search for: ast_writ

Sometimes in huge call volume am facing this type of error, [Jun 4 08:42:46] WARNING[8459][C-000079fa]: channel.c:5075 ast_write: Codec mismatch on channel Local/8038 at xss-call-out-00004774;1 setting write format to slin from ulaw native formats (ulaw) [Jun 4 08:43:04] WARNING[8285][C-000079da]: channel.c:5075 ast_write: Codec mismatch on channel Local/6513 at xss-call-out-00004775;1 setting write format to slin from ula...

...nslate.h> /* Needed for bugfix */ >75c76 >< if (ast_set_write_format(chan, AST_FORMAT_SLINEAR)) { >--- > > if ((chan->nativeformats & AST_FORMAT_SLINEAR) && > ast_set_write_format(chan, AST_FORMAT_SLINEAR)) { >128c129,142 >< ast_write(chan, &ps->f); >--- > > > > // Now, we have a finished SLINEAR frame that we need to > translate, IF > > // the channel doesn't support SLINEAR. Otherwise, we need to just > > // write the SLINEAR frame. > > if (!(chan->na...

...e 4 numbers on my Asterisk 1.8. 3 work perfectly, the 4.th not. I'm not sure, when it finish to work, since a month ago it runs without any problem... Well, if I will be called on this number I can't hear anything and in Asterisk I see these: [Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-000001d1 setting write format to g729 from alaw native formats 0x8 (alaw) [Jul 15 18:59:55] WARNING[8752]: channel.c:5254 set_format: Unable to find a codec translation path from 0x8 (alaw) to 0x100 (g729) [Jul 15 18:59:55] WARNING[8752]: chan_sip.c:67...

...yer(pippo.mp3) What I think it is happening is that <Background> plays its GSM file setting the write format to 2 (GSM), <Ringing> starts the generator setting the write format to 64 (Linear PCM) saving the old write format 2; MP3Player starts, set the write format to 64. When it calls ast_write to write the first stream, ast_release_generator is called and restores the write format to 2 (GSM); further writes produce codec errors since MP3Players writes Linear PCM frames. I extracted some debug log: Jun 5 01:55:33 DEBUG[1158913328]: File chan_sip.c, Line 1327 (sip_alloc): Allocating...

...nt 100000 reached on ao2 object 0x26bffc0 (0) Got 19 backtrace records #0: [0x45d229] /usr/sbin/asterisk(__ao2_ref+0x1a9) [0x45d229] #1: [0x526ce6] /usr/sbin/asterisk(ast_frdup+0x116) [0x526ce6] #2: [0x5fa616] /usr/sbin/asterisk(ast_translate+0x306) [0x5fa616] #3: [0x4bf16b] /usr/sbin/asterisk(ast_write+0x104b) [0x4bf16b] #4: [0x7efeb578230b] /usr/lib/asterisk/modules/res_musiconhold.so(+0x430b) [0x7efeb578230b] #5: [0x4b5b52] /usr/sbin/asterisk() [0x4b5b52] #6: [0x4c259c] /usr/sbin/asterisk() [0x4c259c] #7: [0x4c4a45] /usr/sbin/asterisk() [0x4c4a45] #8: [0x7efeb578478d] /usr/lib/asterisk/mo...

...282) #3: [0x531a99] main/frame.c:196 ast_frisolate() (0x531a76+23) #4: [0x60be51] main/translate.c:459 ast_trans_frameout() (0x60bd6e+E3) #5: [0x60be75] main/translate.c:464 default_frameout() #6: [0x60c46a] main/translate.c:579 ast_translate() (0x60c192+2D8) #7: [0x4c0bf1] main/channel.c:5290 ast_write() (0x4bfb3e+10B3) #8: [0x7fdef8345486] res/res_musiconhold.c:455 moh_files_generator() #9: [0x4ba212] main/channel.c:3014 generator_force() #10: [0x4bc23d] main/channel.c:3872 __ast_read() #11: [0x4be29b] main/channel.c:4399 ast_read() (0x4be27e+1D) #12: [0x4b6312] main/channel.c:1568 ast_saf...

...0 0x41154918 in calc_timestamp (p=0x816b710, ts=0, f=0x424eef24) at chan_iax2.c:2896 #1 0x41153119 in iax2_send (pvt=0x816b710, f=0x424eef24, ts=32, seqno=-1, now=0, transfer=0, final=32) at chan_iax2.c:3091 #2 0x41166e17 in iax2_write (c=0x20, f=0x424eef24) at chan_iax2.c:3551 #3 0x0805cd41 in ast_write (chan=0x816bd90, fr=0x424eef24) at channel.c:1634 #4 0x080610e3 in ast_activate_generator (chan=0x816bd90, gen=0x407ca918, params=0x20) at channel.c:1554 #5 0x407c725e in ast_moh_start (chan=0x0, class=0x20 <Address 0x20 out of bounds>) at res_musiconhold.c:598 #6 0x41804e3d in dial_exec...

...ess_message: Manager received command 'PlayDTMF' [Oct 2 11:14:46] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread -1216881776 Blocking 'SIP/1000-0a292360', already blocked by thread -1217414256 in procedure ast_waitfor_nandfds [Oct 2 11:14:47] DEBUG[29533]: channel.c:3341 ast_write: Deadlock avoided for write to channel 'SIP/1000-0a292360' [Oct 2 11:14:47] DEBUG[30054]: manager.c:2776 process_message: Manager received command 'PlayDTMF' [Oct 2 11:14:47] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread -1216881776 Blocking 'SIP/1000-0a292360'...

...ess_message: Manager received command 'PlayDTMF' [Oct 2 11:14:46] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread -1216881776 Blocking 'SIP/1000-0a292360', already blocked by thread -1217414256 in procedure ast_waitfor_nandfds [Oct 2 11:14:47] DEBUG[29533]: channel.c:3341 ast_write: Deadlock avoided for write to channel 'SIP/1000-0a292360' [Oct 2 11:14:47] DEBUG[30054]: manager.c:2776 process_message: Manager received command 'PlayDTMF' [Oct 2 11:14:47] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread -1216881776 Blocking 'SIP/1000-0a292360'...

...ess_message: Manager received command 'PlayDTMF' [Oct 2 11:14:46] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread -1216881776 Blocking 'SIP/1000-0a292360', already blocked by thread -1217414256 in procedure ast_waitfor_nandfds [Oct 2 11:14:47] DEBUG[29533]: channel.c:3341 ast_write: Deadlock avoided for write to channel 'SIP/1000-0a292360' [Oct 2 11:14:47] DEBUG[30054]: manager.c:2776 process_message: Manager received command 'PlayDTMF' [Oct 2 11:14:47] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread -1216881776 Blocking 'SIP/1000-0a292360'...

...ess_message: Manager received command 'PlayDTMF' [Oct 2 11:14:46] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread -1216881776 Blocking 'SIP/1000-0a292360', already blocked by thread -1217414256 in procedure ast_waitfor_nandfds [Oct 2 11:14:47] DEBUG[29533]: channel.c:3341 ast_write: Deadlock avoided for write to channel 'SIP/1000-0a292360' [Oct 2 11:14:47] DEBUG[30054]: manager.c:2776 process_message: Manager received command 'PlayDTMF' [Oct 2 11:14:47] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread -1216881776 Blocking 'SIP/1000-0a292360'...

...ar 26 12:11:32] == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'SIP/7051-00000254' in macro 'stdexten' [Mar 26 12:11:32] == Spawn extension (ddi, 7003, 1) exited non-zero on 'SIP/7051-00000254' [2015-03-26 12:11:32] WARNING[1561][C-0000015c]: channel.c:5070 ast_write: Codec mismatch on channel SIP/pabx-e1-00000252 setting write format to slin from alaw native formats (alaw) [Mar 26 12:11:40] -- Channel SIP/pabx-e1-00000252 left 'simple_bridge' basic-bridge <f4fb9d99-24b9-4d3c-9b63-41a1b84484b2> [Mar 26 12:11:40] == Spawn extension (macro-st...

...d of %d bytes: %s\n", res, len, strerror(errno)); @@ -413,7 +414,8 @@ f.mallocd = 0; f.datalen = res; f.samples = res / 2; - f.data = buf + AST_FRIENDLY_OFFSET / 2; + /* XXX SEGFAULT f.data = buf + AST_FRIENDLY_OFFSET / 2;*/ + f.data = buf; f.offset = AST_FRIENDLY_OFFSET; if (ast_write(chan, &f)< 0) { ast_log(LOG_WARNING, "Failed to write frame to '%s': %s\n", chan->name, strerror(errno));

...:06.402 H225 Caller:811c000 H225 Received connect PDU. -- H323:15452 answered SIP/1800-6411 Segmentation fault (gdb) bt #0 ast_smoother_feed (s=0x57e0880, f=0x810bf78) at frame.c:72 #1 0x4617d71a in oh323_write (c=0x8110068, f=0x810bf78) at chan_oh323.c:1379 #2 0x080584da in ast_write (chan=0x8110068, fr=0x810bf78) at channel.c:1386 #3 0x0805a63e in ast_channel_bridge (c0=0x810e868, c1=0x8110068, flags=0, fo=0x47ee8ea4, rc=0x47ee8ea8) at channel.c:2278 #4 0x412024f3 in ast_bridge_call (chan=0x810e868, peer=0x8110068, allowredirect_in=0, allowredirect_out=0, allowdisconnect=0)...

...ted to a no-name btusb dongle. Nothing unexpected shows up in the Asterisk console: -- Executing [*00 at house-phones:4] Dial("SIP/200-00000037", "MOBILE/JS6021/4444,60,rTK") in new stack -- Called MOBILE/JS6021/4444 [2012-04-30 19:13:15] WARNING[25326]: channel.c:4913 ast_write: Codec mismatch on channel Mobile/JS6021-6e5b setting write format to alaw from slin native formats 0x40 (slin) -- Mobile/JS6021-6e5b is making progress passing it to SIP/200-00000037 -- Mobile/JS6021-6e5b answered SIP/200-00000037 -- Executing [sms at from-stocksy-orange:1] Verbose(&q...

Are there any adapters that would allow me to connect asterisk to wifi or we are not there yet? I have Digium adapter S101i that was discontinued but similar device that would connect to wifi network and a cell phone would be handy. -- Joseph

...t;bt" and send me the output > of it. if it is of use, here it is (from asterisk v.0.5.0) ----------------------------- (gdb) bt #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72 #1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504 #2 0x0805884f in ast_write (chan=0x8214488, fr=0x5de5c4a8) at channel.c:1385 #3 0x0805afa1 in ast_channel_bridge (c0=0x5de5c4a8, c1=0x0, flags=0, fo=0x6ef20e50, rc=0x6ef20e54) at channel.c:2262 #4 0x418bdd7a in ast_bridge_call (chan=0x5de5ed98, peer=0x8214488, allowredirect_in=0, allowredirect_out=0, allowdisconnect=0) at...

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to