Yes we can reproduce this crash scenario by running calls between portsip and Xlite soft phones. The issue we have observed is CODEC translation between iLBC and alaw with following warning messages,
[Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to 0x400 (ilbc) [Sep 2 15:59:53] WARNING[24418] chan_sip.c: Asked to transmit frame type ilbc, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to 0x400 (ilbc) [Sep 2 15:59:53] WARNING[24418] chan_sip.c: Asked to transmit frame type ilbc, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to 0x400 (ilbc) [Sep 2 15:59:53] WARNING[24418] chan_sip.c: Asked to transmit frame type ilbc, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: No path to translate from SIP/18252-0002d010 to SIP/18203-0002d01e [Sep 2 15:59:53] WARNING[24418] channel.c: Can't make SIP/18252-0002d010 and SIP/18203-0002d01e compatible [Sep 2 15:59:53] WARNING[24418] features.c: Bridge failed on channels SIP/18252-0002d010 and SIP/18203-0002d01e We can reproduce the problem as below, 1. Call between Xlite(iLBC) to portsip(G711), RTP through asterisk. 2. portsip attended transfer the call to another portsip client 3. on complete transfer asterisk crashes (then started by safe_asterisk) with above warning. FYI, we have not installed asterisk with iLBC support. We will try to upgrade asterisk and try to reproduce this scenario. Regards Rajib ------------------------------ Message: 13 Date: Wed, 4 Sep 2013 09:28:12 -0500 From: Rusty Newton <rnew...@digium.com> Subject: Re: [asterisk-users] Asterisk crash To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <cagwdfcvp_vtcpwgw_0zspcqxiflqsrhthnl4c0zfuf92dud...@mail.gmail.com> Content-Type: text/plain; charset=UTF-8 On Tue, Sep 3, 2013 at 4:17 AM, Deka, Rajib IN MAA SL <rajib.d...@siemens.com> wrote: > In our lab asterisk has crashed due to some unknown reason and it has been > restarted by safe_asterisk service. But before crash we can see lots of > below log entry (asterisk version 1.8.9.3). That is quite old. Lots of bugs (and several security issues) have been fixed since then. Try the latest in the 1.8 branch. For the crash , follow the instructions here https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace and gather a backtrace after recompiling with the required options. (preferably after upgrading to the latest 1.8, as there may have been improvmen > Sep 3 07:55:21] WARNING[16287] res_rtp_asterisk.c: RTP Transmission error > of packet to [2002:c117:a683::c117:a683]:20940: Address family not supported > by protocol > > chan_sip.c: Purely numeric hostname, and not a peer--rejecting! These messages alone don't show the whole picture. https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Collect a log with VERBOSE and DEBUG turned up to level 5, SIP debug turned on, and pastebin that. I'd wait until after you test with the latest in 1.8 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users