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| 01:55.09 | Nemus | So I am using asterisk 11 and realtime queues I setup to pull from a mysql database I set the time out but the queue keeps cycling over and over again and doesn't fall back to the context in extensions.conf |
| 01:57.41 | Nemus | so basicly the realtime queue doesn't fallout of the queue after the set timeout. |
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| 06:56.13 | swiftkey | hello there! |
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| 06:58.33 | ChannelZ | ahoyhoy |
| 06:58.48 | swiftkey | I want to start a simple pabx system using my computer and want to test it like 4 phones in our house |
| 06:59.02 | swiftkey | any suggestions on where to start ? |
| 06:59.30 | swiftkey | each of them has there local number |
| 06:59.38 | swiftkey | i hope this is the right place to ask |
| 06:59.48 | swiftkey | or hope im asking the right questions ;/ |
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| 07:09.59 | kaldemar | ~book |
| 07:09.59 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 07:10.08 | kaldemar | swiftkey: ^ is a good start |
| 07:10.33 | swiftkey | ops i got dc |
| 07:10.39 | swiftkey | * Blashyrkh (~michaeld@043-054-094-081.as39912.net) has joined #asterisk |
| 07:10.39 | swiftkey | * mirela666 (~mirko.bra@109-93-2-133.dynamic.isp.telekom.rs) Quit (Ping timeout: 256 seconds) |
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| 07:15.02 | swiftkey | as you where saying ? |
| 07:15.52 | kaldemar | infobot gave you a link to a book that is a good start. did you get that? |
| 07:15.53 | infobot | okay, kaldemar |
| 07:16.21 | kaldemar | oh come on... |
| 07:16.34 | kaldemar | ~gave |
| 07:16.45 | kaldemar | ~gave you a link to a book that |
| 07:16.45 | infobot | well, gave you a link to a book that is a good start. did you get that? |
| 07:18.19 | kaldemar | infobot: no, gave you a link to a book that is |
| 07:18.26 | kaldemar | ~gave you a link to a book that |
| 07:18.27 | infobot | hmm... gave you a link to a book that is a good start. did you get that? |
| 07:18.52 | kaldemar | infobot: no, gave you a link to a book that is |
| 07:18.57 | kaldemar | ~gave you a link to a book that |
| 07:18.57 | infobot | extra, extra, read all about it, gave you a link to a book that is a good start. did you get that? |
| 07:19.11 | kaldemar | stupid bot... |
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| 08:01.43 | KevinBMontague | anyone around that might be able to help with a voicemail question? |
| 08:01.52 | ChannelZ | f*ing comcast |
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| 08:04.51 | ChannelZ | KevinBMontague: ask |
| 08:07.59 | KevinBMontague | thanks CZ. I'm trying to figure out when/where the ${VM_MESSAGEFILE} variable is being set. I'm running a shell script for externnotify to grab the mailbox id and number of new messages left in a mailbox and pass that to a PHP script which creates a call file to set off an outbound notification call. However, within the outbound dialplan the variable is always retruning empty in NoOP. Im not sure where to grab the proper file location f |
| 08:19.54 | ChannelZ | well I think that's a channel variable and not something exported to the script |
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| 08:23.49 | KevinBMontague | Yeah, I saw that and thats a problem. How would the callflow know what box the message was for then? Where is it actually being set when a message is left in a mailbox? Do you know of any other programmatic ways (maybe PHP) to possibly take a passed mailbox name and find the file path/name to be able to use that in a callflow? |
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| 08:32.13 | ChannelZ | It gets passed arguments.. [context] [extension] [new voicemails] [old voicemails] [urgent voicemails] |
| 08:33.21 | ChannelZ | do you need the actual voicemail filename for some reason? |
| 08:33.51 | ChannelZ | I mean you could use VM_MESSAGEFILE in your dialplan after the fact but that's obviously outside of externnotify |
| 08:34.31 | KevinBMontague | yes and i grab those and pass them to the php script which creates the callfile, which places a call to LOCAL/s@myroutine to start the notification process. But in the notification process the messagefile variable is always empty and I'm not sure how to get the file name info into the dialplan at that point. |
| 08:34.54 | KevinBMontague | yes I need the filename to be able to email it as an attachement from within the callflow |
| 08:37.15 | KevinBMontague | the default asterisk email notification wont work for this use |
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| 09:36.07 | Rico | hi there |
| 09:36.18 | Rico | I have a problem with cdr odbc backend |
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| 09:37.19 | Rico | http://pastebin.com/rZ1TzbCL |
| 09:37.42 | Rico | I don't know why odbc can not write in cdr table |
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| 10:27.29 | skrusty | is there an irc channel for #asterisk-app-dev? |
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| 11:40.36 | StaRetji | Howdy |
| 11:43.18 | StaRetji | folks, I had to pass exten => _X.,n,Set(CALLERID(num)=${CALLERID(dnid)}) to a2billing in order to bill did forwarding to mobile phone. Now, I have problem, mobile phone sees dnid as callerid. Can someone suggest workaround, can I pass rdnis to a2billing too? |
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| 12:59.56 | yebyen | any fop2 users here that ever tried bridging a live "listen" into a conference/meetme? |
| 13:00.08 | yebyen | or without fop2 |
| 13:00.32 | yebyen | i want to give one of our clients an opportunity to remote monitor for a day |
| 13:00.58 | Greenlight | Just orginate a Local channel between them both |
| 13:01.35 | yebyen | where do I start? |
| 13:01.54 | Greenlight | DO you have a live listen extension ? |
| 13:02.21 | yebyen | we actually have this FOP2 panel that lets me log in, click on the extension I want to monitor, press listen and my phone rings |
| 13:02.32 | Greenlight | shrugs |
| 13:02.36 | yebyen | so it's possibly already doing that |
| 13:02.37 | Greenlight | Never used FOP2 sorry |
| 13:03.09 | yebyen | so in the asterisk console, channel originate |
| 13:03.46 | Greenlight | Yea, you'll need to originate a call between your "live listen" (presumably a ChanSpy) extension and your "conference" extension |
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| 13:04.39 | yebyen | alright lets give this a shot |
| 13:06.12 | yebyen | Launching ChanSpy(DAHDI/70,qsv(2)b) on DAHDI/82-1 |
| 13:06.18 | yebyen | Spying on channel DAHDI/70-1 |
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| 13:06.52 | yebyen | then I see app_chanspy send NOTICE that it's bridging those two channels and also 82 (my seat) to DAHDI/25-1 |
| 13:07.09 | yebyen | can I originate a channel between my dahdi channel and the conference extension? |
| 13:07.36 | Greenlight | Umm |
| 13:07.45 | Greenlight | Originate will *create* the channels |
| 13:07.52 | yebyen | so i need extensions |
| 13:07.55 | Greenlight | Exactly |
| 13:08.43 | Greenlight | ANd you'd use the Local "loopback" channel driver to join two extensions |
| 13:09.16 | Greenlight | So, say your "listen" extension was 100 and your "conference" extension was "200", you'd originate a call from Local/100 to 200 |
| 13:09.27 | yebyen | so, I don't want channel originate DAHDI/82-1 770 |
| 13:09.33 | yebyen | if 770 is my conference extension |
| 13:10.40 | Greenlight | And what's your ChanSpy extension ? |
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| 13:12.19 | yebyen | that's what I don't know |
| 13:12.25 | yebyen | i think it's just an "application" |
| 13:12.31 | yebyen | i'm not dialing into it |
| 13:12.33 | Greenlight | Ahh okay well we can work with that too |
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| 13:14.57 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:15.19 | yebyen | i feel like i'm halfway to understanding this... |
| 13:15.22 | yebyen | channel originate DAHDI/82-1 application ChanSpy(DAHDI/70,qsv(2)b) |
| 13:15.33 | yebyen | so that's doing the same thing as clicking on the Listen button |
| 13:15.52 | yebyen | rings my desk channel and connects it to the ChanSpy |
| 13:16.07 | yebyen | i want to connect the application to the conference bridge |
| 13:16.15 | yebyen | do I need to put it on a channel first? |
| 13:17.00 | yebyen | i tried originating to the channel while I was already on the listen app |
| 13:17.02 | *** join/#asterisk sframe (~sframe@107.7.137.66) |
| 13:17.06 | yebyen | and it told me the channel wasn't available |
| 13:17.54 | yebyen | channel redirect! |
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| 13:19.21 | yebyen | nope |
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| 13:23.18 | Greenlight | It may be easier to create an extension to do the chanspy bit |
| 13:23.39 | Greenlight | I'm more used to the AMI Origiante syntax, however |
| 13:24.03 | yebyen | so far I have channel redirect DAHDI/82-1 meetme,770,1 |
| 13:24.09 | Greenlight | channel originate Local/770 application ChanSpy <CHANNEL TO SPY ON> |
| 13:24.22 | Greenlight | Whatcha redirecting for? |
| 13:24.31 | yebyen | which seems to stop what i'm doing and put me in the conference room |
| 13:24.43 | Greenlight | eg it "redirects" you.. |
| 13:24.53 | yebyen | and channel originate DAHDI/82-1 extension 770@meetme |
| 13:25.09 | yebyen | which will do it even if i'm not on the line... |
| 13:25.14 | yebyen | and ring my phone |
| 13:25.14 | Greenlight | What is DAHDI/82-1 ? |
| 13:25.17 | yebyen | that's my desk |
| 13:25.44 | Greenlight | I thought we were connecting a conference and a live listen |
| 13:25.53 | yebyen | yeah, i can connect the live listen to my desk |
| 13:25.57 | Greenlight | Where does your desk come into it |
| 13:25.58 | yebyen | but i get the idea that's doing it backward |
| 13:26.07 | yebyen | i can dial my desk into the conference if everything works :D |
| 13:26.26 | yebyen | the problem is i have to connect a channel to an application or an extension |
| 13:26.41 | Greenlight | Yea, the channel being "Local/770" |
| 13:26.45 | yebyen | and I am not sure how to get either chanspy or meetme on a channel without connecting it to my desk |
| 13:26.58 | yebyen | they're both applications |
| 13:27.25 | Greenlight | Fine. |
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| 13:27.59 | yebyen | ok |
| 13:28.02 | yebyen | i needed to say context |
| 13:28.07 | yebyen | i think |
| 13:28.47 | Greenlight | Easiest thing to do is add a new extension to your dialplan, which does the chanspy. |
| 13:28.49 | yebyen | now it's trying, but i'm not calling chanspy with the right syntax |
| 13:29.13 | yebyen | the problem is there are half a dozen reps calling this campaign and we need to potentially switch between them |
| 13:29.15 | Greenlight | Also, which context is "770" in ? |
| 13:29.17 | yebyen | and they're not a queue |
| 13:29.19 | yebyen | @meetme |
| 13:29.24 | yebyen | so I have |
| 13:29.37 | Greenlight | SO, Local/770@meetme/n |
| 13:29.41 | yebyen | "channel originate Local/770@meetme application ChanSpy(DAHDI/70,qsv(2)b) |
| 13:29.49 | yebyen | what's /n? |
| 13:29.59 | yebyen | it doesn't like this syntax for calling ChanSpy either |
| 13:29.59 | Greenlight | You can omit that, just habit |
| 13:30.12 | yebyen | No such application 'ChanSpy(DAHDI/70,qsv(2)b)' |
| 13:30.14 | Greenlight | Yea, that's not the right syntax, you'll be easier putting that into your dialplan |
| 13:30.31 | Greenlight | channel originate Local/770@meetme application ChanSpy DAHDI/70,qsv(2)b |
| 13:30.33 | Greenlight | Try that |
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| 13:33.13 | yebyen | looks like the problem is that the conference bridge needs you to say your name, press pound, press 1 |
| 13:33.21 | yebyen | i think it is connecting |
| 13:34.23 | Greenlight | Right, well obviously you'll need to change that. |
| 13:35.19 | yebyen | actually it did fall through and connect it |
| 13:35.59 | yebyen | now i have chanspy on the conference extension and I can't hang it up |
| 13:36.12 | yebyen | it worked for about 30 seconds |
| 13:36.25 | yebyen | until it stopped spying (?) |
| 13:36.45 | yebyen | hard to know what's going on when there are people filling up my console logs and I have to run back and forth |
| 13:37.09 | Greenlight | I'm not 100% sure if ChanySpy answers |
| 13:37.24 | yebyen | it definitely put chanspy on the conference bridge |
| 13:37.28 | Greenlight | I would recommend moving the ChanSpy to an extension in your dialplan |
| 13:37.31 | yebyen | i can make calls now out from the spied station |
| 13:37.53 | yebyen | we have a jerk client who wants to do it today, i need instructions to him by noon |
| 13:38.03 | yebyen | the client isn't the jerk it's the csr that told him he could have his way |
| 13:38.13 | SuperNull | gotta love ignorant sales. |
| 13:38.18 | SuperNull | our sales department is horrible. |
| 13:38.30 | yebyen | "He wants to do it today" |
| 13:38.32 | SuperNull | they used to sell 25megabit cable packages when it would never happen |
| 13:38.34 | yebyen | "We have never done that" |
| 13:38.39 | yebyen | "OK, I'll tell him tomorrow" |
| 13:39.03 | yebyen | how do I shutdown a meetme extension |
| 13:39.34 | Greenlight | channel request hangup <Channel> |
| 13:40.12 | yebyen | great, the channel number is long gone from my screen :D |
| 13:41.02 | Greenlight | channel request hangup Local/770 (Now press tab twice) |
| 13:41.37 | yebyen | oh dude! |
| 13:41.40 | yebyen | lifesaver |
| 13:41.42 | yebyen | tab completion |
| 13:41.57 | Greenlight | Yup :) |
| 13:43.08 | yebyen | i think i have everything I need |
| 13:43.39 | yebyen | i would ask for a bitcoin address to say thank you but my employer doesn't keep bitcoins :\ |
| 13:43.42 | yebyen | lol |
| 13:43.48 | Greenlight | heh |
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| 13:57.17 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:57.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:10.56 | StaRetji | if I set callerid on my asterisk and pass it to 2nd asterisk can 2nd asterisk still get rdnis from original caller? |
| 14:11.15 | StaRetji | Is there a way to achieve this? |
| 14:12.04 | Penguin | What channel technology are you using between the two asterisks? |
| 14:12.25 | StaRetji | SIP |
| 14:12.42 | Penguin | I've never seen SIP carry RDNIS anyway. |
| 14:12.43 | StaRetji | 1st asterisk has: exten => _X.,n,Set(CALLERID(num)=${CALLERID(dnid)}) |
| 14:13.04 | StaRetji | so, second one carry dnid as callerid |
| 14:13.18 | StaRetji | I had to do it, cause I need to bill DID forwarding to mobile phone |
| 14:13.27 | StaRetji | and there is no other way to authenticate it |
| 14:13.33 | StaRetji | on a2billing |
| 14:14.10 | Penguin | If you set the callerID to any arbitrary value, that's what it will be when it arrives at the other asterisk. Doesn't matter if the value is dnid or something else. |
| 14:14.20 | Penguin | But I have never seen RDNIS present on SIP calls. |
| 14:14.21 | StaRetji | got it |
| 14:14.48 | StaRetji | ok, in that case, I can't do anything, 2nd asterisk get callerid from 1st |
| 14:14.54 | StaRetji | and story ends there, right? |
| 14:15.20 | Penguin | There are some callerID fields that you can shove data into and retrieve on the other asterisk. Maybe that will help. |
| 14:15.21 | StaRetji | I get original callerid on 1st asterisk, from the caller |
| 14:15.34 | StaRetji | but that one is unknown to a2billing |
| 14:15.43 | Penguin | The original callerid can go on to the second asterisk. |
| 14:15.48 | [TK]D-Fender | that is an a2billing issue then |
| 14:15.49 | StaRetji | hm |
| 14:16.15 | StaRetji | yes, it is [TK]D-Fender, but I am trying to solve it in extensions.conf of [did] context |
| 14:16.39 | [TK]D-Fender | you can pass whatever you want from A to B via extra SIP headers |
| 14:16.42 | StaRetji | Penguin: thx, I am trying this since yesterday, googling for examples, but I fail |
| 14:16.44 | [TK]D-Fender | so go put it in there. |
| 14:17.46 | StaRetji | ok, googling extra SIP headers :) thx |
| 14:18.56 | [TK]D-Fender | StaRetji: "core show application SIPAddheader", "core show function SIP_HEADER" |
| 14:33.13 | StaRetji | exten => _X.,n,Set(CALLERID(num)=${CALLERID(dnid)}) |
| 14:33.13 | StaRetji | exten => _X.,n,SipAddHeader(P-Asserted-Identity: <sip:${CALLERID(num)}>) |
| 14:33.33 | StaRetji | sends changed callerid to 2nd asterisk |
| 14:34.35 | StaRetji | I am not sure if I can do anything bcs once it goes to a2billing, I can't modify it |
| 14:38.56 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:38.57 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 15:26.31 | ghost75 | is chan_mobile working only with few phones in fxo mode or most of phones? |
| 15:28.37 | *** join/#asterisk aberrios (~aberrios@195.130.201.200) |
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| 15:51.49 | *** join/#asterisk magespawn (~Eames@105-236-71-217.access.mtnbusiness.co.za) |
| 15:51.57 | magespawn | howdy all |
| 15:52.09 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
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| 15:53.29 | magespawn | i am trying to have two ip door phone call a single extension in an asterisk pbx |
| 15:53.52 | Penguin | And the problem? |
| 15:54.12 | yebyen | Greenlight: any idea why it would connect and work perfectly for 10 seconds, then cut out with no related messages in asterisk console? |
| 15:54.20 | magespawn | the door phone call the main switchboard not the extension i want |
| 15:54.42 | Penguin | How does the door phone enter the extension? |
| 15:54.45 | Penguin | hotline? |
| 15:55.11 | Greenlight | yebyen: As I said earlier, if ChanSpy *doesn't* answer, then it may think it's still "ringing" |
| 15:55.19 | magespawn | single button, programmed in the door phone itself to call the extension |
| 15:55.35 | yebyen | Greenlight: ah, great, that makes sense |
| 15:55.46 | Greenlight | That would be my hunch |
| 15:56.01 | yebyen | so originate needs an answer |
| 15:56.03 | Penguin | magespawn: Change the extension that the button dials. |
| 15:56.26 | yebyen | and i would need to answer on that Local/770 |
| 15:56.32 | Greenlight | yebyen: Yea, I'd recommend putting it in the dialplan so you have more control, rather than origianting directly to the application |
| 15:56.47 | magespawn | i have but the asterisk seems to overide that setting and dial the main switchboard |
| 15:56.51 | Greenlight | I would *assume* that 770 already answers |
| 15:57.18 | Greenlight | magespawn: Lets see a sip trace of a call, to be sure |
| 15:57.19 | Penguin | magespawn: Asterisk takes whatever extension your phone is calling. |
| 15:57.41 | magespawn | Greenlight how do i do that? |
| 15:57.44 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 15:58.35 | Greenlight | https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 15:58.45 | magespawn | ty |
| 15:59.21 | yebyen | Greenlight: is that what /n is for? |
| 15:59.40 | Greenlight | It prevents local channel optimisation |
| 15:59.49 | Greenlight | Which in this case won't happen anyway |
| 15:59.55 | yebyen | hm |
| 16:00.37 | yebyen | maybe i need more -vvv's |
| 16:01.41 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 16:01.55 | *** join/#asterisk boom^time (~boom^time@75-151-20-174-Michigan.hfc.comcastbusiness.net) |
| 16:02.35 | yebyen | now this time it did not hang up |
| 16:02.42 | Penguin | If you're already on the asterisk console, you change the verbose level with core set verbose <level>. |
| 16:03.20 | Greenlight | yebyen: What did you do differently ? |
| 16:03.42 | Penguin | Anything useful is seen at core set verbose 3 or higher. |
| 16:04.18 | Greenlight | Hmm not seen this before: |
| 16:04.20 | Greenlight | [Oct 24 16:52:50] ERROR[1998]: res_rtp_asterisk.c:2229 ast_rtcp_write_rr: RTCP RR transmission error, rtcp halted: Invalid argument |
| 16:05.36 | [TK]D-Fender | [11:56]magespawni have but the asterisk seems to overide that setting and dial the main switchboard <- Asterisk will process whatever number that device sends.... |
| 16:06.02 | yebyen | Greenlight: nothing... it seems to hang up after a random time, not always 10 seconds |
| 16:06.06 | yebyen | sometimes I listen to the end of the call |
| 16:06.12 | yebyen | common thread: i see this |
| 16:06.13 | yebyen | Channel will hangup at 2013-10-24 16:05:23.333 EDT. |
| 16:07.17 | Greenlight | Very odd.. channels shouldn't just randomly hangup.. |
| 16:08.04 | Penguin | That is probably caused by a TIMEOUT() setting in the dial plan. |
| 16:08.27 | Greenlight | But, why would it be random |
| 16:08.34 | Penguin | It isn't random. |
| 16:08.47 | Greenlight | As in the length of time |
| 16:08.58 | Greenlight | If it was a timeout, I'd expect the time interval to be constant |
| 16:09.14 | Penguin | Find the TIMEOUT(). Remove or change it. |
| 16:09.48 | magespawn | Greenlight [TK]D-Fender i have just set the door phone to call another extension and it still dials the main switchboard |
| 16:09.57 | Greenlight | SHOW US |
| 16:10.01 | [TK]D-Fender | magespawn: Show us the call |
| 16:11.01 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:12.27 | magespawn | just a bit of history, i have inherited this setup from another tech, so i am a bit in the deep end here |
| 16:12.47 | yebyen | i don't know what's timing out, would it be chanspy? |
| 16:12.57 | Penguin | TIMEOUT() |
| 16:13.26 | Penguin | More specifically, TIMEOUT(absolute). |
| 16:13.29 | Greenlight | yebyen: You should start by doing as I recommended and moving the ChanSpy to the dialplan |
| 16:14.04 | yebyen | ok, I'll make the meetme connect the ChanSpy as soon as you dial in. |
| 16:15.15 | Greenlight | Ensure that you Answer() before the ChanSpy |
| 16:15.18 | [TK]D-Fender | magespawn: History on back-burner. call debug now. |
| 16:17.49 | Penguin | Does a four-hour timeout really cause a problem that often? |
| 16:18.07 | yebyen | MeetMe(...);Answer();ChanSpy(DAHDI/53,...); |
| 16:18.29 | yebyen | ChanSpy does not seem to connect to the channel |
| 16:18.38 | Greenlight | WHat |
| 16:18.44 | Greenlight | Why do you have MeetMe in there |
| 16:18.55 | yebyen | don't need it? |
| 16:18.58 | yebyen | it's a meetme conference line |
| 16:19.11 | Greenlight | You have that at extension 770 |
| 16:19.19 | *** join/#asterisk navaismo (~navaismo@189.241.6.52) |
| 16:19.34 | Greenlight | You're going to use a Local channel to bridge your "confernce" 770 extnesion, and the ChanSpy one you're about to create.. |
| 16:19.45 | yebyen | ok |
| 16:19.51 | Greenlight | So, literally all you need to do is answer, and then chanspy |
| 16:20.25 | *** join/#asterisk saxa (~sasa@2-234-206-126.ip224.fastwebnet.it) |
| 16:20.29 | yebyen | so, i'm still doing channel originate from the console |
| 16:20.36 | yebyen | i'm just putting the ChanSpy into my dial plan somewhere |
| 16:20.40 | Greenlight | Yea |
| 16:21.23 | Greenlight | If you want to have a live listen bridged into a conference you'll need to originate that call from somewhere |
| 16:22.33 | magespawn | where does it store the log file? |
| 16:22.44 | magespawn | sorry stupid |
| 16:23.00 | Greenlight | Usually /var/log/asterisk |
| 16:23.02 | Penguin | I don't care about any logs. |
| 16:23.06 | Penguin | Show us the call. |
| 16:27.21 | yebyen | if i'm using _77X in my dialplan |
| 16:27.26 | yebyen | ${EXTEN} gives me 77X |
| 16:27.35 | yebyen | is there a string function I can use or something else for just the X part? |
| 16:27.47 | Penguin | Offset the variable. |
| 16:27.53 | magespawn | okay Penguin, how? |
| 16:28.07 | Penguin | ${EXTEN:2} |
| 16:28.15 | yebyen | yeah |
| 16:28.19 | yebyen | perfect |
| 16:29.07 | StaRetji | <PROTECTED> |
| 16:29.09 | StaRetji | cheers |
| 16:29.42 | *** join/#asterisk jhlavacek (~jirka@80.215.33.75) |
| 16:30.10 | *** join/#asterisk Bkhan (ca7d90e3@gateway/web/freenode/ip.202.125.144.227) |
| 16:30.16 | Bkhan | Hi |
| 16:32.16 | Bkhan | I am using asterisk 11.5. When call in queue and ringing to agent agent status is "not in use" even agent received call still its status is "not in use" |
| 16:32.42 | Bkhan | we are using queues through realtime |
| 16:33.31 | magespawn | Penguin how do I show you the call? |
| 16:33.39 | Penguin | bkhan: Does the device have call counters enabled? |
| 16:33.44 | yebyen | Greenlight: this is working perfectly |
| 16:33.50 | [TK]D-Fender | magespawn: Definitely not a good sign |
| 16:33.54 | Greenlight | yebyen: Glad to hear it :) |
| 16:34.00 | Penguin | magespawn: core set verbose 3, sip set debug on |
| 16:34.01 | [TK]D-Fender | magespawn: asterisk -rvvvvvvvvvvvvvvvvvvvv |
| 16:34.07 | [TK]D-Fender | magespawn: "core set verbose 10" |
| 16:34.15 | [TK]D-Fender | magespawn: "sip set debug on" |
| 16:34.23 | *** join/#asterisk gbit (~gbit@unaffiliated/gbit) |
| 16:34.23 | [TK]D-Fender | magespawn: PASTEBIN the complete call. |
| 16:34.23 | yebyen | Greenlight: i do "channel originate Local/770@meetme extension 203" after I've dialed into the spy extension |
| 16:34.25 | [TK]D-Fender | ~pb |
| 16:34.25 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:34.27 | [TK]D-Fender | ^^^^ |
| 16:34.43 | *** join/#asterisk dxd828 (~dxd828@217.39.7.254) |
| 16:34.45 | magespawn | [TK]D-Fender, thank you, i very mush in the deep end |
| 16:34.47 | Bkhan | Penguin: Please explain. I am using softphone (eye beam) |
| 16:35.01 | Penguin | bkhan: sip.conf |
| 16:35.06 | Penguin | bkhan: The device entry. |
| 16:35.19 | Penguin | bkhan: Call counters. Enable it for your device. |
| 16:41.03 | Bkhan | Penguin: thanks a lot issue has resolved |
| 16:42.02 | *** join/#asterisk davlefouAMD (~david@197.15.206.48) |
| 16:47.32 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 16:50.38 | magespawn | [TK]D-Fender, Penguin http://pastebin.com/JG3F2QBb |
| 16:50.52 | [TK]D-Fender | magespawn: that is no a call attempt |
| 16:50.53 | [TK]D-Fender | not* |
| 16:51.05 | [TK]D-Fender | magespawn: pastebin the ENTIRE call from beginning to end |
| 16:51.14 | [TK]D-Fender | magespawn: there should be hundreds of lines of output for this |
| 16:51.18 | magespawn | okay |
| 16:51.56 | magespawn | okay brb |
| 16:57.02 | magespawn | http://pastebin.com/LFpXtW1w is this right? |
| 16:57.49 | [TK]D-Fender | magespawn: no, that is still at the END of a call... never the START |
| 16:59.09 | Greenlight | magespawn: It might be easier if you use something like tee to get a copy of your CLI output |
| 16:59.35 | magespawn | never heard of that let me google |
| 16:59.44 | magespawn | brb |
| 17:00.38 | Penguin | If you are using PuTTY, you can increase the scrollback lines, enable logging, or both. These things can help get the entire call. |
| 17:08.43 | magespawn | thanks for that Penguin, asterisk is rather new to me, i have some linux experience with ubuntu, so this is a nice learning curve for me |
| 17:09.47 | magespawn | [TK]D-Fender, Penguin I have to go home now, it is 19:30 here but I will be back tomorrow |
| 17:10.20 | magespawn | [TK]D-Fender, Penguin thanks again for the help and the patience. |
| 17:15.02 | yebyen | Greenlight: well, almost perfect... we wound up giving the ChanSpy extensions to our client |
| 17:15.33 | yebyen | Greenlight: when i originate a call from the meetme to the chanspy extension, the same thing happens (10 seconds later, the chanspy is disconnected silently) |
| 17:16.00 | yebyen | it's too late to change anything now, but are you sure I shouldn't be Answer() after I ChanSpy()? |
| 17:16.06 | yebyen | rather than before |
| 17:16.17 | yebyen | or maybe I'm doing it backwards |
| 17:16.58 | Penguin | Does that even make sense to you? |
| 17:17.16 | Penguin | Why would you bring up the line AFTER the call has ended? |
| 17:17.47 | Penguin | You answer the call, then run chanspy. How could it work better the other way? |
| 17:18.11 | yebyen | well if it's ChanSpy that's hanging up, |
| 17:18.26 | yebyen | do you need to answer ChanSpy? |
| 17:18.34 | Penguin | DId you find the timeout yet? |
| 17:18.38 | Penguin | Did you even look for it? |
| 17:18.48 | ghost75 | what could this be: [Oct 24 19:16:41] ERROR[22006]: chan_mobile.c:4102 do_sco_listen: ast_io_wait() failed for audio on adapter blue |
| 17:19.06 | yebyen | I don't see one, I see some other things doing Set(TIMEOUT(...)=number); |
| 17:19.08 | yebyen | but not in this context |
| 17:19.47 | Penguin | I'd have to see the entire dial plan to figure it out, I guess. |
| 17:19.54 | yebyen | you think I could set it to four hours and it would help? |
| 17:20.08 | yebyen | Set(TIMEOUT(response)=3600); |
| 17:20.09 | yebyen | say |
| 17:20.18 | yebyen | after Answer() |
| 17:21.04 | Penguin | You need to find the timeout that is on your call and either fix the value or remove it. |
| 17:21.45 | yebyen | the other thing that bothers me is that when ChanSpy gets connected, the two listeners can't hear each other |
| 17:22.22 | yebyen | i was hoping they would be able to talk about the call |
| 17:24.56 | ghost75 | if i compile * from same source folder as before and just change code from one module, can i just copy this one module after? |
| 17:25.43 | [TK]D-Fender | yes |
| 17:26.34 | ghost75 | lets see how long it compiles :> |
| 17:28.40 | *** join/#asterisk dxd828 (~dxd828@212.183.128.230) |
| 17:33.06 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 17:34.05 | *** join/#asterisk Tokeiito (~quassel@78-62-103-201.static.zebra.lt) |
| 17:43.36 | ghost75 | is there a file spandsp.h to be supposed in source folder? |
| 17:43.59 | [TK]D-Fender | ghost75: that is a separate lib, not part of * itself |
| 17:46.16 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 17:55.18 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:57.36 | boom^time | Does anyone have experience using an alias for cdr_default_odbc.conf? I've set alias src => cid and it doesn't populate my cid column. |
| 17:57.50 | boom^time | not quite sure how to go about debugging it |
| 18:02.22 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 18:08.50 | *** part/#asterisk StaRetji (~LittleAll@178.79.6.17) |
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| 18:11.25 | yebyen | so, this is what I have: http://pastebin.com/VeQpiLDz |
| 18:11.29 | yebyen | rather than make you guess |
| 18:11.46 | yebyen | the bottom are the calls to channel originate and channel request hangup as examples |
| 18:11.59 | yebyen | I am not sure what timeouts might be set as defaults, I don't see any set explicitly |
| 18:12.15 | yebyen | but it's not random, it's about 10 seconds |
| 18:12.39 | yebyen | and it might be that I'm too silent |
| 18:13.04 | ghost75 | hmm mobile search is showing phone but mobile show devices is showing it as not connected |
| 18:13.22 | Penguin | yebyen: Why would you want to put a chanspy into a meetme? |
| 18:13.28 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 18:13.39 | yebyen | Penguin: i want outside person to call and join the meetme |
| 18:13.44 | yebyen | and i connect him with chanspy |
| 18:13.57 | Penguin | Then people in the meetme can spy on a single channel together? |
| 18:13.58 | yebyen | and we disconnect later and talk about the call |
| 18:14.01 | yebyen | right |
| 18:14.37 | yebyen | or talk about the call without being heard by the caller and callee |
| 18:14.43 | yebyen | while it's going on, ideally |
| 18:14.46 | yebyen | not sure if that will work |
| 18:19.18 | yebyen | would you say I'm answering correctly? there must be a timeout set, and it's going to extension t=>? |
| 18:19.38 | *** part/#asterisk gbit (~gbit@unaffiliated/gbit) |
| 18:20.31 | Penguin | Based only on what you've shown, I don't see a reason for timeout to be set. |
| 18:22.04 | Penguin | But the way you put the chanspy into the meetme looks okay to me. |
| 18:24.29 | yebyen | i know that if you dial in with a handset to any of those extensions, you are not disconnected after 10 seconds |
| 18:24.59 | yebyen | only if you originate like it's shown at the bottom |
| 18:25.15 | yebyen | and it doesn't Hangup, it just turns to dead air |
| 18:25.21 | Penguin | Calling to 7757 from a phone does not show the timeout problem? |
| 18:25.32 | yebyen | in other words, i'm not getting back to my music on hold until i channel request hangup the Local/770 |
| 18:25.38 | yebyen | exactly |
| 18:25.41 | Penguin | Calling to 770 from a phone does not show the timeout problem? |
| 18:25.46 | yebyen | right |
| 18:25.55 | yebyen | only when you do originate Local/... |
| 18:26.00 | Penguin | Very weird. |
| 18:26.07 | yebyen | and not every time |
| 18:26.28 | Penguin | Originate it in dialplan instead of CLI and see if it happens ever. |
| 18:29.36 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:29.41 | yebyen | so, i'm making an extension that when I dial, will connect these two extensions with Originate( and let me hang up... |
| 18:30.01 | Penguin | Just to test it, of course. |
| 18:31.10 | yebyen | i'm not sure how to write that... does Originate "background"? |
| 18:31.14 | yebyen | on its own |
| 18:31.59 | Penguin | I don't think so. Just stay on the phone long enough to see if it gives the timeout problem. |
| 18:32.30 | Penguin | Instead of typing the originate command on the CLI, you're going to dial an extension from a phone to do the same thing. |
| 18:32.37 | yebyen | alright, i need a few phones to do this lol |
| 18:32.57 | Penguin | A soft phone on your computer would be handy for that. |
| 18:33.24 | *** join/#asterisk serafie1 (~erin@nat/digium/x-jpbqnwjpocpilvmk) |
| 18:38.05 | yebyen | Originate(Local/770@meetme,exten,7757@meetme); Wait(45); |
| 18:38.22 | yebyen | it seems to be originating in the console but nothing ever joins my conference line |
| 18:39.03 | yebyen | maybe i have the number wrong |
| 18:39.23 | Penguin | Your syntax is wrong. |
| 18:40.09 | *** join/#asterisk dxd828 (~dxd828@host-2-97-72-213.as13285.net) |
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| 18:41.58 | Penguin | Originate(Local/770@meetme,exten,meetme,7757,1) |
| 18:42.45 | ghost75 | i just wonder if port what chan_mobile refers to, is same as channel in rfcomm |
| 18:44.54 | drmessano | ghost75, port = port |
| 18:46.59 | yebyen | Penguin: originating the call in the dial plan seems to help, i can also hang up the line that calls the originate extension |
| 18:47.39 | yebyen | i am not the least bit sure how to turn this into a usable thing instead of puppetmaster |
| 18:49.00 | yebyen | ohp... maybe not |
| 18:49.15 | Penguin | Originating from the dial plan was more to test for that timeout problem rather than to use it regularly. |
| 18:49.23 | yebyen | i think that first call didn't last long enough to see it time out |
| 18:49.37 | yebyen | one second into the next call it cut out again |
| 18:50.17 | Penguin | When a channel has a timeout set, you usually see that on the console. |
| 18:50.25 | Penguin | (like you saw earlier) |
| 18:50.37 | yebyen | everything's moving very fast, i'm not sure that timeout message I saw was from my channel |
| 18:50.46 | yebyen | this will be easier to test at night |
| 18:50.52 | yebyen | but there won't be anyone to spy on |
| 18:50.53 | Penguin | That's an important bit. |
| 18:52.07 | ghost75 | aha phone is connected :> |
| 18:52.31 | Penguin | If that timeout wasn't for THIS call, then we're chasing ghosts. |
| 18:52.45 | ghost75 | runs |
| 18:53.12 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 18:53.39 | admin0 | hi guys .. i want to use asterisk just for transcoding purpose only .. for this, will a compiled one be better to achieve good performance ? |
| 18:53.49 | admin0 | what determines the time/quality of transcoding |
| 18:53.57 | admin0 | is it the way its compiled or the CPU it runs on ? |
| 18:54.01 | Penguin | "compiled one" ? |
| 18:54.40 | admin0 | what modules might I need .. there will be 2 sip endpoints .. IN and OUT .. in accepts all codecs, and in OUT just one codec will go .. |
| 18:54.50 | drmessano | 2 endpoints? |
| 18:54.52 | admin0 | so for this bare minimum purpose, what modules/flags might I need to compile |
| 18:54.55 | admin0 | peers |
| 18:55.15 | drmessano | How many concurrent calls? |
| 18:55.37 | admin0 | lets say 50 |
| 18:56.21 | admin0 | i am going to use a server at hetzner .. Intel(R) Core(TM) i7-2600 CPU @ 3.40GHz x 8 threads , |
| 18:56.26 | drmessano | I always recommend building from source, but a binary install is fine with 50 calls. There's no room to really optimize anything |
| 18:56.47 | drmessano | If you said 5000 or something then yeah |
| 18:57.28 | admin0 | well , if binary vs compiled improves just the concurrent calls and nothing else, then binary is fine .. i was thinking more like if a compiled one shows lower times if i do show codec translation recall |
| 18:57.31 | admin0 | recalc |
| 18:57.38 | Penguin | What makes one that you compile yourself better than one Qwell compiled and put into the repo? |
| 18:58.34 | boom^time | Any good way to prepend the unique_id of all channels of a specific server? |
| 18:58.46 | admin0 | that is my tought also .. that community is already doing a good job .. my question was if during compile i can exclude anything that might improve transcoding times |
| 18:58.49 | boom^time | so multiple servers hitting the same db have no chance of a conflict |
| 18:59.48 | boom^time | nm, there is an option in asterisk.conf. |
| 19:00.41 | drmessano | admin0, you may find stripping things down improves some performance, but then again, you're talking about 50 concurrent calls. Little load. WHo cares |
| 19:01.55 | drmessano | You had me at 50 calls |
| 19:02.20 | admin0 | well who knows this grows big and it grows to 500 calls and i have to compile again .. for start, i will use the precompiled binaries |
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| 19:02.59 | yebyen | Penguin: well, the thing is over for today, and it could be 24 months before we have to try it again, so thanks for your help |
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| 19:44.26 | devhappy | hey, I'm having a strange issue with asterisk 11 and confBridge |
| 19:44.38 | devhappy | the bridge is working, but I keep getting the following Error message |
| 19:44.40 | devhappy | WARNING[3855][C-00000002]: channel.c:5956 ast_request: No channel type registered for 'Bridge' |
| 19:49.52 | yebyen | Penguin: so you say I should be seeing some kind of timeout message in the console |
| 19:50.00 | yebyen | even with 6 -v's i don't |
| 19:50.23 | devhappy | what version are you using yebyen |
| 19:51.03 | yebyen | 1.6.2.14 |
| 19:51.39 | devhappy | i've found the easiest way to set up console logging is just to configure it in logger.conf and dont be afraid to spam the v's |
| 19:52.11 | devhappy | but 1.6 usually doesn't need logger.conf |
| 19:52.22 | devhappy | at least i get all the info i need without it |
| 19:52.34 | yebyen | well i'm just doing asterisk console -rc -vvvvvv |
| 19:52.47 | yebyen | so i can see everything in real time, I think |
| 19:54.36 | yebyen | this manual says the default timeout is 5 or 10 seconds |
| 19:54.41 | yebyen | depending on what timeout you're talking about |
| 19:54.42 | yebyen | hmm |
| 19:54.53 | yebyen | if i set absolute, digit, response all to 0 |
| 19:54.57 | ChannelZ-Wk | devhappy: do you have a bridge entry defined (type=bridge) or using default_bridge |
| 19:54.57 | yebyen | maybe that will help |
| 19:55.41 | devhappy | ChannelZ-Wk: yes, i have confbridge with [default_bridge] |
| 19:55.42 | devhappy | type=bridge |
| 19:56.01 | devhappy | and a default_user and general section |
| 19:57.42 | ChannelZ-Wk | oh actually that's a channel message.. hrmm |
| 19:57.46 | yebyen | Penguin: i think that idea of adding a 4 hour wait after calling ChanSpy is what I needed to do. |
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| 19:58.48 | ChannelZ-Wk | What devices are involved? |
| 19:58.51 | yebyen | damn |
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| 19:59.57 | yebyen | that doesn't help either :D |
| 20:00.56 | devhappy | just cisco sip phones |
| 20:01.27 | devhappy | the message shows up with just 1 person, or more than 1 person in the channel |
| 20:02.39 | devhappy | actually looking closer, it generally only shows up right after someone joins the ConfBridge, |
| 20:03.06 | ChannelZ-Wk | does "core show channeltypes" show Bridge? |
| 20:03.42 | devhappy | no ConfBridge, SIP, Agent, local |
| 20:03.44 | Penguin | yebyen: It doesn't really make any sense that anything AFTER the chanspy will help. |
| 20:04.39 | ChannelZ-Wk | Well you can 'module load chan_bridge' and that should make the error go away, though I'm not sure why other things work without it. Not positive what all chan_bridge is responsible for I guess. |
| 20:04.58 | devhappy | yeah, me neither, it sounds like that will fix it though |
| 20:05.30 | ChannelZ-Wk | are you not using autoload in modules.conf or have specifically noload'd that one? |
| 20:07.04 | devhappy | had something to do with some problems with autoload in 1.6 years ago, in the process of upgrading to 11, I'll have to see if those reasons still make sense, hope they are documented (cross fingers!) |
| 20:08.10 | ChannelZ-Wk | FWIW I've been using autoload since forever I think. Just clean out the configs you aren't using so it doesn't load up a bunch of crap with crazy sample configs |
| 20:09.26 | devhappy | oh yeah, no sample configs here, just a very complicated dial plan, and lots of agi scripts |
| 20:12.53 | devhappy | ChannelZ-Wk: thanks for your help |
| 20:15.12 | yebyen | Penguin: yeah, what worries me is that chanspy isn't even terminating, so I'm having to go back in there and request hangup the Local/ channel each time |
| 20:15.24 | yebyen | even if I add Hangup() after it |
| 20:15.43 | yebyen | (which shouldn't be necessary right) |
| 20:15.55 | yebyen | so, it's not actually going to t=> |
| 20:15.58 | Penguin | Anything AFTER the chanspy is not relevant. |
| 20:16.40 | Penguin | Are you using any option to make chanspy end? |
| 20:17.30 | yebyen | we have qsv(2)b |
| 20:17.34 | yebyen | for options |
| 20:18.06 | yebyen | which is "quiet beep, skip channel announce, volume 2, only spy on bridged channels |
| 20:18.41 | Penguin | If you are not using the options to end chanspy, why do you expect chanspy to hangup? |
| 20:19.25 | Penguin | I'd use E for that. |
| 20:19.26 | yebyen | only because it seems to be quitting |
| 20:19.41 | yebyen | E doesn't seem to be an option in our ChanSpy application |
| 20:19.52 | Penguin | ChanSpy() will sit quietly waiting on something to comes onto the channel. |
| 20:20.00 | Penguin | If you don't have E, try S. |
| 20:20.05 | yebyen | yes, but the call is still going on :) |
| 20:20.14 | yebyen | we don't have S either |
| 20:20.21 | Penguin | ~upgrade asterisk |
| 20:20.21 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 20:20.41 | yebyen | yeah, that's on the bucket list |
| 20:20.48 | yebyen | so 1.8.24.0 or better |
| 20:21.48 | yebyen | well this is nice, I don't think we really wanted B |
| 20:21.53 | yebyen | now I can hear the line ringing |
| 20:22.20 | Penguin | You didn't say B, you said b, which is completely different. |
| 20:22.42 | yebyen | i did mean b |
| 20:22.43 | yebyen | sorry |
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| 20:26.38 | yebyen | ok, well i'll try and reproduce on a newer version and if I can't, then we know what the answer is |
| 20:27.03 | yebyen | i think it probably is a bug, seeing how no error messages or any kind of messages are printed on the console when the line goes dead |
| 20:33.22 | yebyen | ok, thanks again |
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| 21:46.43 | pancho_jay | Hi, someone can helpme configuring DIDs? |
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| 21:48.28 | WIMPy | ~ask |
| 21:48.28 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:49.36 | pancho_jay | I receive call from our provider, but it never rings on extensions |
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| 21:50.18 | WIMPy | Turn up verbose an see if the call arrives at all. |
| 21:51.49 | pancho_jay | WIMPy, ok, i will try |
| 21:51.52 | pancho_jay | thanks |
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| 22:03.37 | pancho_jay | WIMPy, call arrives to my Asterisk |
| 22:04.12 | pancho_jay | WIMPy, but I don't know how to write dialplan to redirect calls from provider to extension 1234 |
| 22:04.20 | WIMPy | Ok, then the message(s) should tell you why it doesn't get further. |
| 22:04.51 | WIMPy | You don't. You have to configure the extension your provider sends the calls to. |
| 22:06.20 | pancho_jay | WIMPy, sorry for my ignorance (I am newbie). How can achieve that? |
| 22:06.59 | WIMPy | You should have seen which extension gets called. |
| 22:07.16 | WIMPy | Configure that one in your extensions.conf. |
| 22:07.35 | pancho_jay | ok, i will try |
| 22:07.39 | navaismo | whisper The boook |
| 22:07.48 | navaismo | ~book |
| 22:07.48 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 22:18.08 | pancho_jay | chan_sip.c:22622 handle_request_invite: Call from '171038XXXX' (a.b.c.d:5060) to extension 's' rejected because extension no |
| 22:18.09 | pancho_jay | t found in context 'from-provider'. |
| 22:18.16 | pancho_jay | WIMPy, I got ^ |
| 22:18.54 | WIMPy | So they don't send any extension so the s extension is used. |
| 22:20.12 | pancho_jay | WIMPy, 'from-provider' context is something like: |
| 22:20.16 | pancho_jay | exten => _X.,1,NoOp(${EXTEN}) |
| 22:20.29 | pancho_jay | should match anything |
| 22:20.37 | WIMPy | _X. doesn't match s. |
| 22:21.02 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.64) |
| 22:21.06 | WIMPy | No. _X. will only match extension that are at least 2 characters long and start with a digit. |
| 22:21.15 | pancho_jay | WIMPy, nice.... OK. I will check dialplan docs and try again |
| 22:21.41 | pancho_jay | I don't understand why call comes without extension |
| 22:22.16 | WIMPy | Ask your provider or check your registration. |
| 22:23.01 | pancho_jay | WIMPy, which format should have registration string? |
| 22:23.30 | WIMPy | It's documented in the sample config. |
| 22:24.38 | pancho_jay | WIMPy, my registration string is username:secret@ipaddress ---> i think this is right |
| 22:25.27 | WIMPy | That's missing an extension. |
| 22:25.57 | pancho_jay | WIMPy, something like username:secret@ipaddress/didnumber ? |
| 22:26.06 | WIMPy | yes |
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| 22:41.06 | pancho_jay | WIMPy, you rocks! It works fine. Missing /extensionnumber in registration string was the problem |
| 22:42.50 | pancho_jay | thank you! |
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| 23:10.33 | admin0 | i have a strange issue .. i get: chan_sip.c:23298 handle_request_invite: Call from 'PEER-IN' (a.b.c.d:5060) to extension '91xxxx' rejected because extension not found in context 'phonein'. |
| 23:10.42 | admin0 | my extensions.conf has [phonein] exten => _X.,1,Dial(SIP/${EXTEN}@PEER-OUT) |
| 23:11.00 | admin0 | is there certain module that I am missing to compile for this ? |
| 23:16.06 | [TK]D-Fender | admin0: "dialplan show" <- |
| 23:16.52 | admin0 | this does not show my contet anywhere |
| 23:17.02 | admin0 | just this one: http://pastebin.com/PHx0CKhG |
| 23:17.39 | ChannelZ-Wk | then your dialplan is busted or not loaded |
| 23:17.44 | ChannelZ-Wk | dialplan reload |
| 23:19.04 | admin0 | *CLI> dialplan reload |
| 23:19.04 | admin0 | No such command 'dialplan reload' (type 'core show help dialplan reload' for other possible commands) |
| 23:19.18 | admin0 | looks like i forgot to include some modules/ |
| 23:19.22 | admin0 | while compiling |
| 23:20.45 | ChannelZ-Wk | module load pbx_config |
| 23:20.54 | admin0 | yeah |
| 23:21.00 | admin0 | compiled now |
| 23:21.49 | admin0 | here is the difference I get .. apt-get vs compiled one : http://pastebin.com/6B3U2Kfv .. |
| 23:23.02 | [TK]D-Fender | that doesn't show what modules you have loading. |
| 23:23.08 | [TK]D-Fender | just transcodiing info |
| 23:23.12 | admin0 | yes .. |
| 23:23.25 | [TK]D-Fender | that doesn't show us a sane modules.conf |
| 23:23.28 | admin0 | was saying that its worth the effort |
| 23:23.32 | admin0 | :D |
| 23:23.34 | [TK]D-Fender | or that your config file condition is sane |
| 23:27.45 | admin0 | thanks guys ... will chat tomorrow |
| 23:27.48 | admin0 | i have to log off |
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