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| 00:21.53 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); 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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| 07:47.34 | pawiecki | Hi. I have a question about a basic subject - sip peers. How should i interpret a sip peer/sip account? Sometimes i see mac address being used as a sip account id, but often it's just extension or did. What is the most sane/logical solution in your opinion, and why? |
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| 09:13.12 | Kuunsi | it's just a username |
| 09:13.31 | Kuunsi | you could use czquwidviauevk as sip peer name, if you wnat |
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| 10:12.51 | jkroon | i've got a scenario where a channel is already Up, and then I receive RINGING over an IAX/2 channel, but the caller never hears ringing. From what I can gather from chan_iax2.c it just passes the control frames into asterisk core, so my guess is that whatever is on the other end of that bridge should generate ringing (or pass the control frame further back)? |
| 10:13.21 | jkroon | what would chan_sip do in a case where it receives a RINGING control frame from the bridge on an already answered channel? |
| 10:30.18 | Rac-on | jkroon: if the 200 OK has been send, it would be impossible to send an 180 ringing, since the transaction is finished |
| 10:32.18 | jkroon | Rac-on, yes, agreed. but normally chan_sip would use indications to generate a ringing on the incoming channel wouldn't it? ie, we can't send 180 ringing, so send ringing as audio? |
| 10:33.01 | jkroon | chan_sip doesn't handle it at all. |
| 10:33.49 | jkroon | looking at sip_indicate there is quite literally an if (...) { ... } else { /* Well, if it's not reasonable, just send in-band */ } - but that in-band is never set up. |
| 10:36.41 | jkroon | what I do find interesting is that other control frames looks for AST_STATE_UP and treats it special, ringing seems to be a missed case... |
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| 10:55.39 | eject_ck | Hi guys, we have asterisk server and I'm looking for "call from website" functionality via browser to allow customers call our support via web browser. What are possible solutions for now? |
| 10:55.44 | eject_ck | paid or free |
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| 11:35.24 | Rac-on | jkroon: that would be a 183, still cant do that when the call has been answered |
| 11:38.18 | jkroon | Rac-on, but there is no reason why we can't generate the tone locally and just transmit it as normal rtp. |
| 11:38.55 | jkroon | but we're busy rewiring a few things so that if the channel is Up already to rather send MOH anyway, so not major, just an annoyance. |
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| 11:55.10 | file | the res = -1; causes it to send inband |
| 11:55.34 | jkroon | file, interesting. but it doesn't work if the call is already Up. |
| 11:55.58 | file | it should |
| 11:56.05 | file | do you have an indications.conf file? |
| 11:56.20 | jkroon | i do. |
| 11:56.30 | file | if you bump up core debug it'll tell you what it's doing |
| 11:56.47 | jkroon | even debug at level 1 brings this box down this time of the day. |
| 11:57.05 | jkroon | to what level do i need to take it? I'll do after-hours when the box is much more quiet. |
| 11:57.21 | file | 1 would do it |
| 11:57.32 | jkroon | thanks, i'll see what I can find. |
| 11:58.03 | jkroon | i note my default country setting it wrong, but that should at worst cause it to sound wrong, not fail completely. |
| 11:58.24 | jkroon | and g729 licensing isn't a problem either ... or shouldn't be. |
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| 12:33.30 | jkroon | file, it seems channel.c returns -1 for AST_CONTROL_RINGING if the channel is not Up, and 0 if it is. I'm wondering if this is resulting in something happening differently. |
| 12:33.43 | jkroon | other than the warning not being printed ... |
| 12:34.57 | file | the code in there which plays out the tones doesn't care about that |
| 12:36.46 | file | I just used PJSIP against a 14.5.0 build with that scenario in the dialplan - Answer, Ringing, Wait - and got inband ringing |
| 12:36.51 | file | and it uses the same thing |
| 12:38.29 | jkroon | we moved to playing MOH instead but obviously we'd still like to understand the scenario better. |
| 12:39.01 | jkroon | will have to perform the test over the weekend though ... |
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| 15:27.12 | jrun | using journald; asterisk is run with -f; i want all logging to go to the console asterisk hookup to. how do i do that? |
| 15:27.30 | jrun | just comment everything under [logfile] in logger.conf? |
| 15:29.20 | jrun | empty dataformat btw appends [] to the begging of the lines; not cool :) |
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| 15:43.22 | jrun | ip_identify_match_check is too frequent; how do i bring it down? |
| 15:43.32 | jrun | qualify somewhere? |
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| 16:21.27 | jrun | codec_opus.c:94 opus_enc_set: Opus: failed to create encoder (-1) |
| 16:21.54 | jrun | this on 14.5.0 (based on the git tags) |
| 16:22.17 | jrun | codec_opus was pulled as binary |
| 16:22.43 | jrun | i do have libopus installed on my machine (gentoo) but configure script doesn't pick it up |
| 16:23.03 | jrun | sorry, it does pick it up but then it fails to compile. |
| 16:23.19 | jrun | i'm gonna retry with a different PREFIX which i think was the problem. |
| 16:23.39 | jrun | hmm actually i don't see codec_opus.c anywhere. |
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| 17:33.53 | salviadud | I just setup the battletoads pause music as musiconhold, I recommend it. |
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| 19:13.51 | notze | gyus i still have only one lasting problem. i got a number range from 0-9. if i use the 0 everything works perfect. but if i choose any other number the phone is ringing and then it will be hanged up with 487. |
| 19:14.00 | notze | and this is the reaction from asterisk: https://paste.ubuntu.com/24723734/ |
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| 19:30.46 | [TK]D-Fender | notze, you aren't looking at SIp debug so we can't tell what side aborted and why |
| 19:31.11 | notze | [TK]D-Fender, no problem |
| 19:31.14 | notze | give me a sec |
| 19:33.48 | [TK]D-Fender | Executing [4940xxxxxxx9@sipgate-in:1] Dial("SIP/sipgate-0000001a", "SIP/1235") in new stack |
| 19:33.56 | [TK]D-Fender | also this is just dialing what loks to be a FIXED phone |
| 19:34.08 | [TK]D-Fender | nothng looks variable about how * processes that call |
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| 21:23.24 | infotek | hi guys what is the current status of pjsip from a stability standpoint? |
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| 21:52.37 | [TK]D-Fender | infotek, This was already answered for you in #freepbx more than 3/4 hour ago. |
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| 23:49.26 | infotek | [TK]D-Fender: the responses in #freepbx the question was met with an acerbic unhelpful response. |
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