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| 07:29.51 | mpotra | Hello everyone. I've got an issue with Asterisk 14 (from source), on Debian. Apparently res_stasis_playback fails on playing a sound file: WARNING[9236][C-00000005]: res_stasis_playback.c:277 playback_final_update: 1476429943.9: Playback failed for sound:hang-on-a-second |
| 07:30.38 | mpotra | But using the Playback() cmd in an extension, plays it fine (yes, file exists). Does anyone have any ideas what I should look into for fixing it? |
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| 09:35.17 | ^sovo^ | hey guys |
| 09:35.32 | ^sovo^ | i did test the calls betwen 2 servers and here is what errors come up |
| 09:35.34 | ^sovo^ | http://pastebin.com/xv2Bhp1q |
| 09:35.52 | ^sovo^ | bin tryign to call from server 2 to server 1 |
| 09:36.27 | ^sovo^ | asterisk 1.8/freepbx |
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| 10:30.58 | Samot | [2016-10-14 11:23:43] NOTICE[2051] chan_sip.c: Call from '203' (86.61.64.143:531 27) to extension '301' rejected because extension not found in context 'internal |
| 10:32.10 | Samot | [2016-10-14 11:23:10] NOTICE[2061] chan_iax2.c: Host 192.168.1.109 failed to aut henticate as System2 |
| 10:34.05 | Samot | Oh he left. He's so not getting this job. |
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| 11:12.35 | tuxd00d | Samot: Yep, may have been more appropriate for #freepbx anyway |
| 11:16.22 | Samot | Yeah, he was told that last week. |
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| 12:49.23 | jpthomasset | Hello everyone, I have an issue with asterisk 13 where a thread end up using 100% cpu. During a redirect with an extra channel, one of the channel thread is stuck in the while loop at the end of bridge_channel_internal_join() where it waits for the flag AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT to be cleared. I had a look at the code and I may have found an issue but as I am new to asterisk, I would be happy to have some advices |
| 12:49.23 | jpthomasset | to check if my thinking is correct. |
| 12:55.51 | [TK]D-Fender | Always confirm your version |
| 12:55.53 | jpthomasset | What I have found is that bridge_channel_internal_join() calls bridge_channel_internal_pull() which clears the flag AST_FLAG_OUTGOING without locking the channel first so maybe the ast_clear_flag done inside the action_redirect from the manager thread is overwritten by this call in bridge_channel_internal_pull(). Any advice on that ? |
| 12:56.35 | jpthomasset | I'running asterisk 13.11.2 |
| 13:00.43 | [TK]D-Fender | I'd try taking this to #asterisk-dev |
| 13:01.17 | [TK]D-Fender | You're pretty heavy into the code and they pay more attention over there |
| 13:01.47 | jpthomasset | ok, thanks, sorry for posting in the wrong channel. |
| 13:03.34 | [TK]D-Fender | No, it's just fine here too, just that the devs may pay more attention to new msgs there. Justt offers you more chances at getting someone's attention who can look at this with you |
| 13:04.29 | jpthomasset | Ok, many thanks for your help ! |
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| 13:14.23 | infernix | i've set up call forwarding for an extention to a local sip phone SIP/A and a cell phone B (using outbound via a carrier, so no VOIP from asterisk to cell) with a Dial(A&B). Sometimes the cellphone incorrectly goes to voicemail immediately (e.g. bad reception) and then the call gets terminated to the cell |
| 13:14.57 | infernix | as far as asterisk is concerned, the call is handed off correctly, whether the cellphone actually answers or if the cellphone carrier sends it to voicemail |
| 13:15.27 | infernix | any ideas on detecting when it goes to voicemail and not terminating it there? |
| 13:16.38 | infernix | e.g. something like "press X to accept forwarded call", or even fancier by detecting audio keywords "voicemail of" |
| 13:17.21 | [TK]D-Fender | "core show application dial" <- |
| 13:17.30 | [TK]D-Fender | "core show application AMD" <- |
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| 13:21.45 | infernix | AMD looks ideal, but I'm not sure what you would suggest as a dial option to handle this |
| 13:22.52 | infernix | you mean to add a macro to Dial that waits for DTMF input? |
| 13:23.32 | [TK]D-Fender | DTMF or AMD |
| 13:29.03 | infernix | neat. thanks! |
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| 15:24.39 | jrun | can a template inherit another section? |
| 15:24.55 | jrun | what is the syntax for it (!,another_section)? |
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| 15:47.54 | Penguin | yes |
| 15:48.00 | stefan27 | What's the correct way of disabling music-on-hold on sip peer basis? |
| 16:02.44 | igcewieling | stefan27: one common way is to set it to a MoH class with no soundfiles. |
| 16:08.22 | stefan27 | using those attributes mohsuggest and mohinterpret? Thanks ill try create such a Moh class |
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| 16:15.09 | [TK]D-Fender | <stefan27> What's the correct way of disabling music-on-hold on sip peer basis? <- set to an invalid or empty class |
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| 16:25.11 | stefan27 | Thanks, I'm trying to create an empty class in musiconhold.conf right now |
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| 16:33.15 | TandyUK | any suggestions on how to get sngrep, sipcapture, etc working for TLS traffic? |
| 16:40.56 | Samot | Set them to watch for TCP traffic. |
| 16:47.38 | stefan27 | when [test] is configured in musiconhold.conf as directory=/var/lib/asterisk/moh/waves then everything works as expected when i set mohsuggest to test on a peer |
| 16:48.38 | stefan27 | but when I remove the only sound file test.mp3 from moh/waves/test.mp3 then asterisk says res_musiconhold.c:884 _get_mohbyname: Music on Hold class 'test' not found in memory. Verify your configuration. |
| 16:49.14 | stefan27 | how is an empty music class supposed to look like? |
| 16:49.53 | igcewieling | stefan27: exactly the same as regular moh class, but the directory it is pointed at has no sound files. |
| 16:50.12 | stefan27 | yes that's exactly how I configured it |
| 16:51.55 | stefan27 | I'm using asterisk 13.5.0 maybe there's a bug reported and solved between then and the latest version. Let me check |
| 16:55.03 | Penguin | Maybe you could create a single empty file and that would help? |
| 16:59.42 | stefan27 | unfortunately no, what's worse is that module load res_musiconhold.so reports no errors but the class is evidently not loaded correctly |
| 17:00.57 | stefan27 | There's 3 lines like res_musiconhold.c:1091 moh_scan_files: Scanning '/var/lib/asterisk/moh/ringing' for files for class 'ringing' for my correct classes called default, ringing and native |
| 17:01.39 | stefan27 | but there's no debug output mentioning the 4th class [test] when I made the directory it pointed to empty |
| 17:02.59 | stefan27 | when I add the file again it says [2016-10-14 19:02:31] DEBUG[24333]: res_musiconhold.c:1091 moh_scan_files: Scanning '/var/lib/asterisk/sounds/emptyfolder' for files for class 'test' |
| 17:03.37 | Penguin | I guess you need a silent file. |
| 17:04.04 | Penguin | 60 seconds of silence wouldn't be too large. |
| 17:04.25 | stefan27 | asterisk must not send any RTP to the holded side |
| 17:04.34 | stefan27 | so I'll have to dig deeper |
| 17:04.35 | Penguin | hmm |
| 17:06.21 | Samot | Well there needs to be an RTP stream. |
| 17:06.26 | Samot | Otherwise the call will drop. |
| 17:07.06 | Samot | When you put someone on hold, it updates the SDP with the media only being sendonly |
| 17:12.27 | stefan27 | sip.conf is configured so that asterisk doesnt touch the rtp |
| 17:13.42 | Samot | You mean you have directmedia on? |
| 17:14.18 | stefan27 | the only work-around which makes everything work is creating a moh-class pointing to a non-empty sound-folder, and then remove the sound-folder contents, but everything breaks again on next core reload/restart |
| 17:14.28 | stefan27 | yes, directrtpsetup |
| 17:15.53 | stefan27 | But I dont get it, internet and also this channel suggest having empty moh classes should be possible |
| 17:16.56 | Samot | So are these two local endpoints? |
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| 17:18.04 | stefan27 | local? they are two sip phones on my desk corresponding to 2 sip friends |
| 17:18.14 | Samot | Are they on the same network? |
| 17:18.21 | Penguin | Also the default musiconhold.conf suggests it's possible. |
| 17:18.22 | stefan27 | yes, same lan |
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| 17:18.46 | Samot | And you dont want the MOH played? |
| 17:20.02 | jrun | can (+) be applied to a template? |
| 17:20.08 | stefan27 | yes, because before phone A holds phone B, rtp flows from A to B which is what i want, but when A holds B, asterisk sends rtp to B which makes B start sending its rtp to asterisk |
| 17:20.16 | stefan27 | due to B having symmetric rtp setup |
| 17:20.40 | Penguin | jrun: What does it mean? |
| 17:21.15 | DivideBy0 | are there any asterisk apps/scripts that detect MOH playing on the remote end? For example, to notify you while you're on hold with your ISP for 30 mins and they finally pick up? |
| 17:22.18 | Samot | How would it know? |
| 17:22.24 | Samot | It's an audio stream. |
| 17:22.56 | Samot | It's receiving audio, whether it's an audio file or voice...it's just audio. |
| 17:23.34 | Samot | Unless you're going to look at the SDP in the INVITE to see what it is set to. |
| 17:24.26 | Samot | stefan27: MOH is set to the channel, not the user. |
| 17:25.46 | Samot | So if you don't want to use MOH then just don't set it on the channel. |
| 17:25.50 | Samot | Or channels |
| 17:29.06 | stefan27 | But this also seems to be a bug in res_musiconhold.c so I'm preparing a JIRA ticket; thanks for the assitance and have a nice weekend everyone |
| 17:29.16 | Penguin | There's WaitForSilence() if you can somehow incorporate that. |
| 17:30.02 | Samot | stefan27: You're using directrtpsetup... |
| 17:30.16 | Samot | Asterisk will only inject audio on the call like an IVR or MOH if it's told too. |
| 17:30.21 | Samot | Asterisk will only inject audio on the call like an IVR or MOH if it's told to. |
| 17:31.08 | Samot | When that device makes an outbound call, don't set MOH on the channel it's using. |
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| 17:32.07 | Samot | When it dials the other device or out to the PSTN, make sure that channel doesn't have MOH set on it. |
| 17:33.36 | Samot | Just don't call MusicOnHold() in the dialplan. |
| 17:34.32 | Penguin | You don't have to call MusicOnHold() to get music on hold if the phone has a hold button. |
| 17:35.28 | Penguin | MusicOnHold() is how you get dialplan to play music instead of having a phone initiate it. |
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| 17:44.52 | Samot | OK, so setup a MOH class that just plays back the silence files. |
| 17:47.14 | Penguin | That's what I suggested. |
| 17:47.25 | Penguin | He said no because he doesn't want the RTP stream to be active at all. |
| 17:51.34 | Samot | Uh? |
| 17:51.41 | Samot | If there's no RTP, there's no call. |
| 17:52.46 | Samot | There *has* to be RTP. |
| 17:54.43 | Penguin | Tell him that. |
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| 17:54.52 | Samot | I've said it already. |
| 17:54.57 | Penguin | He said asterisk doesn't know about his RTP anyway, so it doesn't matter. |
| 17:55.10 | Samot | It does when it's requested to. |
| 17:56.10 | Samot | When you put someone on hold the SDP is update with the media direction sendonly |
| 17:56.25 | Samot | So the phone is still sending RTP.. |
| 17:57.26 | Samot | So he should have directrtpsetup=yes and directmedia=yes on both device peers. |
| 17:57.37 | Samot | Well directrtpsetup in sip.conf |
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| 17:58.05 | Samot | If either of the devices have directmedia=no then Asterisk is going to get involved. |
| 17:59.32 | Samot | stefan27: Both device peers have directmedia=yes? |
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| 21:36.01 | Glanzmann | Hello; I have compiled asterisk 14.0.2 and I have two dynamic ports open, how can I find out which module it opening them? https://pbot.rmdir.de/18N9hiNxq8aRvkmjmxMSQg |
| 21:42.01 | rmudgett | That was asked yesterday on the asterisk-users list: http://lists.digium.com/pipermail/asterisk-users/2016-October/290123.html |
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| 21:43.53 | Glanzmann | rmudgett: Thanks a lot for lifting that mystery. |
| 21:44.25 | Glanzmann | We're randomly unloading modules and grepping the source for dgram to identify the offending module. :-) |
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| 22:37.04 | skirmisha | guys |
| 22:37.16 | skirmisha | is there a way to disable stale nonce auth in asterisk |
| 22:40.42 | skirmisha | ??? |
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