Russell Brown wrote: > 100% repeatable (for me). > > Sip phone A calls Sip phone B. > > Either Sip phone A or B does #700. The party that keyed #700 gets the > parked announcement (eg 701) and the other party get MOH. There is > still an audio channel between the two SIP phones at this point. > > When the party that typed #700 hangs up, Asterisk crashes. > > This has been working in previous 1.4's (but not 1.4.10) and I havn't > changed my parking config. > > Here's what comes up on the console as it crashes. > > -- <SIP/Testsnom-00709570> Playing 'digits/7' (language 'en') > -- <SIP/Testsnom-00709570> Playing 'digits/0' (language 'en') > -- <SIP/Testsnom-00709570> Playing 'digits/1' (language 'en') > -- Added extension '701' priority 1 to parkedcalls > -- Stopped music on hold on SIP/115-0072f7a0 > == SIP/115-0072f7a0 got tired of being parked > == Spawn extension (macro-stdsip, s, 6) exited non-zero on > 'SIP/Testsnom-00709570' in macro 'stdsip' > == Spawn extension (macro-stdsip, s, 6) exited non-zero on > 'SIP/Testsnom-00709570' > *** glibc detected *** asterisk: double free or corruption (out): > 0x00002ab23a7ed9f0 *** > ======= Backtrace: ========= > /lib/libc.so.6[0x2ab23a620733] > /lib/libc.so.6(__libc_free+0x84)[0x2ab23a6208b4] > asterisk(ast_channel_free+0xf6)[0x438fa6] > asterisk(ast_hangup+0x35a)[0x43b84a] > /usr/lib/asterisk/modules/res_features.so[0x2aaaab8298c0] > asterisk[0x4a719c] > /lib/libpthread.so.0[0x2ab239da23ca] > /lib/libc.so.6(__clone+0x6d)[0x2ab23a67f55d] > ======= Memory map: ======== > > Anyone got any ideas? >
No ideas but I also found something last night that could be related. I have the following: Home SIP <-SIP-> Home Asterisk <-IAX-> Work Asterisk Calls from Home SIP to Home Asterisk (like vms) sound fine. Calls from Home SIP through Home Asterisk to Work Asterisk sound horrible on the SIP side. I don't know how it sounded on the Work side since I was checking voicemail. It occurred on every call all the time with 1.4.10 and 1.4.10.1. Reverting Home Asterisk to 1.4.9 solved the problem. Maybe something got "fixed" in chan_sip in 1.4.10? -Dave _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
