| 00:09.16 | pchero | Hi, all. |
| 00:09.16 | pchero | Recently, I made some module for the Asterisk. |
| 00:09.16 | pchero | It's an asterisk-tiresias; Audio fingerprinting and recognition module for the Asterisk. |
| 00:09.16 | pchero | It designed for answering machine detection using audio fingerprint and other purposes. |
| 00:09.16 | pchero | Please feel free to look around it! :) |
| 00:09.16 | pchero | https://github.com/pchero/asterisk-tiresias |
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| 00:20.26 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.1 (2018/06/11), Standard: 15.4.1 (2018/06/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 05:40.22 | Janos | hi everyone, trying to get asterisk realtime working but most of the documentation is for mysql. I want to use curl. I was able to get queues and queue_members working but now struggling with sippeers. I see no errors on the console and the peers are not getting created. Has anyone achieve this before? |
| 05:41.15 | Janos | my next step is try to get it working with mysql and try to find why it does not work with curl |
| 05:41.52 | Janos | recompile asterisk with debug and gdb my way to the docs /shrugs |
| 05:42.57 | Janos | if anyone has any info about this scenario I would appreciate it |
| 05:43.58 | Janos | pretty sure I'm just missing some field that I'm not sending back with my http response but the lack of info about this setup is well ... |
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| 07:59.36 | v0lZy | hey |
| 07:59.44 | v0lZy | Quick question regarding asterisk sip trunk.. |
| 08:00.03 | v0lZy | I have a peculiar situation where unlike the stuff I've done before, I have a separate trunk for incoming and outgoing calls from the same provider |
| 08:00.16 | v0lZy | do I need a context= on the one i'm using for outgoing? |
| 08:00.52 | v0lZy | (I already have an [OutgoingPatterns] context in which I directly Dial() |
| 08:02.25 | v0lZy | I know I required the context= in sip.conf for processing incoming calls.. but having my own separate context for outgoing calls in extensions.conf, do I need to specify context= in the sip.conf for the outgoing trunk? |
| 08:12.55 | drmessano | You literally could have just asked âdo I need to specify context= in the sip.conf for an outgoing trunk?â |
| 08:13.01 | drmessano | A: No |
| 08:15.17 | v0lZy | Thanks drmessano |
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| 08:24.55 | AsteriskRoss | rmudgett: Would you say not passing async_operations to pjsip transports is a bug, if so should i create an issue? |
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| 15:42.14 | rfr__ | Hello All, I am having problems getting a dhadi/Asterisk configuration ( based on an old Zaptel/Asterisk configuration that does work ) to work at all. So far I don't even get a dial tone but I don't see any obvious errors from the CLI. Here's my CLI output and config files. Any help would be greatly appreciated. https://pastebin.com/HBtYaQZa |
| 15:43.40 | [TK]D-Fender | signalling=fxo_ks <- should be only 1 "l". |
| 15:44.09 | [TK]D-Fender | Also make sure you have the molex connected if you are using an analog TDM card |
| 15:49.05 | rfr__ | Thanks [TK]D-Fender. Do I need to only make that change in chan_dhadi.conf? Should 'fxoks=9-24' stay the same in system.conf? Does molex refer to a channel bank? If so I'm using an adit600. |
| 15:50.11 | [TK]D-Fender | No, sounds like you're using a digital card to interface to it |
| 15:51.12 | [TK]D-Fender | Actual, AFAIK for FXOKS show be FXOLS for your phones |
| 15:53.04 | rfr__ | Ok, so just change to signalling=fxo_ks to 'signalling=fxo_ls' in chan_dhadi.conf and FXOKS to FXOLS in system.conf? |
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| 16:04.22 | [TK]D-Fender | yes |
| 16:04.29 | [TK]D-Fender | and remove the double L's |
| 16:04.31 | [TK]D-Fender | signaling |
| 16:04.34 | [TK]D-Fender | not signalling |
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| 16:08.24 | rfr__ | yeah, sorry. Thanks again. I will give it a try. |
| 16:09.05 | Toerkeium | Hello Guys, I've an Asterisk install running on a regular PC, for no more than 5 concurrent calls. Is the RasPBX - Asterisk for Raspberry going to handle well that? anyone knoes? I would like to make that PC dissapear if possible |
| 16:14.59 | [TK]D-Fender | Depends what those 5 calls are doing |
| 16:15.07 | [TK]D-Fender | But for that # it should be fine |
| 16:15.12 | [TK]D-Fender | generally.... |
| 16:16.32 | Toerkeium | thanks [TK]D-Fender |
| 16:17.53 | Toerkeium | you did quite the other day, I had a problem with the call being hangup after 30 seconds. Finally it was the EyeBeam Softphone, having an option called something like "In case of network disruption hangup the call" or something like that, just to let you know! |
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| 16:22.04 | rue | if I have a digium card detecting rings that didn't exist, can I fix it by bumping up a timer for ring duration (our provider is known to emit erronious pulses) |
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| 17:52.23 | jeffspeff | I'm trying to compile and run asterisk as non root user. I have followed the instructions from http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-13-SECT-4.html which is old but seems to have worked for the most part. My issue is that when I run 'service asterisk status' i see the following " PID file /home/ast/asterisk-bin/run/asterisk/asterisk.pid not readable (yet?) after start. |
| 17:52.23 | jeffspeff | " and "asterisk.service: Supervising process 60519 which is not our child. We'll most likely not notice when it exits." any suggestions? |
| 17:58.04 | [TK]D-Fender | I'd suggest showing us the attempt to start it, then to connect, showing the process is running. Then if all that fails also a manual start as the proper user to see if it bombs live.... |
| 18:00.12 | jeffspeff | [TK]D-Fender, it starts fine and i can access it by running asterisk -rv all seems to be good with the exception of those two messages. |
| 18:03.24 | [TK]D-Fender | We should look at this directly then. |
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| 19:06.46 | *** topic/#asterisk by bford -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0-rc1 (2018/07/03), Standard: 15.5.0-rc1 (2018/07/03); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 20:26.13 | Samot | Man I can't wait until the end of October. |
| 20:26.22 | seanbright | ugh, i know right? |
| 20:26.33 | seanbright | it's going to be fucking awesome |
| 20:26.40 | seanbright | when that thing happens that we're both excited about |
| 20:26.57 | Samot | Asterisk 16 LTS... |
| 20:27.05 | seanbright | are we talking about my grandfather's birthday? |
| 20:27.05 | seanbright | oh |
| 20:27.21 | seanbright | well he'll be 90 |
| 20:27.21 | rue | hah |
| 20:27.27 | seanbright | i'll pass on your well wishes |
| 20:27.49 | Samot | I think he's close to the end of his LTS agreement. |
| 20:28.01 | rue | 16, wait, why did asterisk go so version crazy all the sudden? |
| 20:28.15 | seanbright | all of a sudden? |
| 20:28.15 | Samot | Because 15 was not deemed LTS |
| 20:28.31 | Samot | So it broken the "odd release is LTS" cycle. |
| 20:28.33 | rue | so, we need 2 versions per year or what? |
| 20:28.34 | seanbright | 1.4 -> 1.6 -> 1.8 -> 10 -> ... 16 |
| 20:28.40 | Samot | No. |
| 20:28.43 | rue | ah |
| 20:28.48 | Samot | There is a Standard version |
| 20:28.50 | seanbright | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 20:28.52 | Samot | And an LTS version. |
| 20:29.00 | rue | hmm |
| 20:29.15 | Samot | 13 is the current LTS |
| 20:29.20 | Samot | which is until 2020 |
| 20:29.26 | rue | I'm still on the version that says "Message 17, 17." |
| 20:29.46 | Samot | 15 is the Standard release which last 1 year. |
| 20:29.52 | seanbright | if that is < 1.8 you should never upgrade |
| 20:29.56 | Samot | Standard releases are where new things are introduced. |
| 20:30.04 | seanbright | the last good version of asterisk was 1.2 |
| 20:30.07 | seanbright | everyone knows that |
| 20:30.08 | rue | seanbright, uh, why |
| 20:30.20 | rue | are there bugs? |
| 20:30.20 | malcolmd | seanbright: totally |
| 20:30.22 | seanbright | i'm talking out of my ass, rue |
| 20:30.29 | seanbright | keep up |
| 20:30.58 | rue | oh good, maybe the signalling bug for my NewBridge channelbank has been fixed |
| 20:31.11 | seanbright | i know some of those words... |
| 20:31.34 | rue | 'the' in a common one, yes |
| 20:32.01 | rue | the most popular word in the english language is 'the' for that reason alone |
| 20:32.11 | seanbright | i have a few 3624s in production with asterisk 15 |
| 20:32.16 | seanbright | no problems at all |
| 20:32.43 | seanbright | BUT... i am also making that up |
| 20:32.49 | seanbright | so your mileage may vary |
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| 20:54.00 | chrisbur2 | Hi guys how are you? |
| 20:54.25 | seanbright | chrisbur2: personally, i am wonderful. how are you? |
| 20:54.39 | chrisbur2 | great seanbright :) |
| 20:54.51 | seanbright | awesome. i don't answer questions here. |
| 20:54.57 | seanbright | but i just wanted to say 'hi' |
| 20:56.18 | chrisbur2 | ohhh thanks, anycase, i just wondering to know o understand if someone is expert in Asterisk i have runnign a pretty old one , and we implement a Framework using for be callcenter screen, i need just to understand if you guys know anyone new framework poensource to apply and implement Phone, plus some API |
| 20:56.37 | chrisbur2 | i wondering to make a implementation with SAP |
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| 21:19.49 | Samot | chrisbur2: You are going to have to repeat that a little more clearly. |
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| 23:24.02 | wyoung | Samot: I like it when people ask a question in broken English, stick around for 30 minutes then leave without getting an answer. |
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| 23:31.37 | Maliuta_ | wyoung: question answer you not? |
| 23:33.40 | wyoung | Maliuta_: Nice, is this when you leave now? |
| 23:57.06 | Maliuta_ | wyoung: you trying to kick me out? |
| 23:57.08 | Maliuta_ | typical :P |
| 23:57.35 | Maliuta_ | but, hey a couple of other people left. That should make up for it :) |