Re: Contents of asterisk-dev digest... 2017-02-22 1:00 GMT+07:00 <[email protected]>:
> Send asterisk-dev mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-dev > or, via email, send a message with subject or body 'help' to > [email protected] > > You can reach the person managing the list at > [email protected] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-dev digest..." > > > Today's Topics: > > 1. from [email protected] (sandeep.ananthula) > 2. crashes when bridging opus channels (Moritz Maisel) > 3. Re: crashes when bridging opus channels (Joshua Colp) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 20 Feb 2017 10:27:03 -0800 > From: sandeep.ananthula <[email protected]> > To: "Matt Jordan" <[email protected]>, "wdoekes" > <[email protected]>, "Asterisk Developers" > <[email protected]>, "SANDEEP ANANTHULA" > <[email protected]>, "Santhosh Kumar" > <[email protected]> > Subject: [asterisk-dev] from [email protected] > Message-ID: <[email protected]> > Content-Type: text/plain; charset="us-ascii" > > Hi! > > Have you already seen it? http://goryla.info.pl/aklgcbq. > php?sandeep_ananthula_gmail_com > > > > > > [email protected] > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: <http://lists.digium.com/pipermail/asterisk-dev/ > attachments/20170220/6859f4d5/attachment-0001.html> > > ------------------------------ > > Message: 2 > Date: Tue, 21 Feb 2017 17:13:27 +0100 > From: Moritz Maisel <[email protected]> > To: asterisk-dev <[email protected]> > Subject: [asterisk-dev] crashes when bridging opus channels > Message-ID: > <CANgsjSAUxT+G_avOYRaZdOM9G1=Mtqoa=xdEB55YOYtKeR+dbQ@mail. > gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi, > > we experience reproducable crashes of asterisk with codec_opus. While > asterisks successfully processes a couple of calls (about 8-10) when > bridging two OPUS/48000/2 channels before it crashes, it reproducably > crashes on the first call bridging PCMA/8000 to OPUS/48000/2. > > The setup is asterisk 14.3.0 with bundled pjsip and > codec_opus-14.0_1.1.0-x86_64. The environment is debian 8.7 on amd64 > architecture with kernel version 3.16.0-4-amd64. > > To reduce traffic on the list, I only append parts of the backtrace as well > as the last lines of log output below. I appreciate any suggestions for > debugging this scenario. I'm a bit lost, as the backtrace points into the > codec_opus.so binary blob. Is it recommended to open an issue in the > bugtracker or should we first provide more information on the list? > > Kind regards, > Moritz > > ---------- CLI output --- BEGIN ---------- > -- Called PJSIP/sipgate/sip:[email protected] > -- PJSIP/sipgate-00000003 is ringing > -- PJSIP/sipgate-00000003 is ringing > -- PJSIP/sipgate-00000003 answered PJSIP/proxy-00000002 > -- Channel PJSIP/sipgate-00000003 joined 'simple_bridge' basic-bridge > <d58089f3-c996-4fc7-9678-0fd124b1389e> > -- Channel PJSIP/proxy-00000002 joined 'simple_bridge' basic-bridge > <d58089f3-c996-4fc7-9678-0fd124b1389e> > > 0x7f1c200257f0 -- Probation passed - setting RTP source address to > 217.10.77.244:26918 > ---------- CLI output --- END ---------- > > ---------- backtrace --- BEGIN ---------- > Thread 1 (Thread 0x7f60fe49f700 (LWP 27298)): > #0 0x00007f6147ad1952 in ?? () from /usr/lib/asterisk/modules/ > codec_opus.so > #1 0x00007f6147ac5f26 in ?? () from /usr/lib/asterisk/modules/ > codec_opus.so > #2 0x00000000005f436e in ast_translate () > #3 0x00000000004bf15c in ast_write () > #4 0x0000000000483553 in bridge_channel_internal_join () > #5 0x000000000046d49e in ?? () > #6 0x00000000005fa32a in ?? () > #7 0x00007f615df57064 in start_thread () from > /lib/x86_64-linux-gnu/libpthread.so.0 > #8 0x00007f615d00662d in clone () from /lib/x86_64-linux-gnu/libc.so.6 > ---------- backtrace --- END ---------- > > -- > sipgate GmbH - Gladbacher Str. 74 - 40219 D?sseldorf > HRB D?sseldorf 39841 - Gesch?ftsf?hrer: Thilo Salmon, Tim Mois > Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 > > www.sipgate.de - www.sipgate.co.uk > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: <http://lists.digium.com/pipermail/asterisk-dev/ > attachments/20170221/78e24956/attachment-0001.html> > > ------------------------------ > > Message: 3 > Date: Tue, 21 Feb 2017 12:23:37 -0400 > From: Joshua Colp <[email protected]> > To: [email protected] > Subject: Re: [asterisk-dev] crashes when bridging opus channels > Message-ID: > <1487694217.1825127.888139120.6b5d5...@webmail.messagingengine.com > > > Content-Type: text/plain; charset="utf-8" > > On Tue, Feb 21, 2017, at 12:13 PM, Moritz Maisel wrote: > > Hi, > > > > we experience reproducable crashes of asterisk with codec_opus. While > > asterisks successfully processes a couple of calls (about 8-10) when > > bridging two OPUS/48000/2 channels before it crashes, it reproducably > > crashes on the first call bridging PCMA/8000 to OPUS/48000/2. > > > > The setup is asterisk 14.3.0 with bundled pjsip and > > codec_opus-14.0_1.1.0-x86_64. The environment is debian 8.7 on amd64 > > architecture with kernel version 3.16.0-4-amd64. > > > > To reduce traffic on the list, I only append parts of the backtrace as > > well > > as the last lines of log output below. I appreciate any suggestions for > > debugging this scenario. I'm a bit lost, as the backtrace points into the > > codec_opus.so binary blob. Is it recommended to open an issue in the > > bugtracker or should we first provide more information on the list? > > Please file an issue on the issue tracker[1] and use codec_opus as the > component. We'll triage and ask for the needed information there. > > [1] https://issues.asterisk.org/jira > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > > End of asterisk-dev Digest, Vol 151, Issue 14 > ********************************************* >
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