search for: ast_channel_free

...4569-1' status is 'ANSWER' -- Stopped music on hold on IAX2/192.168.0.231:4569-1 -- Hungup 'IAX2/192.168.0.231:4569-1' == IAX2/192.168.0.231:4569-1 got tired of being parked -- Hungup 'IAX2/192.168.0.231:4569-1' Jan 24 13:43:26 WARNING[24727]: channel.c:897 ast_channel_free: Unable to find channel in list pbx*CLI> Disconnected from Asterisk server The back trace has a similar message about channel.c #6 0x080616bd in ast_channel_free (chan=0x9932c48) at channel.c:864 cur = Variable "cur" is not available. Has anyone run into this before? I cann...

...email has been recorded, if the # key wasn't pressed, if the channel still exists, and if it's a Zap channel. I couldn't see a way to do this with AGI. Question: is this safe? I used a soft hangup because the channel is controlled by another thread. I also modified channel.c so that ast_channel_free() sets chan to NULL after it's freed, just in case. Is there anything else I should be aware of? The code seems to work in my testing, resulting in a proper hangup right after the voicemail has been recorded. I'm not up on my Asterisk internals, so I'm not totally confident about th...

...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? -- Regards, Russell...

.../ld-linux.so.2 #1 0x006267a5 in raise () from /lib/tls/libc.so.6 #2 0x00628209 in abort () from /lib/tls/libc.so.6 #3 0x0065a71a in __libc_message () from /lib/tls/libc.so.6 #4 0x00660fbf in _int_free () from /lib/tls/libc.so.6 #5 0x0066133a in free () from /lib/tls/libc.so.6 #6 0x080615f3 in ast_channel_free (chan=0xb7904e00) at channel.c:959 #7 0x08062bd7 in ast_hangup (chan=0xb7904e00) at channel.c:1392 #8 0x001aa4fb in conf_free (conf=0xb7901d98) at app_meetme.c:789 #9 0x001acfa3 in conf_run (chan=0x96e94a0, conf=0xb7901d98, confflags=4224, optargs=0xb7ddcd4c) at app_meetme.c:1607 #10 0x001aeb26...

.../ld-linux.so.2 #1 0x002487a5 in raise () from /lib/tls/libc.so.6 #2 0x0024a209 in abort () from /lib/tls/libc.so.6 #3 0x0027ca1a in __libc_message () from /lib/tls/libc.so.6 #4 0x002834c0 in _int_free () from /lib/tls/libc.so.6 #5 0x0028363a in free () from /lib/tls/libc.so.6 #6 0x08061e15 in ast_channel_free (chan=0x9ea42c0) at channel.c:972 #7 0x0806896d in ast_hangup (chan=0x9ea42c0) at channel.c:1403 #8 0x08092171 in __ast_pbx_run (c=0x9ea42c0) at pbx.c:2487 #9 0x08093a2c in pbx_thread (data=0x0) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #11 0x002e9c3e in clone...

Hi, I'm using the following setup : Alice ---- IPPhone ------<LAN>----- Media gateway ----<PSTN> ------- Phone ---- Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably,