search for: ast_bridge_call

...2 formats Feb 2 16:53:10 DEBUG[4218]: channel.c:3189 ast_do_masquerade: Released clone lock on 'AsyncGoto/SIP/113-08674628<ZOMBIE>' Feb 2 16:53:10 DEBUG[4218]: channel.c:3198 ast_do_masquerade: Done Masquerading SIP/113-08674628 (6) Feb 2 16:53:10 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels SIP/111-086497c8 and AsyncGoto/SIP/113-08674628<ZOMBIE> Feb 2 16:53:10 DEBUG[4218]: app_dial.c:1636 dial_exec_full: Exiting with DIALSTATUS=ANSWER. -- Executing Set("SIP/111-086497c8", "DYNAMIC_FEATURES=") in new stack -- Executing Goto(...

...Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1) Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge: Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1 Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read from SIP/rafal-89b1 (1,42) -- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1 Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 50 (2), at 10.2.20.65 Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/ra...

...e (c0=0xb659fcd0, c1=0x9455ca0, config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c, bridge_end= {tv_sec = 0, tv_usec = 0}) at channel.c:3260 #9 0x080655fd in ast_channel_bridge (c0=0xb659fcd0, c1=0x9455ca0, config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c) at channel.c:3524 #10 0xb78fad29 in ast_bridge_call (chan=0xb659fcd0, peer=0x9455ca0, config=0xb6c4feb0) at res_features.c:1319 #11 0xb7099301 in dial_exec_full (chan=0xb659fcd0, data=0xb6c4feb0, peerflags=0xb6c50568) at app_dial.c:1577 #12 0xb7097dc5 in dial_exec (chan=0xb7ed1900, data=0xb7ed1900) at app_dial.c:1619 #13 0x0808e445 in pbx_extension_...

...9;' in context 'crystal-sip' -- Playing 'pbx-invalid' (language 'en') Sep 21 10:30:30 WARNING[52987]: file.c:550 ast_readaudio_callback: Failed to write frame -- Stopped music on hold on Local/3044@local-4fee,2 Sep 21 10:30:30 WARNING[52987]: res_features.c:450 ast_bridge_call: Bridge failed on channels Local/3044@local-4fee,2 and SIP/3044-ea92 == Spawn extension (macro-sipline, s, 1) exited non-zero Why doesn't ast_bridge_call do it's thing __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

If I have in my dialplan [AgentQ] exten => _XX.,1,Dial(Sip/{$exten},120,g) exten => _XX.,2,NoOP(here we are) where [AgentQ] is called by the queue command to a member added by addqueuemember(Local/99@AgentQ) why don't I get to the NoOp if the agent hangs up during the announcement message (to the agent) ? I see in the app_dial.c program that the "g" flag is tested thus:

....so.6 #2 0x080675a7 in ast_waitfor_nandfds (c=0xb7469b80, n=2, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb7469b4c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0xb22bf9a8, c1=0xa2ae648, config=0xb746a7a0, fo=0xb7469c40, rc=0xb7469c44) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0xb22bf9a8, peer=0xa2ae648, config=0xb746a7a0) at res_features.c:1365 #5 0x005a40ba in dial_exec_full (chan=0xb22bf9a8, data=Variable "data" is not available. ) at app_dial.c:1633 #6 0x005a6a33 in dial_exec (chan=0xfffffffc, data=0x7fffffff) at app_dial.c:1680 #7 0x08090bad in pb...

...; then dials the user "33" 4. user "ua1" and "33" are connected 5. Now when "ua1" presses the feature code "**" to redirect user "33" to same conference room 300, there is error thrown on Asterisk console that "res_features.c:1415 ast_bridge_call: Bridge failed on channels SIP/ua1-ac750040 and AsyncGoto/Local/33@nway-conf-dest-7ecf,1<ZOMBIE>" Here is my dial plan: [manu] exten => ca1,1,Dial(SIP/ca1,,wWtTkKrR) [nway-conf] exten => _.,1,Answer exten => _.,n,Set(CONFNO=${EXTEN}) exten => _.,n,Set(MEETME_EXIT_C...

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,

...C_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) apps/app_valetparking.c: In function `valetunpark_call': apps/app_valetparking.c:666: error: structure has no member named `allowdisconnect' apps/app_valetparking.c:671: warning: implicit declaration of function `ast_bridge_call' apps/app_valetparking.c: In function `load_module': apps/app_valetparking.c:760: warning: implicit declaration of function `__use_ast_pthread_create_instead__' [root@localhost asterisk]# [root@localhost asterisk]#

...ung up too (which is undesired). We like to keep the channel up and do something else in the dialplan. As I'm not too familiar with the asterisk internals I have problems to interprete the sources, but I think it might be in the method: static int changrab_exec in Line 131 which is res = ast_bridge_call(chan, newchan, &config); ast_hangup(newchan); I have no clue if ast_bridge_call blocks until some statechange happens and ast_hangup hangs up which would lead to the undesired behavior!? Maybe someone can help me understanding some basics about these asterisk functions!? -- Knud A. M?ll...

...there is an alternative called app_changrab. (http://www.freeswitch.org/asterisk_stuff/app_changrab.c) First tests had shown that app_changrab worked well despite CDR logging. Changrab uses mainly (I have very little understanding of the asterisk internals therefore these are just assumptions...) ast_bridge_call to connect the channels. But it doesnt connect the actual channel, but creates a masqueraded (Zombie?) channel that is handed to the ast_bridge_call command. I have seen in the manager interface that the Zombie Channel had the same 'uniqueid' and is hungup instantly after bridging the...

Ist there any possibility to use the funktion "consultative transfer"? ( have 2 ISDN-pones attached to the hfc-nt card, configured as zap) With the "#"-key it ist possible to park the call or to make a "blind transfer" at the moment. I have activated threewaycalling in the zapata.conf file: ; internal S0 bus (first hfc/s card): context=local signalling =

...d air. Does anyone have any ideas on how to fix this or a better way to implement this? Output when the call is dropped: -- Channel 0/3, span 1 got hangup request -- User disconnected -- Stopped music on hold on Local/299@default-d64b,2 Nov 13 16:21:26 WARNING[12709]: res_features.c:1374 ast_bridge_call: Bridge failed on channels Local/299@default-d64b,2 and Zap/3-1 -- Hungup 'Zap/3-1' -- Local/299@default-d64b,1 answered SIP/7960A-Gary1-63f2 -- Stopped music on hold on SIP/7960A-Gary1-63f2 Dialplan: exten => 299,1,Wait(0.2) exten => 299,n,Dial(Zap/3/${CELLNUMBER}|60|gmM(...

...(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0); ast_channel_setoption(peer,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0); } + if (announce && announcemsg) + { + res = ast_streamfile(peer,announcemsg,peer->language); + res = ast_waitstream(peer,""); + } res = ast_bridge_call(chan, peer, allowredir_in, allowredir_out, allowdisconnect | clearchannel); if (clearchannel) {

...buted device state. Initialize pubsubflags to 0 so res_jabber doesn't think there is already an XMPP connection sending device state. Also clean up CLI commands a bit. (Closes issue #18272. Reported by klaus3000. Patched by Marquis42) * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of setting peer->cdr = NULL, set it to not post. (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares) * Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue. (Closes iss...

...buted device state. Initialize pubsubflags to 0 so res_jabber doesn't think there is already an XMPP connection sending device state. Also clean up CLI commands a bit. (Closes issue #18272. Reported by klaus3000. Patched by Marquis42) * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of setting peer->cdr = NULL, set it to not post. (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares) * Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue. (Closes iss...

...nnel.c:2693 ast_channel_make_compatible: No path to translate from SIP/53061-92e0(4) to SIP/10.10.10.10-78fa(1024) Jun 26 17:53:52 WARNING[14248]: channel.c:3520 ast_channel_bridge: Can't make SIP/53061-92e0 and SIP/10.10.10.10-78fa compatible Jun 26 17:53:52 WARNING[14248]: res_features.c:1381 ast_bridge_call: Bridge failed on channels SIP/53061-92e0 and SIP/10.10.10.10-78fa == Spawn extension (iaxtest, 2144466715, 3) exited non-zero on 'SIP/53061-92e0' The call drops. If I enable ILBC codec with Asterisk, here is what I get: == Forcing Marker bit, because SSRC has changed Jun 26 17:56:28...

...ast_channel_make_compatible: No path to translate from SIP/sip.westend.com-082fd1b8(8) to SIP/xxx-3ef8(1) Nov 23 16:27:35 WARNING[1061908]: channel.c:2633 ast_channel_bridge: Can't make SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 compatible Nov 23 16:27:35 WARNING[1061908]: res_features.c:358 ast_bridge_call: Bridge failed on channels SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 == Spawn extension (macro-enumcall, s, 211) exited non-zero on 'SIP/sip.westend.com-082fd1b8' in macro 'enumcall' == Spawn extension (xxx, 911879, 7) exited non-zero on 'SIP/sip.westend.com-082fd1b8'...

...se of Asterisk 1.8.1.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.1.1 resolves two issues reported by the community since the release of Asterisk 1.8.1. * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of setting peer->cdr = NULL, set it to not post. (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares) * Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue. (Clos...