| 00:08.02 | codefreeze | thinks it would be cool if file would send him home, too! |
| 01:12.43 | codefreeze | ***no such luck. codefreeze has to stay & work. Poor codefreeze! |
| 01:15.07 | lmadsen | codefreeze: you can go home now |
| 01:36.19 | codefreeze | shouts "yay!" |
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| 14:35.30 | d1mas | Hello. I have problems on my outbound voip trunk - DTMF just do not work at all. Trying to figure out what is going on. Question: SIP client sends 3 RTP DTMF EV frames in a row, all are "marked". Asterisk relays to the SIP provider only one odf them. (at the same time it relays all 3 RTP DTMF end packets...) Why? |
| 14:36.01 | d1mas | ( i believe 3 frames are just a way to achieve some reliable delivery ) |
| 14:36.25 | jsmith | d1mas: Yeah, you're on the right track there |
| 14:38.12 | d1mas | I have seen dome DTMF issues on a mantis recently where some audio frames (silence?) was added in between DTMF packets to solve some issues. I believe my case is different - when I connect softphone directly to voip provider everything works just fine although softphone does not transmit silence. |
| 14:38.27 | d1mas | only when softphone "proxied" by asterisk - nothing works |
| 14:43.44 | d1mas | so what should I do? :) |
| 14:44.15 | file | how exactly is it proxying? |
| 14:44.29 | file | is a Packet2Packet bridge happening? are you using DTMF based features? |
| 14:44.37 | file | have you tried rfc2833compensate=yes? |
| 14:44.58 | d1mas | file: yes, I'm using features |
| 14:45.18 | d1mas | file: no, I do not have this option explicitly turned on neither off |
| 14:45.46 | file | there is currently a bug open regarding P2P and DTMF with features, it could be what you are encountering |
| 14:46.08 | file | M12212 |
| 14:46.09 | MuffinMan | [acknowledged] [Asterisk] Core/RTP 0012212: [patch] RFC2833 DTMF relaying error when using p2p bridging reported by bamby http://bugs.digium.com/view.php?id=12212 |
| 14:46.23 | d1mas | ohh. |
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| 14:47.55 | d1mas | file: thanks. Will try the patch. |
| 14:54.09 | d1mas | file: great news. the patch does NOT fix my problem. DTMF is still ignored by the remote side |
| 14:54.27 | file | and rfc2833compensate=yes same? |
| 14:54.39 | file | and dtmfmode set to rfc2833 of course? |
| 14:55.12 | d1mas | mode - yes of course. Compansate - no, I did not add it. 1 min... |
| 14:55.44 | Qwell | d1mas: thanks for that patch on the s/DTMF/DIGIT/ thing..not sure how I missed that one - I know I did a grep -r |
| 14:56.16 | d1mas | file: hm... from the sip.conf it looks like compensate stuff will only cause problem if I'm the receiving side.. In my case - I'm oiginating DTMFs... |
| 14:56.33 | d1mas | qwell: oh, no prob, it was easy one, you know :) |
| 14:56.46 | Qwell | yeah, still though |
| 14:57.11 | file | d1mas: try it... never know |
| 14:57.49 | d1mas | file: tried. DTMF are still ignored :( |
| 14:58.23 | file | file an issue with complete console output with core debug, and rtp debug |
| 14:58.40 | d1mas | Can someone please dial 448702233344 and press any digit durign a prompt ? |
| 14:59.04 | d1mas | oh. srry. that was stupid |
| 14:59.16 | d1mas | the problem is using the same provider of course :) |
| 15:04.00 | d1mas | file: should I be making dumps with the patch applied or not ? |
| 15:04.29 | file | with is fine... |
| 15:25.10 | d1mas | aha... packet2packet bridge works fine... |
| 15:32.10 | file | unsurprising. |
| 15:33.19 | d1mas | I cannot grab requested evidences - the server produces A LOT of output. It is live machine... |
| 15:34.44 | file | pokes oej |
| 15:35.01 | oej | hurts. Stop! |
| 15:35.04 | d1mas | file: if I share login/pass of my voip provider (voicetrading) to you - will you be able to verif the problem ? |
| 15:35.22 | oej | and he'll be able to call me internationally in Sweden to test :-) |
| 15:35.28 | d1mas | it is 100% reproducible. I'm using 1.4 branch (week a go or so) |
| 15:35.38 | file | M12323 |
| 15:35.39 | MuffinMan | [new] [Asterisk] Channels/chan_sip/Registration 0012323: Asterisk gives up on registration after receiving 408 Timeout response once reported by voipdealer http://bugs.digium.com/view.php?id=12323 |
| 15:35.43 | file | oej: thoughts? |
| 15:35.45 | file | d1mas: maybe |
| 15:36.09 | oej | file: Well. One thought for 1.4 and another for trunk... |
| 15:36.25 | oej | file: No time really today, more tomorrow. |
| 15:36.46 | file | nods |
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| 15:40.15 | caio1982 | file: can 12323 be related to the latest notes I wrote in 10202? it's regarding a possible registration timeout crashing asterisk in OSX, which I have tried to investigate a little bit, but left it aside since then |
| 15:40.36 | file | Asterisk 1.4 under OSX is very... blargh |
| 15:41.35 | caio1982 | i'm aaaalmost sure it happens with trunk as well, though I didn't test this particular problem using recent checkouts :( |
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| 17:04.37 | putnopvut | M12284 |
| 17:04.39 | MuffinMan | [assigned] [Asterisk] Core-General 0012284: Started to crash every 2-3 hours reported by falves11 http://bugs.digium.com/view.php?id=12284 |
| 17:05.19 | putnopvut | The latest valgrind shows a lot of odd memory accesses within libresolv's dns functions. Is there any sort of required initialization before calling such functions as res_ninit and res_nsearch? |
| 17:05.47 | putnopvut | Google hasn't been especially helpful except to show that apparently there was a bug in opensolaris fixed that involved initialization of the pstat variable. |
| 17:06.00 | putnopvut | (which is what is passed to res_ninit) |
| 17:09.37 | putnopvut | ah ha! |
| 17:09.46 | putnopvut | Finally got some good google data. |
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| 17:28.47 | file | M11823 |
| 17:28.48 | MuffinMan | [ready for testing] [Asterisk] Core/RTP 0011823: RTP gets passed on without early media session reported by SDamm http://bugs.digium.com/view.php?id=11823 |
| 17:29.04 | file | thoughts? |
| 17:34.08 | putnopvut | file: Hmm, I wonder if that has anything to do with an issue that's assigned to me right now... |
| 17:34.15 | file | do tell |
| 17:34.17 | putnopvut | M11854 |
| 17:34.19 | MuffinMan | [feedback] [Asterisk] Applications/app_queue 0011854: The ring option in queue doesn't work if no audio has been played previously on the channel reported by daryll http://bugs.digium.com/view.php?id=11854 |
| 17:34.32 | putnopvut | This person doesn't hear ringing from a queue if they call from an "outside line" |
| 17:34.39 | putnopvut | But if they call internally, it works fine. |
| 17:34.51 | putnopvut | Sounds like it could have to do with this same bug. |
| 17:35.02 | putnopvut | But I haven't time to talk, I need to go. |
| 17:36.45 | file | hrm |
| 17:36.45 | file | i DOUBT IT |
| 17:36.46 | file | argh |
| 17:36.46 | file | I doubt it |
| 17:37.13 | file | hrm, actually maybe |
| 17:37.36 | Juggie | thinks file is going crazy |
| 17:37.54 | jsmith | file: I'm in the "actually maybe" category, fwiw |
| 17:39.37 | file | 11823 is because the two channels are exchanging frames back and forth... in app_queue with the ring option it just sends a ringing indication back and then waits for someone... there is no real frame exchange until answered |
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| 19:00.23 | putnopvut | file: so you think those two are unrelated? |
| 19:06.14 | Qwell | M12273 |
| 19:06.15 | MuffinMan | [feedback] [Asterisk] Core-General 0012273: Receiving RTP PDUs crashes Asterisk core reported by qq12345 http://bugs.digium.com/view.php?id=12273 |
| 19:06.21 | Qwell | app_rtsp isn't even in asterisk O.o |
| 19:06.35 | putnopvut | CLOSE KARMA--!!! |
| 19:06.41 | putnopvut | Actually, I should look first... |
| 19:07.03 | Corydon76-dig | reduces putnopvut's caffeine intake |
| 19:07.06 | Qwell | gonna close and link to the bug with the patch |
| 19:07.16 | Qwell | if mantis will ever finish this filter op... |
| 19:07.20 | Qwell | kicks mantis |
| 19:07.50 | Qwell | which is also closed. nice |
| 19:08.03 | Qwell | M9414 |
| 19:08.04 | MuffinMan | [closed] [Asterisk] Applications/General 0009414: [patch] app_rtsp - playback RTSP media resources reported by sgarcia http://bugs.digium.com/view.php?id=9414 |
| 19:09.17 | Qwell | oh, maybe I should read the comments! |
| 19:09.20 | d1mas | is looking at rfc2833compensate stuff... |
| 19:10.03 | d1mas | just a theoretical question - why is it needed at all? My first impression that "DTMF burst" from 1.2 will be handled Ok by 1.4 even without that option.... |
| 19:11.58 | d1mas | ...at least if 1.2 peroperly "marks" RTP DTMF "begin" and "end" packets |
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| 22:28.23 | Qwell | is there a way to search mantis by username? |
| 22:28.38 | Qwell | rather, search bugs reported by username |
| 22:29.31 | Qwell | nevermind, think I found it. |
| 22:31.19 | Qwell | yay |
| 22:33.37 | Qwell | and back down to 277 |
| 22:33.51 | lmadsen | d1mas: I think it's the other way around... 1.2 not being able to handle 1.4 VLDTMF |
| 22:46.53 | russellb | 1.2 can handle 1.4 dtmf just fine. |
| 23:05.57 | russellb | so um ... i can't replicate this g.722 issue |
| 23:06.20 | Qwell | was just looking at it |
| 23:06.25 | Qwell | I've got nothing :( |
| 23:06.44 | russellb | so in that case |
| 23:06.47 | russellb | 1.6.0-rc1? :) |
| 23:07.50 | Qwell | I don't see any reason why not |
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