15:02.15 | [TK]D-Fender | Ryushin: Encryped in the PHONE? Whats the point? Afraid somone is going to steal the EEPROM's and hack them? |
| 15:02.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 15:02.22 | vAd0r | yes |
| 15:02.32 | Ryushin | yea, that was my feeling when I read it. |
| 15:02.35 | vAd0r | i had the adapter at my house and both ports worked |
| 15:02.58 | vAd0r | i brought it to a test internet connection and change the ata to goto my public ip and only one will connect |
| 15:04.15 | *** join/#asterisk oej (n=olle@70.158.103.10) |
| 15:04.41 | Ryushin | Okay, I'm wrong. I went and looked it up again. You encrypt the file on the server, and provide the same key to the phone to suck them down over whatever, then the phone will decrypt them. |
| 15:05.02 | Ryushin | Though, that would be a serious pain to have to decrypt every file on the server to edit it. |
| 15:05.16 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 15:06.01 | [TK]D-Fender | vAd0r: Well clearly it can work, and does work for others. |
| 15:07.09 | berktr | do you guys know a cheap az termination company? |
| 15:08.06 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:08.56 | berktr | A-Z termination by the way |
| 15:09.18 | *** join/#asterisk codefreeze (n=steve_mu@69-94-197-138.biltmorecomm.com) |
| 15:09.29 | *** join/#asterisk BouYYY (n=BouYYY@81-86-77-70.dsl.pipex.com) |
| 15:12.53 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 15:13.31 | *** join/#asterisk festr__ (n=festr@212.71.169.34) |
| 15:13.31 | SwK | getting there |
| 15:14.09 | festr__ | hello, anyone here tested 1.4latest svn? i'm expiriencing sip deadlocks. i've some backtraces. any help? |
| 15:14.12 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 15:14.53 | BouYYY | I have been looking at 1.4, when did the deadlocks occur? |
| 15:15.38 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 15:15.44 | *** join/#asterisk btsteve (n=btsteve@204.10.20.30) |
| 15:18.57 | festr__ | BouYYY: 1.4.4 is ok |
| 15:19.09 | festr__ | BouYYY: i've deadlocks in latest 1.4.4 svn |
| 15:19.17 | pigpen | yeah..working well with 157 sip phones. |
| 15:19.25 | pigpen | here anyway. |
| 15:19.27 | festr__ | pigpen: which revision pls? |
| 15:19.39 | pigpen | rev of what? |
| 15:19.47 | festr__ | pigpen: asterisk -r;show version |
| 15:19.55 | pigpen | sorry 1.4.4 stable'ish |
| 15:20.05 | festr__ | great |
| 15:20.17 | pigpen | I will to a show ver...1 sec |
| 15:20.51 | pigpen | Asterisk 1.4.4 built by root @ asterisk on a x86_64 |
| 15:21.04 | *** join/#asterisk marta (n=marta@nat-percro2.sssup.it) |
| 15:21.05 | Dovid | can anyone help me with featuremaping ? |
| 15:21.14 | *** join/#asterisk ixela (i=ixela@nat/digium/x-5d93efe2f95ff07b) |
| 15:21.27 | BouYYY | I have had intermitted issues with transfering on 1.4.4. |
| 15:22.18 | BouYYY | This is in SIP with Mitel phones. |
| 15:22.28 | festr__ | pigpen: thx |
| 15:22.40 | pigpen | we are using polycom. |
| 15:22.52 | BouYYY | What are they like? |
| 15:23.03 | pigpen | we like them... |
| 15:23.09 | [TK]D-Fender | Polycom > All |
| 15:23.14 | pigpen | they work. |
| 15:23.25 | BouYYY | ... :) |
| 15:23.58 | BouYYY | thats a good start, are they all the same model type? |
| 15:24.04 | marta | hi, I solved the problem I had yersterday with corefile, it was the presence of the application 'apport' installed by default on ubuntu. if it can helpful in future to someone else. tnx |
| 15:25.54 | festr__ | BouYYY: |
| 15:25.54 | festr__ | Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson) |
| 15:27.05 | pigpen | BouYYY, I have many models online (501,600,601,650,430,4000) |
| 15:27.07 | BouYYY | festr__: Is this an outstanding issue or already available? |
| 15:27.22 | festr__ | BouYYY: already in 1.4svn tree |
| 15:27.53 | pigpen | [TK]D-Fender, what is the max number of sip phones I would want to call using the page app? |
| 15:28.18 | [TK]D-Fender | pigpen: Not sure what the practical limit would be. Its a mettme, so whatever rules apply, its the same. |
| 15:28.23 | [TK]D-Fender | MeetMe* |
| 15:28.57 | BouYYY | festr__: Thanks will check this version out |
| 15:29.03 | pigpen | K. I attempted to have it page, achem...157 polycom's yesterday..it only attempted to process 22 of them. |
| 15:29.28 | pigpen | I have done 120 or so using meetme before...but not on * 1.4 |
| 15:33.24 | *** join/#asterisk ThOr101 (n=bthorson@pool-71-126-163-76.washdc.fios.verizon.net) |
| 15:34.09 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 15:34.11 | [TK]D-Fender | pigpen: OH, there is a DIALPLAN STRING LENGTH LIMIT to consider ;) |
| 15:34.25 | pigpen | ewe.....hmm..didn't think of that. |
| 15:34.42 | [TK]D-Fender | <- i r smrt |
| 15:35.03 | pigpen | < us too (multiple personalities) |
| 15:35.25 | pigpen | Yeah...I have: Page(${PAGE_GRP_1}) |
| 15:35.46 | pigpen | and well, lets just say, that variable has quite a bit in it. |
| 15:35.58 | pigpen | So I guess I would break it out? |
| 15:36.00 | pigpen | kinda sucks. |
| 15:36.54 | [TK]D-Fender | pigpen: yeah. Code your own :) |
| 15:37.06 | pigpen | hence my issues with perl. |
| 15:37.14 | *** join/#asterisk Taadow (n=super@70.70.0.33) |
| 15:37.23 | pigpen | but I am learning master. |
| 15:37.56 | [TK]D-Fender | pigpen: This would require C unless you do something silling like Page-ing Page's to nest them (only dialplan way). |
| 15:38.10 | pigpen | yeah..that is what I was thinking.... |
| 15:38.26 | pigpen | nest something, or pre-record the announcement. |
| 15:38.44 | pigpen | then play it back either individually or via a meetme. |
| 15:38.55 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 15:39.18 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 15:39.31 | [TK]D-Fender | pigpen: Yeah, that works. GroupA, GroupB, and spawn a Call file or Originate to relay it via a recording. |
| 15:40.36 | ThOr101 | So with distinctive ring on, I call in, and I get this: http://paste.debian.net/28832 Do other people actually get the ring values? |
| 15:40.46 | pigpen | would I page them all first, then playback, or process indevidually |
| 15:40.52 | pigpen | geesh..I can't spell either. |
| 15:41.41 | pigpen | ie: page, page, page, playback or page, playback, page, playback, page, playback |
| 15:42.46 | [TK]D-Fender | pigpen: I might group them and do it in a few large batches |
| 15:43.24 | pigpen | well, when I issue the page command, won't it create a seperate meetme for each? |
| 15:43.34 | pigpen | ie: then I would have to playback it for each meetme. |
| 15:43.55 | pigpen | shit..I am just going to have to try it. |
| 15:45.22 | [TK]D-Fender | pigpen: Yes, you would do 1 record, and then spawn X calls to page each grouping to playback the same file. |
| 15:46.03 | ThOr101 | hmm, distinctive ring doesn't show up in the doc subdirectory of the source code. That can't be good. |
| 15:46.23 | pigpen | so If I were to Page, then playback, page, then playback, ..... the playbacks would be associated with the correct page correct? |
| 15:46.31 | killfill_ | hi |
| 15:46.34 | pigpen | or would I need to dump each one out to a seperate extension? |
| 15:46.45 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 15:46.56 | Dr-Linux | anybody tried HUDlight? |
| 15:47.07 | [TK]D-Fender | ThOr101: pastebin your zaptel & zapata |
| 15:47.18 | ThOr101 | will do |
| 15:47.18 | [TK]D-Fender | pigpen: Same exten since its the same message |
| 15:47.27 | pigpen | k. |
| 15:48.09 | [TK]D-Fender | pigpen: Neater still : Make the exten an incremental patter and make the recording that number as well. That way consecutive pages survive better on overlap |
| 15:49.24 | pigpen | Ok..I get the survive better on overlap part....but err.... |
| 15:52.17 | pigpen | hmm..I think I need to assign the playback in the meetme. |
| 15:53.29 | [TK]D-Fender | pigpen:Pages USES MeetMe as the engine. you do not call it yourself. |
| 15:53.43 | ThOr101 | http://paste.debian.net/28834 |
| 15:53.55 | pigpen | well, how do I get it to playback the recording in the meetme? |
| 15:54.31 | [TK]D-Fender | pigpen: Scyning a "kickout" MeetMe for when the pager quits would a tricky mess and use a LOT of call-files to pull people in. |
| 15:54.45 | [TK]D-Fender | pigpen: My methods only use Page direct. |
| 15:55.48 | pigpen | yeah...I am using page direct as well.... |
| 15:55.55 | pigpen | I will screw with it for a bit...see what happens. |
| 15:56.05 | pigpen | needless to say, the page app doesn't scale well. |
| 15:56.12 | pigpen | or my brain doesn't. |
| 15:57.35 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 15:59.41 | [TK]D-Fender | pigpen: Actually the real limit is dialpla string length |
| 15:59.55 | [TK]D-Fender | pigpen: were it not for that it'd work fine |
| 16:00.11 | pigpen | very true. |
| 16:01.01 | ThOr101 | In the chan_zap.c, a line exists that prints out the "Detected ring pattern: ....." I don't ever see that line. But my C coding skills leave a lot to be desired as well. |
| 16:02.59 | Zeeek | Hej! |
| 16:05.01 | ThOr101 | Hmm, I wonder if I am setting this in the wrong place. in chan_zap.c it looks like it pulls most of the config stuff out of: tmp->threewaycalling = conf.chan.threewaycalling; but for the distinctive ring: tmp->usedistinctiveringdetection = usedistinctiveringdetection; there is no conf.chan ... I wonder if the distinctive ring isn't supposed to be defined outside of the [channel] block? If that is indeed how the c |
| 16:07.41 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 16:08.05 | [TK]D-Fender | ThOr101: It does indeed belong in the [channel] block. It can be sport specific |
| 16:08.09 | Zeeek | http://x2z.eu |
| 16:08.32 | Zeeek | Thunderstorming like crazy here |
| 16:08.51 | ThOr101 | Yeah, I just moved it out, and it complained that: No category context for line 12 of /etc/asterisk/zapata.conf |
| 16:09.03 | ThOr101 | which was where I put it. Ok, back to the config. |
| 16:10.46 | *** join/#asterisk dacter (n=dlittrel@207.200.33.213) |
| 16:11.29 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 16:12.38 | Zeeek | {{{Katty}}} |
| 16:12.43 | [TK]D-Fender | ThOr101: No, you need it. |
| 16:13.04 | ThOr101 | Yeah I know. I want to see if I can get that line to pop up. Make sure it is seeing the usedistinctive ring. |
| 16:13.06 | [TK]D-Fender | ThOr101: Or as soon as Zapel sees the ring voltage start it won't wait for the opportunity to see a PATTERN in it. |
| 16:13.06 | BSD_Tech | no conf this am |
| 16:13.17 | ThOr101 | Or do you mean, that I need to purchase CallerID? |
| 16:13.18 | Zeeek | BSD_Tech oh? |
| 16:13.23 | BSD_Tech | the conf bridge is not up |
| 16:13.33 | Zeeek | you went thru the conf bridge? |
| 16:13.48 | [TK]D-Fender | ThOr101: You need to tell * to LOOK for it. Succeeding is irrelevent, its that * has to WAIT before passing the call to the dialplan |
| 16:13.53 | BSD_Tech | I enter 22622 it says its invalid |
| 16:14.14 | BSD_Tech | is it not upyet |
| 16:14.17 | Zeeek | It's exactly like my employees: it doesn't work until I'm there |
| 16:14.26 | BSD_Tech | lol |
| 16:14.31 | Zeeek | and it's storming violently here |
| 16:14.42 | Zeeek | where it was sunny and blue like 20 minute ago |
| 16:14.45 | BSD_Tech | zeek can I work for you |
| 16:14.53 | BSD_Tech | heheh |
| 16:14.55 | Zeeek | "If I go down, the server does too" |
| 16:15.05 | ThOr101 | Ok, so that I understand. I just turned callerID off usecallerid=no wouldn't we assume given the above C code that I should see an error in the log file? I didn't. |
| 16:15.09 | Zeeek | oh sure, just what I need, another lazy employee! |
| 16:15.18 | BSD_Tech | I am far from lazy |
| 16:15.19 | Zeeek | yyyyyyes, I didn't mean... |
| 16:15.37 | Zeeek | speaking of conference, I need a beer before it begins |
| 16:15.51 | BSD_Tech | I am the hard worker who shows up early and stays till I am kicked out of the office |
| 16:15.53 | [TK]D-Fender | ThOr101: Don't think of it as "bad" but rather "logically incompatible due to related circumstances" |
| 16:16.03 | Zeeek | I used to to do that too when I was an employee |
| 16:16.27 | ThOr101 | I don't think that * is picking up that configuration variable. |
| 16:16.43 | *** join/#asterisk syzygyBSD_ (n=chatzill@24-116-151-94.cpe.cableone.net) |
| 16:16.46 | Zeeek | muhahaha bridge up but you still can't talk til I phone in |
| 16:16.58 | BSD_Tech | lol |
| 16:16.59 | syzygyBSD_ | Good morning all |
| 16:17.12 | Zeeek | Folks if you want to hear the latest Nufone scoop, you'll need to join the conference |
| 16:17.17 | ThOr101 | [TK]D-Fender check out this code snipped: http://paste.debian.net/28838 |
| 16:17.23 | *** join/#asterisk wunderkin (i=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 16:17.29 | Zeeek | I'll be there shortly, I need to go through makeup and stop at the bar |
| 16:17.38 | [TK]D-Fender | ThOr101: Looks like it should whine... |
| 16:17.44 | ThOr101 | exactly |
| 16:18.07 | vAd0r | Anyone have any viatalk coupon codes? |
| 16:18.10 | [TK]D-Fender | ThOr101: Does it take 2 rings to start firing up the dialplan currently? |
| 16:18.45 | syzygyBSD_ | anyone know a good sip provider for NZ? |
| 16:18.49 | masked | 'the ring of fire'' |
| 16:19.32 | ThOr101 | [TK]D-Fender: I think it is two by my best analysis: http://paste.debian.net/28840 |
| 16:20.26 | [TK]D-Fender | ThOr101: 5s ring spread. Feels kinda "normal" |
| 16:20.39 | [TK]D-Fender | ThOr101: Try a "distinctive" ring now |
| 16:21.09 | ThOr101 | Should I go back and correct my config or leave it purposefully broken? |
| 16:21.45 | ThOr101 | and the previous output was a distinctive ring. |
| 16:22.07 | *** join/#asterisk awannabe (n=hjh@ip24-251-135-202.ph.ph.cox.net) |
| 16:22.28 | Taadow | HELP! Anyone know what this means? WARNING[11087] channel.c: No path to translate from SIP/sky3-00650880(256) to Zap/1-1(68) |
| 16:22.42 | awannabe | hi guys, im having a problem with call parking, i have it setup for #1 to park a call, but when you press # during a call then it says "transfer to extension", how can I turn that off? |
| 16:23.20 | [TK]D-Fender | Taadow: Means you can't transcode to/from G.729. Go make sure you have enough available licenses for your call. |
| 16:23.34 | Taadow | Aha! I did add a 2nd nick, lemme re-run the registration. |
| 16:24.03 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:24.12 | Taadow | Oh!!! Pass thru is working but not the other... I believe you're onto something good sir. :) |
| 16:24.27 | *** join/#asterisk Alric (n=nbowyer@fr-cg.1stel.com) |
| 16:24.40 | [TK]D-Fender | Taadow: you're not "passing through" if you're going to Zap. |
| 16:25.41 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 16:26.09 | Taadow | Indeed, you are genious. |
| 16:26.13 | Taadow | Err, genius. heh |
| 16:28.33 | [TK]D-Fender | ThOr101: So now you'll have well documented problems :) |
| 16:29.24 | ThOr101 | Heh heh. The road to well documented problem, and sufficiently wasted a load of time on a stupid thing you missed are two parallel roads. Right now, I have my foot on each of them. |
| 16:30.42 | Alric | When zttest shows passes that are 77%, 85%, etc, does this indicate frame slips on the primary clock source? |
| 16:30.43 | [TK]D-Fender | ThOr101: Now watch the tectonic plates shift and you begin your journey to soprano-dom :) |
| 16:31.00 | masked | there are no parallel's |
| 16:31.00 | [TK]D-Fender | Alric: means you are ROYALLY FUBAR'd |
| 16:31.07 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 16:31.10 | masked | you can't afford to draw any right now |
| 16:31.20 | [TK]D-Fender | Alric: pastebin "cat /proc/interrupts" |
| 16:31.22 | Alric | [TK]D-Fender: So how do you become un-royalled fubar'ed :D |
| 16:32.01 | Alric | http://www.pastebin.ca/509716 |
| 16:32.40 | [TK]D-Fender | Alric: Wow.... 2 1st gen's |
| 16:32.42 | *** join/#asterisk alrs (n=lars@170.206.224.58) |
| 16:33.00 | Alric | How can you tell that from proc/interrupts? |
| 16:33.10 | Alric | The others would be using te1xxp drivers? |
| 16:33.12 | [TK]D-Fender | Alric: Clean system otherwise... looks OLD. What OS, and ver of */zaptel? |
| 16:33.25 | [TK]D-Fender | Alric: Correct |
| 16:33.39 | Alric | Zaptel was just upgraded to the latest 1.2, I guess thats 1.2.17.1. OS is, ha ha... RH9 :) |
| 16:33.54 | *** join/#asterisk `Sean (i=Un1x@CPE000c248d137c-CM00111ae601f8.cpe.net.cable.rogers.com) |
| 16:34.04 | [TK]D-Fender | Alric: Hrm.... grab another drive, install something modern on it. test. |
| 16:36.27 | ThOr101 | AH HA! Lil F*())(rs, I've got you now! [May 25 12:35:27] NOTICE[25125]: chan_zap.c:7378 mkintf: A1 Setting the variable usedistinctiveringdetection to: 0 |
| 16:36.51 | Alric | Yeah, its one of those systems that was installed back when Asterisk was at 0.7.0, and has worked since then... until now :( |
| 16:37.00 | *** join/#asterisk DaveCanoe (n=Dave@ool-45789009.dyn.optonline.net) |
| 16:37.15 | [TK]D-Fender | Alric: 1st test is the easy one. |
| 16:37.16 | *** join/#asterisk n00dle (n=ccraft@hillel.springsips.com) |
| 16:37.29 | Alric | So you're thinking OS? |
| 16:37.35 | [TK]D-Fender | Alric: Next involved upping the stakes and taking that new drive to a new system with the same cards. |
| 16:37.41 | [TK]D-Fender | Alric: Last is new cards |
| 16:37.49 | Alric | Luckily I brought a whole new system with me... |
| 16:37.57 | [TK]D-Fender | Alric: OS is the fisrt and easiest to test. |
| 16:38.31 | [TK]D-Fender | Alric: Transplanting drive & cards to other system for testing is fairly easy afterwards |
| 16:40.06 | Alric | Well, at least I know where to look now. Before it was just like "eh? problem? tests are clear, no errors, can't duplicate..." |
| 16:40.58 | Alric | Odd that I've gotten no faxing complaints though (*knocks on wood*). Seems like these errors would kill any faxes. |
| 16:42.10 | [TK]D-Fender | Alric: Should massacre them IMO |
| 16:42.48 | pigpen | [TK]D-Fender, how horrible would it be if I just wrote a perl script, interfacing with the manager, to have each 157 phones call an extension which does a playback individually of an announcement? |
| 16:43.35 | [TK]D-Fender | pigpen: To tell you the truth I don't think it'd be any worse ont he system. |
| 16:43.48 | pigpen | yeah..kinda what I was thinking. |
| 16:43.56 | pigpen | k..tks. |
| 16:44.05 | pigpen | bbiab. |
| 16:44.07 | [TK]D-Fender | pigpen: Only thing is to figure how to pick them. |
| 16:44.18 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 16:44.20 | pigpen | define how to pick them? |
| 16:44.27 | pigpen | ie: which ones to dial? |
| 16:44.40 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 16:44.45 | [TK]D-Fender | pigpen: Yup.... can't be done in the dialplan particularly easily. |
| 16:44.57 | [TK]D-Fender | pigpen: Various ways to cheat, but all are hackish |
| 16:45.10 | pigpen | yeah. I have created a script that awk's out the extensions....I cheated. |
| 16:45.12 | *** join/#asterisk DaveCanoe (n=Dave@ool-45789009.dyn.optonline.net) |
| 16:45.27 | vAd0r | do you need a register string when setting up a trunk to a cisco call manager? |
| 16:45.30 | pigpen | down the road, I will have ruby create the files. |
| 16:45.50 | [TK]D-Fender | pigpen: Here's what I'd do : use a SetVar in sip.conf like "setvar=pagegroup=1" and parse sip.conf live for paging groups. |
| 16:46.22 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 16:46.24 | [TK]D-Fender | vAd0r: Registering has nothing to do with placing calls. |
| 16:46.40 | pigpen | hmm...an idea. |
| 16:46.58 | vAd0r | im having one way voice issues w/ it |
| 16:47.09 | [TK]D-Fender | vAd0r: Unrelated. |
| 16:47.13 | vAd0r | but i have nothing on incoming settings |
| 16:47.14 | pigpen | for that matter, if I have my way, I will just query postgres for the info....but first I need to get asterisk to query variables from psql. |
| 16:47.19 | vAd0r | do i need something on that |
| 16:47.20 | pigpen | yet another project. |
| 16:47.41 | [TK]D-Fender | vAd0r: Incoming also has nothing to do with outgoing connections. |
| 16:47.52 | vAd0r | k |
| 16:47.58 | [TK]D-Fender | vAd0r: Only general networking problems are factored in here |
| 16:48.21 | vAd0r | that sip trunk is through ipx tunnel |
| 16:48.41 | vAd0r | i think it gets screwed up |
| 16:48.48 | vAd0r | as we can't have 2way convo |
| 16:48.53 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-96-242.ph.ph.cox.net) |
| 16:49.41 | vAd0r | what is the code to check vm |
| 16:51.19 | *** join/#asterisk ManxPower (n=manxpowe@246.sub-70-221-223.myvzw.com) |
| 16:56.41 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 16:58.00 | neverblue | morning |
| 16:58.26 | ThOr101 | It looks like the variable is getting read correctly from the config. What a mess it is in there. Somebody hand me a 10 blade. |
| 17:02.53 | ThOr101 | Ok, enough with the debug statements. Let's just start changing some code! |
| 17:07.46 | [TK]D-Fender | ThOr101: a "10 blade"? |
| 17:07.57 | ThOr101 | Knife for surgery |
| 17:08.04 | [TK]D-Fender | ThOr101: How's this instead? http://aocomputing.net/bushi/ |
| 17:08.47 | ThOr101 | Isn't it pointed in the wrong direction? Unless of course the owner is at war with someone. |
| 17:09.07 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 17:09.24 | [TK]D-Fender | ThOr101: Yes, I had it reverse on the stand... didn't know the proper way at first :) |
| 17:09.49 | [TK]D-Fender | ThOr101: Long since corrected and have bought a new wall-mount to accomodate my new kat's |
| 17:10.00 | ThOr101 | Me neither until I got myself a brother in law who always enjoyed pointing out that fact at all the sushi places we dine at :-) |
| 17:10.50 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 17:11.24 | irule | hi there guys, heres my question http://pastebin.ca/509821 please and thanks |
| 17:11.45 | [TK]D-Fender | ThOr101: Its more for the thoguh of strain on the blade so it doesn't go against the curve |
| 17:12.00 | ThOr101 | Did your cats like to jump on the knives? That out to be a pretty quick lesson. |
| 17:12.23 | [TK]D-Fender | irule: like the instrustions say, you don't put the exension in Playback. |
| 17:12.44 | [TK]D-Fender | ThOr101: No, but I could tell you about a brown fox ;) |
| 17:13.03 | ThOr101 | And lazy brown dogs? |
| 17:13.34 | [TK]D-Fender | ThOr101: indeed! |
| 17:14.24 | ThOr101 | My debug statements that include pointers don't seem to make * very happy. Oh what a can of worms in a dark hole I've opened. |
| 17:16.17 | *** join/#asterisk vAd0r (n=IceChat7@216-201-139-51.res.logixcom.net) |
| 17:16.23 | vAd0r | what do i dial to check vm on analog phone |
| 17:16.38 | [TK]D-Fender | ThOr101: You've uncovered the problem and have found a path to the solution. The most important part is done. Next you may want to get someone knowledgable to help you implement the fix. |
| 17:16.51 | [TK]D-Fender | vAd0r: Whatever you set in your dialplan. |
| 17:17.06 | vAd0r | hmm |
| 17:17.14 | vAd0r | is it not already set there |
| 17:17.39 | ThOr101 | GOT IT! |
| 17:18.06 | [TK]D-Fender | vAd0r: Nothing exists except that which you create. EVERYTHING you want to be able to dial you are responsible for coding in your dialplan. |
| 17:18.35 | vAd0r | which file is that in |
| 17:18.37 | irule | [TK]D-Fender plesase excuse me but, may you please be a little more speciffic? which instructions are you talking about? whick playback are you talking about? |
| 17:18.50 | nahirean | extensions.conf for the dial plan, voicemail.conf to set up the voicemail |
| 17:19.03 | [TK]D-Fender | vAd0r: If you don't know where the dialplan is, you're in serious trouble with Asterisk.... |
| 17:19.23 | nahirean | may want to check out the oreily book |
| 17:19.40 | [TK]D-Fender | irule: you must NOT put a file-type extension when you call Playback, Background, etc. |
| 17:19.44 | vAd0r | And i am sure you knew everything 1 day after you set it up |
| 17:19.48 | [TK]D-Fender | irule: ie NO ".wav" on the end |
| 17:19.50 | vAd0r | just tell me the name please |
| 17:19.59 | ThOr101 | So is it better ti post a bug report or to go to asterisk-bugs? Or you have no idea? |
| 17:20.14 | [TK]D-Fender | vAd0r: Yes I knew extensions.conf on day 1. how have you been able to dial anything at all so far? |
| 17:20.21 | nahirean | vAd0r: I did tell you the name, the dial plan is extensions.conf, and the voicemail is voicemail.conf |
| 17:20.27 | vAd0r | i dial ext to extention |
| 17:20.29 | [TK]D-Fender | vAd0r: And you've been here and working with * far longer than a day |
| 17:20.32 | irule | oh I see thanks |
| 17:20.36 | vAd0r | thank you |
| 17:22.11 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
| 17:22.33 | irule | <[TK]D-Fender> [May 25 10:30:57] WARNING[12198]: app_record.c:138 record_exec: No extension specified to filename! |
| 17:22.34 | irule | <PROTECTED> |
| 17:22.42 | x86_ | anyone got a decent example of an asterisk-to-asterisk switch statement in a dialplan? |
| 17:22.59 | [TK]D-Fender | irule: I didn't say on the RECORD. on the ones that play something BACK. |
| 17:23.07 | x86_ | i've seen asterisk-to-realtime, but i want asterisk-to-remote-asterisk |
| 17:23.45 | [TK]D-Fender | irule: You need the extension on the record to specify the format. Playback/Background look for the "most compatible" format for which you are not supposed to specify it |
| 17:24.06 | irule | oh I see, thanks :) |
| 17:24.16 | [TK]D-Fender | irule: Good.... round 3 time... |
| 17:25.46 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 17:26.25 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 17:28.06 | ThOr101 | If I found a bug in chan_zap.c and I want to report it, would that be a core asterisk bug, or a zaptel bug? |
| 17:28.17 | ThOr101 | seems like core * since it is reading a config file |
| 17:28.19 | [TK]D-Fender | ThOr101: Zaptel |
| 17:28.26 | ThOr101 | Ok then. |
| 17:28.37 | jkiff | Does asterisk support text-only SIP/SIMPLE sessions? I'm casually attempting to get pidgin to register to my asterisk server with no success. |
| 17:29.08 | *** join/#asterisk bdheeman (n=bsd@122.162.1.14) |
| 17:29.14 | ThOr101 | The only version I am given is 1.2.14 ugh. |
| 17:29.18 | [TK]D-Fender | jkiff: No, * does not support SIP messaging at all. |
| 17:29.38 | [TK]D-Fender | ThOr101: Perhaps you should upgrade and see if its been fixed or requires a different fix. |
| 17:29.52 | ThOr101 | upgrade? I'm using 1.4.4 |
| 17:30.08 | *** join/#asterisk _deg_ (n=deg@200.195.161.164) |
| 17:30.12 | ThOr101 | And I don't think this is a zaptel issue, because that project looks like it is concerning itself with zaptel issues. |
| 17:30.18 | ThOr101 | Ahh, i got an answer chan_zap under * |
| 17:30.30 | _deg_ | Hopw could I set an expire time of "forever"? |
| 17:30.37 | _deg_ | Is there a way to do that on sip.conf? |
| 17:30.41 | jkiff | [TK]D-Fender: Ah, I gotcha. Is that by design, or is it not implemented due to lack of time/interest/etc'? |
| 17:30.44 | _deg_ | The default expire time 120 |
| 17:30.47 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 17:31.00 | [TK]D-Fender | jkiff: Both. |
| 17:31.33 | [TK]D-Fender | ThOr101: You are consfusing me between 1.2.14 & 1.4.4 |
| 17:31.43 | *** join/#asterisk RobH (n=RobH@rrcs-24-73-86-239.se.biz.rr.com) |
| 17:32.03 | _deg_ | im doing some reliability and stress tests with SIPP and i dont want to register my extensions averytime i need to start the srtess things |
| 17:32.22 | _deg_ | but i want to register them. |
| 17:32.31 | RobH | Not that it is asterisk software, but I am using it with my asterisk server. Does anyone have a good link for how to change the customized button settings on a polycom ip 601? (What file to change, what the settings are called, so on)? |
| 17:32.33 | _deg_ | just once |
| 17:33.30 | *** join/#asterisk gardo (n=gardo@121.97.211.58) |
| 17:33.52 | jkiff | [TK]D-Fender: Hehe, I suppose a clearer question would be: Would a patch implementing SIMPLE text messaging be accepted? (Assuming the patch didn't suck.) |
| 17:34.19 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 17:35.24 | [TK]D-Fender | jkiff: Uless it clashes with some other plan I can't see why not. |
| 17:36.04 | [TK]D-Fender | RobH: its all in the Admin Guide on Polycom's site |
| 17:38.11 | irule | what advanced * documantation do you recommend? I am ready for the next book! :) |
| 17:39.09 | [TK]D-Fender | irule: Nothing more really. Just the WIKI for individual little bits. |
| 17:39.33 | irule | ok thanks |
| 17:39.56 | [TK]D-Fender | irule: Once you understand how the pieces work, its the combination that makes it interesting. That is up to your needs/imagination. |
| 17:40.36 | ThOr101 | http://bugs.digium.com/view.php?id=9806 And now back to our regularly scheduled configuring. Nothing like filing a bug report on your third day of using the software. Why does everything break around me? ;-) |
| 17:40.39 | *** part/#asterisk bdheeman (n=bsd@122.162.1.14) |
| 17:40.44 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 17:41.54 | ThOr101 | Who wants to send me their root passwords? Anyone? :-) |
| 17:42.00 | ThOr101 | Time to test those backups? |
| 17:42.07 | irule | [TK]D-Fender may you please recommend some excercises? |
| 17:42.42 | [TK]D-Fender | irule: I can't tell you what you want to do. Just go DO it, and show that you are able to read the documentation for the little bits and put them together sanely. |
| 17:42.45 | ThOr101 | Detected ring pattern: 387,0,0 Sweeeeet! |
| 17:42.57 | [TK]D-Fender | ThOr101: Unbroken? |
| 17:43.06 | irule | o hehe thanks anyways |
| 17:44.55 | ThOr101 | Mostly unbroken. It just detect 2 different ring patterns for the ring. Not so distinctive now is it? |
| 17:45.17 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 17:49.10 | ThOr101 | I think I found more broken stuff now |
| 17:49.33 | ThOr101 | It isn't checking the first dring setting (dring1) and it is sending it to the wrong context. |
| 17:51.37 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 17:52.21 | n00dle | Ok, what if I find a bug that claims it's solved in the bugtracker? |
| 17:53.17 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 17:55.24 | n00dle | Ok, here it is: Line rings, lights on stations start blinking. I press the button, get 404, and caller hangs up... LED blinks eternally, until I restart *. |
| 17:57.18 | *** part/#asterisk putnopvut (n=putnopvu@69-94-197-46.biltmorecomm.com) |
| 17:58.21 | n00dle | "sla show stations" says "SLA_TRUNK_STATE_RINGING" on those lines, but the lines are hung up, cli sez " -- Hungup 'Zap/pseudo-1726078533' |
| 17:58.21 | n00dle | <PROTECTED> |
| 17:58.21 | n00dle | " |
| 17:59.59 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 18:05.13 | n00dle | The only modifications I've done besides naming the trunks/stations was to add a parameter to [macro-slaline] to specify a voicemail box for the call to go to... |
| 18:05.35 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 18:05.53 | ThOr101 | I care, but I don't know enough to help you. |
| 18:06.00 | [TK]D-Fender | n00dle: "SLA" as it exists now is a freakish hack usable by only those with tons of buttons w/ presence and a handful of analog lines to match. |
| 18:06.22 | ThOr101 | [TK]D-Fender cares too |
| 18:06.22 | [TK]D-Fender | n00dle: As such I wouldn't even bother. |
| 18:06.31 | [TK]D-Fender | ThOr101: Yeah... only not so much ;) |
| 18:07.07 | n00dle | Well, it happens to be a hack that my boss is requiring... gods, I hope I don't have to dive into source... too many other projects. |
| 18:07.56 | n00dle | Hang on... I think I caught a typo. |
| 18:09.54 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 18:10.08 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:10.37 | ThOr101 | so I have a dring2context=incoming, and that seems to be working. But is my context worded incorrectly to cause the included error? http://paste.debian.net/28848 |
| 18:12.05 | ThOr101 | it didn't read in the directed context, that is why it says " ,s,1" there is supposed to be a context in there isn't there. |
| 18:12.12 | vAd0r | What do i need to do to allow the calls to pass through viatalk. they keep answering them |
| 18:12.20 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:13.31 | *** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 18:17.00 | ThOr101 | Found another bug. Ugh. |
| 18:17.52 | ManxPower | ThOr101: That is not surprising. Very few people use distinctive ring with Asterisk |
| 18:18.52 | ThOr101 | Well I guess I am the tester. Surprising. I thought it would be fairly well used. Thanks for the insight. |
| 18:19.51 | ManxPower | ThOr101: Asterisk is a PBX and PBXs don't normally support distinctive rings. If a customer is too cheap to get 2 phone lines, then they are normally too cheap to get a PBX. |
| 18:21.22 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 18:21.51 | ThOr101 | Makes sense. I am using * more as a IVRS & glorified answering machine than a PBX. |
| 18:22.37 | ManxPower | *nod* Asterisk was never designed for answering machine use. |
| 18:22.59 | [TK]D-Fender | ManxPower: And all too often thats what people want it for... |
| 18:24.12 | ThOr101 | I'm also using it to route calls. Multiplex 1 analog line to multiple other lines. |
| 18:25.12 | ThOr101 | Does the order that things appear in zapata.conf matter? Other than being below [STUFF IN BRACKETS] ? |
| 18:25.35 | ManxPower | Yes, the order ALWAYS matters with respect to the channel= line and the options. |
| 18:25.56 | ManxPower | You SET the option, then you apply the option to a channel using the channel= line. All settings are set until you change them.l |
| 18:26.12 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 18:26.19 | ManxPower | so if you set callerid=bob <1234> then a channel=5, all following channels will have that callerid unless you override it. |
| 18:26.27 | ThOr101 | ahh, so that is how I would configure other channels. And that is why the channel =>1 is currently the last line in my config |
| 18:26.35 | ThOr101 | I gotcha. Thanks. |
| 18:26.38 | ManxPower | you CAN think of this as the file being parsed form the bottom up, but that is not actually the case. |
| 18:27.42 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 18:28.01 | ThOr101 | Starting Zap/1-1 at ,s,1 failed so falling back to exten 's' So there really should be a context in between "at " and ",s,1" correct? |
| 18:30.42 | [TK]D-Fender | ThOr101: pastebin your zapata |
| 18:31.18 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 18:31.24 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 18:32.23 | ThOr101 | http://paste.debian.net/28850 |
| 18:33.06 | *** join/#asterisk DaveCanoe (n=Dave@ool-45789009.dyn.optonline.net) |
| 18:33.55 | ThOr101 | I'm also interested in what the effect of having two context= statements is. (Which is why I asked about order earlier). |
| 18:35.10 | [TK]D-Fender | ThOr101: Looks fine.... perhaps another bug. |
| 18:35.52 | Arsenick-TC2L | [TK]D-Fender: If I don't do the callback part, what kind of info should i look for ? I mean for the Auth. and to allow the caller to dial out.. ? |
| 18:36.21 | [TK]D-Fender | Arsenick-TC2L: "show application disa" |
| 18:37.22 | Arsenick-TC2L | nice! thx |
| 18:38.34 | ThOr101 | Ok, I'll go file that one. If I could ask one last question: http://paste.debian.net/28851 The first call is to the distinctive ring pattern, it ends up going to default. The second one is a normal call, and it goes to demo. How are those two sent to different contexts? Is falling back to default hard coded? The last line of zapata says context=demo. My [default] context in extensions.conf has one line include => d |
| 18:40.29 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 18:40.56 | *** join/#asterisk Cresl1n (n=matt@65-182-39-144.cre.bil.biltmorecommunications.net) |
| 18:40.56 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 18:41.23 | *** part/#asterisk RobH (n=RobH@rrcs-24-73-86-239.se.biz.rr.com) |
| 18:41.54 | [TK]D-Fender | ThOr101: Its just completely bombing for lack fo context. The error isn't right. |
| 18:42.08 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 18:42.43 | ThOr101 | Yeah, I just tested something. I put one line in default Wait(15) and another at the head of demo Wait(18). |
| 18:42.52 | ThOr101 | When I call in with distinctive ring it goes to default |
| 18:43.17 | ThOr101 | When I call in with a regular ring, it goes to demo. Looks like the distinctive ring overrides zapata.conf and just goes to default. |
| 18:43.59 | [TK]D-Fender | ThOr101: Yeah, I knows its supposed to bypass the "context=" line so it'll boomb everywhere ELSE first |
| 18:44.43 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:44.43 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:44.57 | x86_ | anyone ever use the switch statement in a dialplan to talk to a remote asterisk box? |
| 18:45.12 | ThOr101 | Interesting. Well at least I found a way to go to 2 different contexts with distinctive ring ;-) |
| 18:46.02 | champster | Anyone know of a working webphone that can be used to let people call into our asterisk server as an outside caller? |
| 18:47.39 | Qwell | champster: there is some java iax softphone, or something |
| 18:47.57 | [TK]D-Fender | ThOr101: "I haven't failed... I've just found 100 new ways that don't work" |
| 18:47.58 | docelmo | Say can someone explain the queue timeout option? |
| 18:48.20 | Qwell | docelmo: how long before it calls a queue member busy |
| 18:48.22 | [TK]D-Fender | champster: Remove the concept of outside & inside caller. |
| 18:48.32 | docelmo | I am looking for a way if the customer has been in the queue for say 2 minutes it will dump them out of the queue back into the dialplan |
| 18:48.48 | Zeeek | Today's conference can be heard here: http://x2z.eu/trixbox.htm |
| 18:48.49 | Qwell | docelmo: Dial() a local channel that calls queue, and put the timeout on the Dial() |
| 18:48.51 | [TK]D-Fender | docelmo: set that timeout when you call Queue in the dialplan |
| 18:48.52 | Zeeek | goodnight |
| 18:48.52 | champster | I would need to make sure that they couldn't make calls as one of our extensions. |
| 18:48.57 | Qwell | Zeeek: trixbox? |
| 18:49.05 | Zeeek | did I say that? |
| 18:49.07 | champster | I would want to treat it like a trunk call |
| 18:49.27 | Zeeek | there is the nufone scoop too |
| 18:49.29 | champster | I knwo you meant to think in contexts |
| 18:49.39 | docelmo | Zeeek when is the conference? |
| 18:49.40 | Zeeek | But you're not interested in that stuff |
| 18:49.47 | Qwell | nufone scoop? |
| 18:49.51 | docelmo | nevermind its about trixbox.. :) |
| 18:49.59 | Zeeek | docelmo it's earlier int he day, Fridays at 12:30 PM EDT |
| 18:50.11 | docelmo | sigh |
| 18:50.12 | Zeeek | Nufone scoop. |
| 18:50.23 | Zeeek | Well it was about trixbox *this* time. |
| 18:50.24 | Qwell | what nufone scoop? |
| 18:50.33 | Zeeek | About canada? |
| 18:50.38 | Zeeek | ask JerJer |
| 18:50.40 | Qwell | what about canada? |
| 18:50.47 | Zeeek | ~seen JerJer |
| 18:51.39 | jbot | jerjer is currently on #asterisk, last said: 'yep'. |
| 18:51.40 | champster | lol |
| 18:51.40 | Zeeek | "426 days ago..." |
| 18:51.45 | Zeeek | heh |
| 18:51.47 | n00dle | ...and I still have no explanation as to why my setup is b0rk3d. |
| 18:51.59 | docelmo | Qwell ok here is what I am looking for.. Im looking to have a caller enter the queue.. then say if they are in there for X amount of seconds they get dumped back into the dialplan and sent elsewhere |
| 18:53.07 | Qwell | docelmo: what I said should work |
| 18:53.40 | *** join/#asterisk bbryant (i=Brett@65-182-39-131.cre.bil.biltmorecommunications.net) |
| 18:54.03 | Zeeek | "but NuFone is preparing to launch C******n DIDs within the next month or so, along with a few other related service offerings." |
| 18:54.23 | Qwell | old news |
| 18:54.38 | Zeeek | not that old, apparently |
| 18:54.46 | docelmo | I already offer Canada |
| 18:54.48 | docelmo | :) |
| 18:55.06 | Qwell | docelmo: all of Canada? |
| 18:55.07 | Zeeek | so do I. You can buy the whole country from me for $1,000 |
| 18:55.09 | Qwell | or just Canada DIDs? |
| 18:55.21 | Zeeek | I'm selling it first |
| 18:55.37 | docelmo | Qwell yes.. about 70% |
| 18:55.43 | [TK]D-Fender | Zeeek: Va t'ens tabarnac ;) |
| 18:57.17 | *** join/#asterisk sysdebug (n=chatzill@200.195.161.164) |
| 18:57.17 | Zeeek | wtf is tarbarnac when it's at home having a lager? |
| 18:57.36 | [TK]D-Fender | Zeeek: Figured being where you are you might better understand it than most, but lets says its sufficiently vulgar :) |
| 18:57.43 | [TK]D-Fender | not "lager" ;) |
| 18:58.05 | Zeeek | some kind of canadian pervesion of French, eh? |
| 18:58.45 | tzanger | heh |
| 18:58.48 | tzanger | tabarnac! |
| 18:58.56 | Zeeek | http://www.montrealite.com/catalog/index.php?cPath=22 |
| 19:00.18 | [TK]D-Fender | Zeeek: From the word "tarbarnacle", where relgious words have often become swear words in outrage against the oppressions of the church. |
| 19:00.31 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 19:00.35 | Zeeek | good. I hate all organized religions |
| 19:00.51 | [TK]D-Fender | Zeeek: So in in rough context for you : "Get the ^%#$ out!" |
| 19:01.24 | karlhaines | Zeeek: me too! it's a lot of BS if you really consider things spiritually, how you think any higher being might consider it |
| 19:02.08 | [TK]D-Fender | Zeeek: its hard to properly lynch someone without organization ;) |
| 19:02.11 | Zeeek | organized religion includes linux |
| 19:02.21 | Zeeek | and all distros :) |
| 19:02.44 | Zeeek | it includes all vinyl and audio geeks |
| 19:03.14 | Zeeek | it includes (especially) anyone who actually thinks .NET and ASp rocks |
| 19:03.26 | Hmmhesays | asp is fine for web shit |
| 19:03.27 | Zeeek | it includes all installations of IIS, especiall 4,5 and 6 |
| 19:03.42 | Zeeek | asp would be fine if it ran on a real server |
| 19:03.58 | Zeeek | just another language |
| 19:04.03 | ThOr101 | .NET and ASP worshipers are more of a cult |
| 19:04.05 | Hmmhesays | pretty much |
| 19:04.24 | ThOr101 | The MONO people just outright scare me |
| 19:04.30 | Zeeek | if Tom Cruise was a programmer he would use what manguage? ASP!!!! |
| 19:04.37 | karlhaines | Hmmhesays: php |
| 19:04.42 | Zeeek | s/manguage/language/ |
| 19:04.59 | ThOr101 | Ruby on rails, to build a car to take him to outerspace to meet the mother ship |
| 19:05.09 | Zeeek | nah, php is more Wesley Snipes |
| 19:06.19 | Zeeek | Matt Damon would be python |
| 19:06.25 | Zeeek | Woody Allen, a basic program that modified itself using PEEK and POKE |
| 19:07.13 | Zeeek | so if we all agree, I can go veg out in front of the tv? |
| 19:08.19 | ThOr101 | Only if you agree that Woody Allen would only program in Basic on a Timex Sinclair |
| 19:08.32 | Zeeek | yes, I'm good with that |
| 19:08.47 | tzanger | awesome |
| 19:08.49 | tzanger | http://www.pastebin.ca/510082 |
| 19:09.03 | ThOr101 | Enjoy your TV viewing ;-) |
| 19:09.36 | Zeeek | Thanks, it will be commercial free |
| 19:11.08 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 19:13.04 | *** join/#asterisk tzafrir_laptop (i=tzafrir@69-94-204-127.biltmorecomm.com) |
| 19:15.21 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-96-242.ph.ph.cox.net) |
| 19:18.21 | *** join/#asterisk harleya (n=harleya@c-67-177-0-233.hsd1.ut.comcast.net) |
| 19:21.08 | ThOr101 | This noob just got a dialtone and a background audio playback. Whoo hoo! |
| 19:22.44 | n00dle | It took wireshark and a hub, but... |
| 19:23.15 | karlhaines | anyone know of any docs descibing options for fault tolerance options for asterisk systems? i |
| 19:23.31 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 19:24.06 | karlhaines | i've noticed in my phone config that there are spots for multiple SIP servers, but not really sure how that whole thing works |
| 19:24.10 | Mercestes | <PROTECTED> |
| 19:24.12 | Mercestes | damnit |
| 19:24.24 | karlhaines | Mercestes: me? |
| 19:24.33 | Mercestes | karlhaines, yes. |
| 19:24.48 | Mercestes | no, of course not, I mistyped something. |
| 19:24.58 | Mercestes | and, server.1 server.2 server.3 btw |
| 19:25.06 | karlhaines | Mercestes: wtf is se5r ? |
| 19:25.19 | Mercestes | karlhaines, Assuming they are polycoms...which is probably a stupid assumption on my part. What TYPE of phone?? |
| 19:25.23 | ThOr101 | n00dle I just bout 3 4 port Netgear hubs the other day. Wow those things are getting hard to find. |
| 19:25.57 | [TK]D-Fender | n00dle: ... |
| 19:25.59 | [TK]D-Fender | ~gs |
| 19:26.13 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:26.46 | ThOr101 | Ok, well I'm out o' here. Don't want to be tempted to chat and/or ask dumb questions. I'll be back when I'm stumped. Thanks for all your help, especially you [TK]D-Fender and n00dle . There's no place like 0, there's no place like 0 |
| 19:28.13 | Mercestes | ... |
| 19:28.18 | Mercestes | what a strange, strange little man |
| 19:30.45 | Hmmhesays | heh |
| 19:30.45 | Hmmhesays | this is IRC, what do you expect? |
| 19:30.45 | n00dle | [TK]D-Fender: Yeah... but I'm learning to deal with their quirks. |
| 19:30.45 | [TK]D-Fender | Hmmhesays: We somehow thought he'd be ... Taller :) |
| 19:30.45 | Hmmhesays | lol |
| 19:30.45 | Hmmhesays | I think my dvd drive is going to start on fire |
| 19:31.56 | Mercestes | Hmmhesays, take pictures |
| 19:32.02 | Hmmhesays | lol |
| 19:32.13 | Hmmhesays | its an old 2x and i'm ripping a dvd with it |
| 19:32.19 | Mercestes | ... |
| 19:32.20 | Mercestes | take pictures |
| 19:34.43 | Hmmhesays | Hey [TK]D-Fender: i got my first stylus pick yesterday |
| 19:34.55 | [TK]D-Fender | Hmmhesays: What is that exactly? |
| 19:35.16 | Hmmhesays | http://www.styluspick.com/ |
| 19:35.29 | *** join/#asterisk MrChicken (n=Dorphals@200.71.58.39) |
| 19:35.43 | Hmmhesays | frustrating, but it forces you to have good right hand technique |
| 19:36.04 | [TK]D-Fender | Hmmhesays: ...... |
| 19:36.20 | Hmmhesays | click the link |
| 19:37.05 | [TK]D-Fender | Hmmhesays: I DID |
| 19:37.10 | Hmmhesays | http://www.styluspick.com/theory.htm |
| 19:37.13 | [TK]D-Fender | Hmmhesays: GIMMICKS.... |
| 19:37.20 | Hmmhesays | [TK]D-Fender: negative |
| 19:37.26 | [TK]D-Fender | Hmmhesays: I'd read that before you linked it |
| 19:37.30 | Hmmhesays | ok |
| 19:37.35 | Hmmhesays | it works incredibly well |
| 19:37.51 | [TK]D-Fender | Hmmhesays: Record something for me! |
| 19:38.13 | Hmmhesays | well i'm only about about 140bpm right now |
| 19:38.18 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:38.21 | justdave | hmm, so I keep getting this in my logs over and over: |
| 19:38.22 | justdave | Rotated Logs Per SIGXFSZ (Exceeded file size limit) |
| 19:38.22 | justdave | May 25 12:37:23 WARNING[23862]: format_wav.c:247 update_header: Unable to find our position |
| 19:38.26 | justdave | <PROTECTED> |
| 19:38.28 | justdave | Asterisk Event Logger restarted |
| 19:38.31 | justdave | Asterisk Queue Logger restarted |
| 19:38.36 | justdave | the logfiles are all tiny, so no idea which file is exceeding size limit |
| 19:38.43 | Hmmhesays | look at logger.conf |
| 19:38.45 | justdave | and the format_wav thing being thrown in there is just weird |
| 19:39.35 | [TK]D-Fender | Hmmhesays: Yeah, you've gott pick that up a notch.... so go record something for me to hear! |
| 19:39.50 | justdave | what am I looking for in logger.conf? |
| 19:40.07 | justdave | there's 5 lines not commented out, including the [general] and [logfiles] headers |
| 19:40.34 | justdave | console, messages, and full are the defined files |
| 19:40.49 | karlhaines | Mercestes: yes, they are polycoms, not a stupid assumption, i think most anyone who could want/afford a somewhat fault tolerant voip pbx would probably be using a nice phone ;) |
| 19:41.17 | justdave | the thousands of copies of messages and full that it's created in the last half hour are all tiny. |
| 19:41.54 | [TK]D-Fender | karlhaines: * is NOT fault tolerant. * can be a part of a fault tolerant system, but that usually starts with a front-end like SER / OpenSER |
| 19:43.47 | *** join/#asterisk ToyMan (n=Stuart@74-32-0-75.dsl1.mdl.ny.frontiernet.net) |
| 19:44.12 | justdave | hmm, asterisk just crashed. guess it's too late to debug it |
| 19:44.23 | justdave | the rotating endlessly problem is gone after restarting it |
| 19:44.34 | justdave | so something was in a mood, and not really an oversized file |
| 19:46.08 | Mercestes | karlhaines, you'd be surprised. |
| 19:46.31 | Mercestes | [TK]D-Fender, or redundant servers with a rollover agreement with your PRI provider (with redundant PRIs) |
| 19:48.58 | *** join/#asterisk bbryant (i=Brett@65-182-39-131.cre.bil.biltmorecommunications.net) |
| 19:50.22 | karlhaines | Mercestes: so what is se5r ? |
| 19:50.35 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 19:50.53 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:52.21 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 19:53.41 | Mercestes | karlhaines, a typo |
| 19:56.55 | karlhaines | oh, lol |
| 19:57.16 | karlhaines | well, anyone have any suggestions on the fault tolerance thing? or links, etc? |
| 19:57.31 | [TK]D-Fender | karlhaines: Was I not blatant enough? |
| 19:57.48 | [TK]D-Fender | karlhaines: and give you a hint on what Mercestes typo'd? |
| 19:58.25 | *** join/#asterisk galeras (n=root@201.244.240.115) |
| 20:01.32 | Hmmhesays | ponders what? |
| 20:04.11 | *** join/#asterisk oej (n=olle@65.124.181.2) |
| 20:04.15 | crimethinker | scox is up 28% today. |
| 20:05.23 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-96-242.ph.ph.cox.net) |
| 20:06.18 | syzygyBSD_ | anyone know of a good sip privider in NZ? |
| 20:06.40 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 20:09.23 | karlhaines | [TK]D-Fender: i apparently didn't see what you typed |
| 20:09.59 | karlhaines | [TK]D-Fender: i see now, thanks, no need to be rude |
| 20:10.31 | [TK]D-Fender | karlhaines: No prob... just wondering if I was somehow just completely deficient today :) |
| 20:11.42 | *** join/#asterisk oej (n=olle@65.124.181.2) |
| 20:12.46 | *** join/#asterisk scurb (n=scurb@c-25aae355.14-16-64736c13.cust.bredbandsbolaget.se) |
| 20:21.59 | *** join/#asterisk naitram (n=ttech@216.77.58.40) |
| 20:22.07 | *** part/#asterisk naitram (n=ttech@216.77.58.40) |
| 20:23.45 | justdave | ah, here's my problem exactly: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172241.html |
| 20:24.47 | justdave | they never found a resolution either |
| 20:24.56 | *** join/#asterisk CrazyTux (n=CrazyTux@spyglass.houston.hostgator.com) |
| 20:28.29 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 20:28.29 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 20:36.25 | *** join/#asterisk Marx2 (i=mnazarko@ip159-c14.gl.digi.pl) |
| 20:37.26 | Marx2 | hello, i'm lloking for help with configuring Phonejack in Asterisk@Debian |
| 20:39.07 | *** join/#asterisk shinao1 (n=shinao1@196.3.63.252) |
| 20:41.23 | Marx2 | maybe somebody can tell me where can I get help, Digium forum can't help too |
| 20:45.52 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-28.plaicmtc01.tx.dh.suddenlink.net) |
| 20:46.27 | Marx2 | so many people and no answer :( |
| 20:48.36 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:49.27 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:49.59 | *** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:53.39 | *** join/#asterisk CrazyTux (n=CrazyTux@spyglass.houston.hostgator.com) |
| 20:56.49 | matsk | Maybee PhoneJack is a obscure less used product so the knowledge is less |
| 20:58.59 | *** join/#asterisk marcan (i=1337@198.Red-83-54-248.dynamicIP.rima-tde.net) |
| 21:02.22 | [TK]D-Fender | matsk, its antiquated and not really supported |
| 21:03.49 | Mercestes | Can anyone suggest a PRI splitter that can take a single PRI and split it to two lines and provide real time failover to two servers? |
| 21:08.40 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:08.44 | pipwerk | Mercestes: junghanns isdnguard |
| 21:08.57 | *** join/#asterisk DaveCanoe (n=Dave@adsl-70-235-73-216.dsl.mrdnct.sbcglobal.net) |
| 21:14.23 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 21:14.31 | *** join/#asterisk DaveCanoe (n=Dave@adsl-70-235-73-216.dsl.mrdnct.sbcglobal.net) |
| 21:15.24 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 21:17.43 | Mercestes | I have serious doubts about a company that doesn't advertise their prices. |
| 21:20.08 | Mercestes | Anything else anyone? Hot ISDN failover devifes? |
| 21:20.14 | Mercestes | s/devifes/devices/ |
| 21:20.18 | pipwerk | Mercestes: so find a reseller, speakup.nl advertises these for 500 euro |
| 21:20.36 | Mercestes | Oh yes, I have 500 euro in my pocket. |
| 21:21.19 | pipwerk | on the scale of a pri, 500 euro seems not that steep |
| 21:21.47 | pipwerk | a decent single port pri interface will set you back double that |
| 21:23.30 | Marx2 | so why it's written on asterisk homepage that phonejack IS supported? |
| 21:23.38 | Mercestes | I really don't think my company (in America) would approve a proposal estimated in euros from a company in the Netherlands for a product manufactured in Austria. Is there anything with US resources? I really don't want to attempt an international lawsuit if the thing catches fire and burns down my main server room. |
| 21:24.08 | Mercestes | Marx2: call Digium and ask them. Why are you asking us? |
| 21:24.09 | pipwerk | Mercestes: so find a local reseller |
| 21:24.34 | Marx2 | because they don't answer at all, you do ;) |
| 21:24.35 | Mercestes | pipwerk, Do you happen to know one? |
| 21:24.47 | pipwerk | Mercestes: yes speakup is local to me :) |
| 21:24.52 | Mercestes | Marx2: Right, and my answer was ,"call digium and ask them." |
| 21:24.52 | Mercestes | ... |
| 21:25.23 | Marx2 | i can't call them cause I live in Poland and my VOIP isn't configured yet... |
| 21:25.23 | Mercestes | Does anyone know of a PRI failover unit? |
| 21:25.35 | Mercestes | Marx2: then your screwed. |
| 21:25.53 | Mercestes | Marx2: I don't even know what Phonejack is |
| 21:26.15 | Marx2 | it's one of first VOIP cards |
| 21:26.27 | Marx2 | made by Quicknet |
| 21:26.46 | Mercestes | Marx2: Kind of like a Model T ford? |
| 21:26.55 | Marx2 | hehe |
| 21:26.59 | pipwerk | Mercestes: have you ever heard of google? |
| 21:27.26 | pipwerk | Marx2: so you have problems getting it to work? |
| 21:27.31 | Marx2 | this card is very cheap and has port to connect analog phone |
| 21:27.47 | Marx2 | it even has DSP |
| 21:27.57 | Marx2 | so it's still quite good |
| 21:27.59 | Mercestes | Marx2: "is very cheap" pretty much identifies whats wrong. :P |
| 21:28.15 | Marx2 | yes, I have problems with finding docs how to configure it |
| 21:28.26 | Mercestes | pipwerk: Uh, yes, that's how I found the isdnguard on Junghamms with no price tag. |
| 21:28.32 | pipwerk | Mercestes: and you were bitching about 500 euro web listprice for a pri failover? :P |
| 21:28.47 | Mercestes | I dont' care about the price, I care about the denomination. |
| 21:29.18 | pipwerk | so google a bit more, I found a voip-info page about * HA :) |
| 21:29.30 | Marx2 | I'm sure asterisk support this card |
| 21:29.38 | Mercestes | I'm not looking for *, I'm looking for a PRI failover. |
| 21:29.48 | *** join/#asterisk joebob777as7 (n=thomask@71-36-200-115.eugn.qwest.net) |
| 21:30.27 | pipwerk | this is #asterisk, so I assumed your question had something to do with asterisk, sorry, my mistake |
| 21:31.00 | Marx2 | I know I should configure phone.conf - it's conf file for this card. It still exists in asterisk |
| 21:31.04 | joebob777as7 | hey could someone give me a helping hand? I just installed asterisk on ubuntu following this guide http://blog.thegoldfish.net/asterisk-with-freepbx-on-ubuntu-704-desktop-tutorial/ and it had me do this command right before finishing Force the safe_asterisk script to use BASH instead of DASH: |
| 21:31.05 | joebob777as7 | sed -i 's!^#!/bin/sh!#!/bin/bash!' /usr/sbin/safe_asterisk and when i go to start asterisk i get this |
| 21:31.21 | joebob777as7 | root@ltspserver:/home/richard# amportal start |
| 21:31.21 | joebob777as7 | SETTING FILE PERMISSIONS |
| 21:31.21 | joebob777as7 | Permissions OK |
| 21:31.21 | joebob777as7 | STARTING ASTERISK |
| 21:31.21 | joebob777as7 | /usr/sbin/safe_asterisk: line 1: /bin/sh!/bin/sh: No such file or directory |
| 21:31.21 | joebob777as7 | /usr/sbin/safe_asterisk: line 5: /bin/shNOTIFY=ben@alkaloid.net: No such file or directory |
| 21:31.30 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ut.comcast.net) |
| 21:31.40 | *** join/#asterisk sparkylyle (n=e0cc9c1f@nj-65-40-236-62.sta.embarqhsd.net) |
| 21:31.57 | Mercestes | I'm hooking it to an asterisk box. This Junghamms thing looks like a complete hack |
| 21:31.59 | pipwerk | joebob777as7: so the sed script fucked up |
| 21:32.12 | joebob777as7 | that'd be my guess |
| 21:32.16 | Mercestes | no body sells this damned thing in the US |
| 21:32.17 | pipwerk | restore the safe_asterisk script from the originnal |
| 21:32.30 | Mercestes | the only US references I see to it is "uhh..does anyone use this thing?" |
| 21:32.37 | joebob777as7 | pipwerk how do i do that? |
| 21:32.46 | pipwerk | and just edit it and on the first line replace sh with bash |
| 21:33.24 | Mercestes | no product reviews |
| 21:33.48 | joebob777as7 | pipwerk, like this sed -i 's!^#!/bin/bash!#!/bin/bash!' /usr/sbin/safe_asterisk ? |
| 21:34.14 | pipwerk | hmmm, I wouldn't trust that sed |
| 21:34.26 | n00dle | That sed's broken. |
| 21:34.36 | anonymouz666 | whats the difference between nat=yes [general] and nat=yes on each [user] |
| 21:35.12 | pipwerk | joebob777as7: just rerun 'make install' |
| 21:35.24 | [TK]D-Fender | anonymouz666, in [general] tells * it may have to forge the SIP return IP. in the peer, not to trust the one coming in on invites. |
| 21:35.40 | n00dle | sed -i "s/^#!\/bin\/sh/#!\/bin\/bash/" /usr/bin/safe_asterisk |
| 21:36.11 | n00dle | ...but that'll only work on the original... after running the broken sed, all your comments are fskd. |
| 21:36.46 | pipwerk | and yes, safe_asterisk should not use bashisms and call /bin/sh, that is just wrong |
| 21:36.50 | joebob777as7 | n00dle, so i should rerun make install and then run that sed you posted? |
| 21:37.09 | pipwerk | that should do |
| 21:38.11 | n00dle | joebob777as7: Yep. |
| 21:38.12 | pipwerk | Mercestes: so google for 'pri failover' and find out for yourself that there are other like products |
| 21:38.24 | n00dle | ...or just vi the file and fix it. |
| 21:38.28 | Mercestes | I did |
| 21:38.41 | joebob777as7 | n00dle, just did that and got this |
| 21:38.41 | joebob777as7 | bash: !\/bin\/sh/#!\/bin\/bash/": event not found |
| 21:38.48 | Mercestes | I got distracted trying to find one useful US reference to this Austrian crap. |
| 21:38.53 | n00dle | Oh, hang on... |
| 21:39.02 | n00dle | sed -i "s/^#\!\/bin\/sh/#\!\/bin\/bash/" /usr/bin/safe_asterisk |
| 21:39.04 | n00dle | ...there. |
| 21:39.13 | pipwerk | Mercestes: if I had ops, you've had a bankick by now *hint* |
| 21:39.15 | joebob777as7 | btw what does this do? lol |
| 21:39.34 | Mercestes | good think you don't have ops |
| 21:39.40 | Mercestes | s/think/thing/ |
| 21:39.42 | pipwerk | n00dle: use singe quotes |
| 21:39.55 | n00dle | It changes the shell used to interpret the script |
| 21:40.08 | n00dle | pipwerk: Ja. I forget that too... |
| 21:40.16 | Mercestes | Maybe the fact that you'd kick/ban me just because I don't like your suggestion because it has no US resources or references is a key reason why you don't have ops?? |
| 21:40.58 | pipwerk | so stop cursing, thank me for the kind advise and go f*ck yourself or somthing, but stop bitching |
| 21:41.23 | *** part/#asterisk pipwerk (i=pip@ringbreak.dnd.utwente.nl) |
| 21:41.42 | r0d3nt | hahahahha |
| 21:41.47 | r0d3nt | good ol #asterisk ... |
| 21:42.25 | Mercestes | pipwerk: Your a retarded international f*ck. How's that for not cussing you hypocritical prick? |
| 21:42.38 | joebob777as7 | n00dle, ok either i'm just an idiot or something is really screwed up... i just reinstalled asterisk and used your sed and i still get... STARTING ASTERISK |
| 21:42.39 | joebob777as7 | /usr/sbin/safe_asterisk: line 1: /bin/sh!/bin/sh: No such file or directory |
| 21:42.39 | joebob777as7 | /usr/sbin/safe_asterisk: line 5: /bin/shNOTIFY=ben@alkaloid.net: No such file or directory |
| 21:42.44 | r0d3nt | umm Mercestes he already left. |
| 21:42.50 | Mercestes | oh. |
| 21:42.54 | Mercestes | damn, that was a good rant too |
| 21:43.02 | Mercestes | someone copy paste that for me and send it to him later. |
| 21:43.08 | r0d3nt | ... |
| 21:43.28 | n00dle | joebob777as7: Do you use vi or pico? |
| 21:43.29 | [TK]D-Fender | Mercestes, "you're" ;) |
| 21:43.41 | joebob777as7 | n00dle, nano |
| 21:43.58 | r0d3nt | [TK]D-Fender: yeah. .hehe |
| 21:44.09 | Mercestes | Thanks Fender. :) |
| 21:44.13 | n00dle | joebob777as7: Ok, edit the file /usr/bin/safe_asterisk, make sure the first line reads #!/bin/bash |
| 21:44.38 | n00dle | ...and that everywhere else "/bin/bash" appears at the front of the line, change it back to "#". |
| 21:44.41 | mocker | Um, it looks like /bin/sh is in front of every line in /usr/sbin/safe_asterisk |
| 21:44.44 | [TK]D-Fender | Mercestes, YOU'RE an ascerbic ass. Reap what you've sown! |
| 21:44.49 | mocker | ;) |
| 21:45.07 | Mercestes | that reminds me. Can you recommend a PRI failover device for multiple * boxes?? |
| 21:45.12 | n00dle | oh... yeah... have someone email you a fixed /usr/bin/safe_asterisk |
| 21:45.25 | n00dle | That works, too. |
| 21:45.52 | [TK]D-Fender | Mercestes, "acerbic". darn keys too close :) |
| 21:45.53 | Mercestes | http://dictionary.reference.com/browse/ascerbic |
| 21:45.54 | r0d3nt | Mercestes: a good telco provider can set PRI's to fail over to other termination equipment... |
| 21:45.56 | Mercestes | Ahh |
| 21:46.16 | Mercestes | r0d3nt, I have that already at a Colo...now I want to split them with failover. |
| 21:47.00 | r0d3nt | ... |
| 21:47.42 | [TK]D-Fender | Mercestes, For this go call a local telco interconnector. This is the sort of stuff they have in their circles, not ours. |
| 21:48.13 | Mercestes | Okies. |
| 21:48.14 | Mercestes | :) |
| 21:48.20 | Mercestes | Found a page |
| 21:48.45 | n00dle | Ok, I've finally gotten SLA working... now I need to find out why my card won't pick up the line properly on incoming call. |
| 21:49.05 | n00dle | ...after I go install two DSLs... gah! I need 3 more of me! |
| 21:50.22 | joebob777as7 | I got past that! sweet! now i get this cool error!?! |
| 21:50.24 | joebob777as7 | http://paste.ubuntu-nl.org/22491/ |
| 21:51.02 | *** join/#asterisk ploieel (n=manni@Fb2c9.f.ppp-pool.de) |
| 21:51.41 | [TK]D-Fender | joebob777as7, Thats FOP (Flash Operator Panel). Looks like its expected and not installed |
| 21:51.55 | [TK]D-Fender | joebob777as7, Keep in mind : |
| 21:51.58 | [TK]D-Fender | ~freepbx |
| 21:52.14 | jbot | i guess freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:52.28 | [TK]D-Fender | And * is dying. |
| 21:53.11 | *** join/#asterisk greekguy8888 (n=alex@c-24-91-191-77.hsd1.ma.comcast.net) |
| 21:53.19 | greekguy8888 | hi all |
| 21:53.28 | greekguy8888 | any experts here? |
| 21:54.37 | *** join/#asterisk alrs (n=lars@170.206.224.58) |
| 21:55.08 | [TK]D-Fender | greekguy8888, sure... my kniotting skills are unparalleled! |
| 21:55.19 | greekguy8888 | lol too funny! :) |
| 21:55.35 | greekguy8888 | i'm having a very strange issue with 1.4.2 and 1.4.4 |
| 21:55.48 | joebob777as7 | greekguy8888, prove it! |
| 21:55.59 | greekguy8888 | [May 25 11:16:50] WARNING[18156]: chan_sip.c:11843 handle_response_invite: Received response: "Forbidden" from '"asterisk" <sip:asterisk@63.131.149.30>;tag=as6881dbf3' |
| 21:55.59 | greekguy8888 | <PROTECTED> |
| 21:55.59 | greekguy8888 | [May 25 11:16:50] WARNING[18458]: cdr.c:509 ast_cdr_disposition: Cause not handled |
| 21:55.59 | greekguy8888 | [May 25 11:16:50] NOTICE[18458]: pbx_spool.c:341 attempt_thread: Call failed to go through, reason 8 |
| 21:55.59 | greekguy8888 | Really destroying SIP dialog '52f2e21f2778946c620d175b5e9aea21@63.131.149.30' Method: INVITE |
| 21:56.25 | joebob777as7 | [TK]D-Fender, is it a matter of installing FOP? |
| 21:56.30 | greekguy8888 | i have NEVER seen this before, same config worked fine on centos 4.x and then i moved it to another machine running RHEL 4 |
| 21:57.41 | [TK]D-Fender | <PROTECTED> |
| 21:58.47 | *** part/#asterisk ploieel (n=manni@Fb2c9.f.ppp-pool.de) |
| 22:01.23 | greekguy8888 | so no one has seen my errors b4? |
| 22:01.49 | *** join/#asterisk stevej (n=stevej@mail.joneslinux.com) |
| 22:03.05 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:07.32 | *** join/#asterisk angom (n=angom@red-corp-201.143.54.251.telnor.net) |
| 22:07.42 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 22:20.27 | irule | is it a good or bad idea to allow=all codecs in sip.conf? |
| 22:21.42 | blitzrage | bad |
| 22:21.45 | blitzrage | disallow=all |
| 22:21.50 | blitzrage | allow=ulaw |
| 22:21.52 | blitzrage | allow=gsm |
| 22:21.52 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 22:21.54 | blitzrage | etc... |
| 22:23.01 | IOscanner | I am having an issue hearing tdm messages. I see asterisk get the 183 but I can't hear the message. I can call from a land-line or cell and it is fine. |
| 22:23.14 | IOscanner | I turnned on progressinand=yes |
| 22:23.22 | IOscanner | and removed r from the dial string. |
| 22:23.27 | IOscanner | Anything I am missing? |
| 22:27.02 | greekguy8888 | any asterisk experts on? |
| 22:29.06 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 22:31.54 | *** join/#asterisk fujin (i=aj@unaffiliated/fujin) |
| 22:38.22 | *** join/#asterisk stridernzl (n=neville@222-155-180-33.jetstream.xtra.co.nz) |
| 22:39.28 | crimethinker | They're all at the pub drinking |
| 22:41.28 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 22:41.53 | diclophis-work | wtf is the point of not requiring the full file name in AGIs STREAM FILE command |
| 22:42.06 | diclophis-work | why must the file extension be excluded from the path name |
| 22:42.36 | diclophis-work | and surely there must be a way to fix that without recompling |
| 22:45.04 | [TK]D-Fender | <PROTECTED> |
| 22:45.18 | diclophis-work | whats the reasoning? |
| 22:45.30 | [TK]D-Fender | diclophis-work, So you can play back the most native recorded version of whatever you wish to play back if you have multiple encoded versions. |
| 22:45.31 | diclophis-work | i suppose if there are 2 media streams, like a video and audio one |
| 22:45.42 | diclophis-work | mm |
| 22:46.05 | diclophis-work | but how much does the nativity of the recordings codec really effect things? |
| 22:46.13 | [TK]D-Fender | diclophis-work, Lets say your channel is G.729. * would look for a ".g729" version of the file first so as not to waste effort transcoding. |
| 22:46.21 | diclophis-work | mmm |
| 22:46.29 | [TK]D-Fender | diclophis-work, or take up a LICENSE |
| 22:47.08 | diclophis-work | yea |
| 22:47.10 | diclophis-work | i reckon |
| 22:47.53 | diclophis-work | but like what if you know what the exact filename is that asterisk should attempt to play |
| 22:48.05 | diclophis-work | and you dont care about it "looking" for anything else |
| 22:48.09 | diclophis-work | cause there wont ever be anything else |
| 22:48.39 | diclophis-work | surely there should be a way to modify the behavour |
| 22:50.49 | diclophis-work | like some simple logic like, if there is a . within the last 4 characters of the filename, dont look for anything else |
| 22:51.34 | [TK]D-Fender | diclophis-work, how is this a problem for you? |
| 22:51.49 | diclophis-work | well, i am storing filenames in a db |
| 22:51.51 | [TK]D-Fender | diclophis-work, it looks for the current codec first and then some sort of pecking order for the rest. |
| 22:52.02 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 22:52.07 | diclophis-work | so that means i either store them without an extensions, and then wherever i need to reference them outside of asterisk |
| 22:52.14 | diclophis-work | i need to come up with a guessing strategy of my own |
| 22:52.32 | [TK]D-Fender | diclophis-work, More like you store the exact name, and parse off the extension |
| 22:52.33 | diclophis-work | or i store them with an extension, and when i reference them in asterisk, i remove the extebnsion (via some other guesing mechanism) |
| 22:52.54 | diclophis-work | but i reference them more in asterisk than i do outside of asterisk |
| 22:53.08 | diclophis-work | well maybe not the user created ones |
| 22:53.15 | diclophis-work | i guess that makes sense |
| 22:53.22 | diclophis-work | i could just make a getter method |
| 22:53.41 | [TK]D-Fender | diclophis-work, You are neurosing WAY too much over this... |
| 22:53.48 | diclophis-work | well yea |
| 22:53.52 | lee_is_me | hi all, I have a zap question. Is there a settings which controls if the line is hung up after a timeout? I have a customer who places customer on hold and then call is dropped. |
| 22:54.11 | diclophis-work | lee_is_me: there is a general timeout |
| 22:54.28 | diclophis-work | er absolute timeout option |
| 22:54.45 | diclophis-work | Set(TIMEOUT(absolute)=3600) in your dialplan will force nuke the call after an hour |
| 22:54.53 | *** join/#asterisk Laureano (i=[U2FsdGV@OL155-33.fibertel.com.ar) |
| 22:55.02 | diclophis-work | [TK]D-Fender: still, it should be an option, or at the very least an ifdef |
| 22:55.08 | diclophis-work | in the source |
| 22:55.15 | lee_is_me | diclophis-work: thanks. and that would cause a call to be dropped if place on hold? |
| 22:55.29 | diclophis-work | cause i am sure theres no way to just hop into the source and remove the "looking for codec" stuff |
| 22:55.43 | diclophis-work | lee_is_me: iirc |
| 22:56.05 | diclophis-work | theres also a "response", and "digit" timeouts |
| 22:56.13 | diclophis-work | though i am not exactly clear on how they operate |
| 22:56.30 | lee_is_me | diclophis-work: cool. I knew about that setting, but thought it didn't apply to calls on hold... |
| 22:56.44 | lee_is_me | diclophis-work: I'll track those down. Thanks again. |
| 22:58.10 | diclophis-work | you could maybe increase the timeout before going into hold? |
| 22:58.10 | lee_is_me | I guess I could, although this customer's setup is pretty simple. |
| 22:58.22 | lee_is_me | I'll make sure its set to an hour to start... |
| 22:58.46 | diclophis-work | the absolute timeout has saved me more than a couple times, esp after my agi stuff locks up |
| 22:59.50 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:00.20 | *** part/#asterisk danp (i=danp@elmer.glueless.net) |
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| 23:04.13 | diclophis-work | something tells me asterisk isnt gonna like gsm data in a .wav file |
| 23:07.54 | Laureano | BRB, |
| 23:08.21 | *** join/#asterisk Laureano (n=chatzill@OL155-33.fibertel.com.ar) |
| 23:08.53 | diclophis-work | thanks for your time [TK]D-Fender |
| 23:20.55 | IOscanner | anyone know how to get asterisk to allow us to hear the TDM messages from the carrier? 180 and 183? |
| 23:20.59 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 23:21.04 | IOscanner | I tried progressinband=yes |
| 23:21.24 | IOscanner | even removed the r and it still doesn't work. What am I missing? |
| 23:22.35 | CrazyTux | Hello everyone, have a few questions with * and voicemail, and the .txt files found along with the .wav files, I'm wondering if there is a better method such as a database to store that information found in the .txt files? |
| 23:25.01 | lee_is_me | <Continuing on zap lines being dropped while on hold> I checked out TIMEOUT which sets the absolute timeout. That value is actually set to 0, disabled. No drops calls except one putting the caller on hold. Can anyone offer further suggestion/s? |
| 23:25.34 | lee_is_me | To clarify: only calls that are placed on hold get dropped... |
| 23:31.20 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 23:31.20 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits, #astridevcon -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 23:35.52 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 23:44.04 | *** join/#asterisk csd-199 (n=adsf@189.158.190.64) |
| 23:44.56 | csd-199 | Hi. I want a virtual fax on my asterisk server, I know there are some option, but I want an advice on an easy to configure virtual fax |
| 23:45.20 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 23:46.53 | LeddyHM | We use iaxmodem and hylafax |
| 23:48.43 | csd-199 | what about asterfax? |
| 23:56.53 | LeddyHM | never heard of it |