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| 00:21.45 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.1 (2017/04/04), 11.25.1 (2016/12/08), Standard: 14.3.1 (2017/04/04); 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 |
| 00:50.44 | Kobaz | just say 'Them' and 'Their' |
| 00:50.51 | Kobaz | apparently that's the new gender neutral thing going on |
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| 04:07.34 | wyoung | Kobaz: Yeah, it's all about gender neutrality these days, equal optitunity (unless violence or chivalry is involved), etc.. All good things but I usually let "hey guys" or similar slip. I am getting better. |
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| 08:46.49 | sotoz | Is there a shorthand to set a channel variable in the dialplan only if it's not set? |
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| 10:51.15 | MacroMan | Is it possible to see what codec a channel is/has been using? |
| 10:51.30 | MacroMan | I don't see anything on the CLI that tells me directly. |
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| 11:01.15 | MacroMan | I managed to find it. I used `core show channel channelname` |
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| 11:20.32 | sibiria | hello. when declaring a trunk in sip.conf, is it possible to somehow specify multiple host= segments and have asterisk use them in a round-robin fashion? |
| 11:20.36 | sibiria | asterisk 13 |
| 11:21.26 | sibiria | the type is peer, btw. |
| 11:24.32 | file | no. |
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| 12:35.37 | MacroMan | sibiria, You could achieve this using dialplan logic, but there is nothing built in. |
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| 12:40.15 | sibiria | MacroMan: i guess using a hostname with SRV records are the proper solution |
| 12:40.29 | sibiria | is* |
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| 13:38.27 | [TK]D-Fender | sibiria, Probably not. because even if you rotate on lookup, think about quality and other comms like registration flipping them. Not great |
| 13:38.38 | [TK]D-Fender | Still a job for dialplan |
| 13:42.26 | sibiria | in this case the provider offers to distribute the load over two peers, free of choice |
| 13:42.41 | sibiria | the two are identical (barring that one can't predict current load on any of the two) |
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| 13:52.32 | [TK]D-Fender | If the provider will balance it, why not... |
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| 14:09.49 | sotoz | How can I use AMD from ARI? |
| 14:10.13 | sibiria | we're simply trying to get around having a handful of [providername_1] [providername_2] etc. for each of their peers |
| 14:10.23 | Samot | The same why you would with AMI, AGI or dialplan. |
| 14:10.38 | Samot | You originate a call and send it to AMD. |
| 14:10.39 | sibiria | so that we can just issue to "providername" |
| 14:10.42 | sotoz | Samot: there is no endpoint for the AMD() application |
| 14:11.04 | sotoz | I try to just continue in dial |
| 14:11.23 | Samot | What are you trying to do? |
| 14:12.42 | sotoz | I want to originate a call through ARI, and then I want to continue in dialplan (in order to use the AMD() app. After that I wanto to reregister the channel to the stasis app somehow in order to decide what to do with the results of the AMD() app |
| 14:13.35 | Samot | So you would CreateChannel/Originate, etc... |
| 14:14.28 | Samot | That Dial()'s a SIP channel out a trunk, for example, you get the stasis of that channel.. |
| 14:14.38 | Samot | When that channel answers, originate will send it to the AMD context.. |
| 14:15.03 | zaf | just got cdr logging to an azure sql db... i feel so dirty |
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| 14:15.56 | sotoz | lemme show you my dialplan |
| 14:17.04 | sotoz | https://pastebin.com/qFKfxhEj |
| 14:18.18 | sotoz | hmm I think I know what I'm doing wrong |
| 14:18.37 | sotoz | when I'm originating a new call from ARI the channel is not in the stasis |
| 14:19.10 | sotoz | so when i'm doing a continue in dial plan from ARI it doesn't continue where I wanted |
| 14:20.17 | sotoz | that would explain this error that I'm getting "WARNING[2499][C-000000cf]: pbx.c:4407 __ast_pbx_run: Channel 'PJSIP/voip-000001f6' sent to invalid extension but no invalid handler: context,exten,priority=voip,s,2" |
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| 15:38.25 | adeeln | does * 11.7 support CN passthrough? |
| 15:39.54 | igcewieling | You might start by defining "CN" |
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| 15:40.55 | igcewieling | Calling Name, Caller Name, Chinchilla Nude? |
| 15:41.05 | adeeln | sorry, comfort noise / vad |
| 15:41.17 | adeeln | but i like your idea of Chinchilla Nude; that sounds interesting |
| 15:41.42 | igcewieling | Asterisk 11 doesn't support comfort noise. I don't think Asterisk 13 does either, but I'm not totally sure about that. |
| 15:42.05 | igcewieling | Then entire SIP stack changed in Asterisk 12+ |
| 15:42.09 | adeeln | igcewieling: doesn't support generating Comfort noise? or passing it through? |
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| 15:43.01 | igcewieling | That would be a question for the developers. I'd expect neither to work. |
| 15:43.16 | igcewieling | So little out there requires it, it isn't usually an issue. |
| 15:44.33 | [TK]D-Fender | <igcewieling> Then entire SIP stack changed in Asterisk 12+ <- More like "An entire alternate stack got added". Nothing was taken away from chan_sip |
| 15:45.17 | igcewieling | I tried Asterisk 13 the other day...for about 2 hours. 8-| |
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| 15:46.06 | [TK]D-Fender | For you that's a big step.. I'm so proud! |
| 15:46.07 | igcewieling | chan_sip has been replaced. The fact it is still available doesn't change that. |
| 15:46.24 | igcewieling | [TK]D-Fender: I was verifying a bug was not fixed before complaining about it. |
| 15:46.44 | [TK]D-Fender | Just because you've got a side chick doesn't mean the old one is gone. |
| 15:46.49 | [TK]D-Fender | it's called MULTI-TASKING |
| 15:51.44 | igcewieling | [TK]D-Fender: the old one doesn't get the same level of attention as the new one. |
| 15:52.05 | [TK]D-Fender | BE THANKFUL FOR WHAT YOU'VE GO YOU UNGRATEFUL WHORE! |
| 15:52.08 | [TK]D-Fender | ;p |
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| 15:54.33 | igcewieling | LOL! Thats me! |
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| 18:10.11 | LunaLovegood | How do I stop Asterisk from qualifying the same phone twice? I use PJSIP, and in my endpoint section I have contact_permit=172.19.10.82 , and in the aor section I have contact=sip:YWuvMwl2Zr@172.19.10.82:5060 , ( I use static IPs ). So Asterisk already has contact info for the phone, but then the phone does a REGISTER, and Asterisk ends up with a duplicate contact. "CLI> pjsip show aors" shows two |
| 18:10.18 | LunaLovegood | contacts for that SIP phone. |
| 18:11.43 | LunaLovegood | All my phones have static IPs, so I'd like to be able to make one ring even if it hasn't REGISTER'd yet. So I'd like to keep the IP informations in my pjsip.conf |
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| 18:22.10 | igcewieling | LunaLovegood: what phoines do you use? |
| 18:22.50 | igcewieling | On polycom phones if you put in the failover server as the same as the main server, you will get that symptom. |
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| 19:13.07 | LunaLovegood | They're cable modem MTAs. I've had trouble getting them to use passwords, so I instead gave them static IP addresses and made sure the connections where encrypted at L2. |
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| 19:15.49 | LunaLovegood | Is it possible to make Asterisk respond with ACK to REGISTERs, but otherwise ignore them and not add contacts to the aors? |
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| 20:10.38 | igcewieling | why use register at all? |
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| 20:19.01 | LunaLovegood | I wouldn't, except the modems do it and I can't stop them. |
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| 21:44.00 | freexer | Is the Send: Pack coming from 10.10.10.10 or going to? https://hastebin.com/umucuseseg.css |
| 21:44.23 | sotoz | What is the best way to use AMD for a channel that got originated from a stasis app through ARI? |
| 21:55.08 | *** join/#asterisk war9407 (war@static-72-73-18-14.clppva.fios.verizon.net) |
| 22:02.38 | *** join/#asterisk Samael28 (~Samael28@79.110.128.128) |
| 22:11.19 | Samot | sotoz: Are you trying to do some sort of auto dialing? |
| 23:01.24 | *** part/#asterisk kharwell (kharwell@nat/digium/x-coyiiieqglvynyke) |
| 23:13.31 | *** join/#asterisk Samael28 (~Samael28@79.110.128.128) |
| 23:16.17 | *** join/#asterisk dasjoe (~dasjoe@americangirlscouts.org) |
| 23:38.51 | *** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net) |
| 23:47.30 | *** join/#asterisk freexer (~freexer@cpe-45-37-13-248.nc.res.rr.com) |
| 23:47.49 | freexer | Is the Send: Pack coming from 10.10.10.10 or going to? https://hastebin.com/umucuseseg.css |
| 23:51.17 | *** join/#asterisk averythomas (~averythom@cpe-72-224-252-57.maine.res.rr.com) |