search for: ast_channel_bridg

...park = yes pickupexten = *8 [featuremap] atxfer => *2 blindxfer => # disconnect => *0 automon => *1 and when I press *2 console says something like this: Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 42 (*), 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/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) -- Attempt...

...originating side ends the call first (unexpectedly stops at priority 1 in macro): (normal call setup and progress not shown - I show everything after hangup) *CLI> DEBUG[30737]: File rtp.c, Line 739 (ast_rtp_raw_write): Difference is 10888, ms is 1381 DEBUG[30737]: File channel.c, Line 2015 (ast_channel_bridge): Didn't get a frame from channel: SIP/2203-751b DEBUG[30737]: File channel.c, Line 2083 (ast_channel_bridge): Bridge stops bridging channels SIP/2203-751b and SIP/2205-12f1 == Spawn extension (intern-post, 2205, 1) exited non-zero on 'SIP/2203-751b' -- Executing Macro("SIP/...

...ing Aug 13 10:19:05 DEBUG[245774]: chan_sip.c:799 __sip_ack: Acked pending invite 102 Aug 13 10:19:05 DEBUG[245774]: chan_sip.c:4411 build_route: build_route: Contact hop: <sip:xlite1@Ip of Xlite1:5060> -- SIP/xlite1-89a7 answered OH323/R27469 Aug 13 10:19:05 DEBUG[524304]: channel.c:2613 ast_channel_bridge: Got a FRAME_CONTROL (4) frame on channel SIP/xlite1-89a7 Aug 13 10:19:05 DEBUG[524304]: channel.c:2675 ast_channel_bridge: Bridge stops bridging channels OH323/R27469 and SIP/xlite1-89a7 Aug 13 10:19:05 DEBUG[524304]: res_features.c:422 ast_bridge_call: Read from SIP/xlite1-89a7 (4,4) Aug 13 1...

...--PSTN. As endpoint I had tested another asterisk box (with a FXS), ciscoATA, cisco1750 and cisco827. The problem is the same with all. Eduardo debug when I call a PSTN number from the ATA186: ========================================== Oct 17 19:20:02 DEBUG[1605650]: File channel.c, Line 2210 (ast_channel_bridge): Didn't get a frame from channel: SIP/atasuporte-1413 Oct 17 19:20:02 DEBUG[1605650]: File channel.c, Line 2278 (ast_channel_bridge): Bridge stops bridging channels SIP/atasuporte-1413 and Zap/1-1 Oct 17 19:20:02 DEBUG[1605650]: File chan_zap.c, Line 1593 (zt_hangup): Hangup: channel: 1 index...

...rom demo and put it on exten 600) but what wouldn't properly work is is sip to iax bridging linphone -> * -> firefly More specifically, firefly rings properly, but when I press Accept it just keeps ringing, and finally * tells me that linphone didn't send any frames: channel.c:2655 ast_channel_bridge: Didn't get a frame from channel: SIP/10-e8bd Jan 12 21:55:02 DEBUG[22661]: channel.c:2725 ast_channel_bridge: Bridge stops bridging channels SIP/10-e8bd and IAX2/100/2 Doing my tcpdumps I checked that there's really no data sent by linphone, while nothing is dropped by firewalls either....

Hello all, This new version (0.6.1) of asterisk-oh323 fixes the "one-way audio" problem of the previous release. Download from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael.

...force (data=0x9455ca0) at channel.c:1405 #7 0x08061c50 in ast_read (chan=0x9455ca0) at channel.c:1857 #8 0x08069293 in ast_generic_bridge (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=0xb6c5056...

....... Thread 6 (process 20135): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002dfdf4 in poll () from /lib/tls/libc.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 no...

...to Asterisk which then dials out a local number using the Digium E100P. We have purchased the G729 codec licenses from Digium and loaded them into Asterisk successfully. However, the call drops immediately after being answered with the debug error message saying something like: "channel.c:2646 ast_channel_bridge: Didn't get a frame from channel: SIP/6568543197-9af5" Any idea why this may be happening? Here is the debug log: Oct 4 00:53:41 DEBUG[1083546560]: chan_sip.c:5269 check_user_full: Setting NAT on RTP to 4 Oct 4 00:53:41 DEBUG[1083546560]: chan_sip.c:7087 handle_request: Check for res...

...uration 0 ms [Jul 7 14:41:11] DTMF[9852]: channel.c:2436 __ast_read: DTMF begin emulation of '*' with duration 100 queued on IAX2/6001-2 [Jul 7 14:41:11] DEBUG[9852]: channel.c:4163 ast_generic_bridge: Got DTMF begin on channel (IAX2/6001-2) [Jul 7 14:41:11] DEBUG[9852]: channel.c:4446 ast_channel_bridge: Bridge stops bridging channels IAX2/6001-2 and SIP/6002-08383a78 Rx-Frame Retry[ No] -- OSeqno: 009 ISeqno: 012 Type: DTMF_E Subclass: 2 Timestamp: 27467ms SCall: 08434 DCall: 00002 [127.0.0.1:40310] [Jul 7 14:41:11] DEBUG[1268]: chan_iax2.c:7016 socket_process: Received packet 9, (1, 50...

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,

...to 565606422 due to a source change [Jan 12 23:13:55] DEBUG[8717]: channel.c:4610 ast_generic_bridge: Got DTMF begin on channel (IAX2/419-13088) [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2118 ast_rtp_new_source: Setting the marker bit due to a source update [Jan 12 23:13:55] DEBUG[8717]: channel.c:4927 ast_channel_bridge: Bridge stops bridging channels IAX2/419-13088 and SIP/xtreme-00000639 [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2130 ast_rtp_change_source: Changing ssrc from 565606422 to 226872656 due to a source change [Jan 12 23:13:56] DEBUG[8717]: channel.c:4610 ast_generic_bridge: Got DTMF end on channel (IAX2/4...

...ived on SIP/1000-0a292360 [Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read: DTMF begin passthrough '#' on SIP/1000-0a292360 [Oct 2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF begin on channel (SIP/1000-0a292360) [Oct 2 11:09:20] DEBUG[29533]: channel.c:5150 ast_channel_bridge: Bridge stops bridging channels SIP/1000-0a292360 and SIP/1001-0a026408 [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rt...

...ived on SIP/1000-0a292360 [Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read: DTMF begin passthrough '#' on SIP/1000-0a292360 [Oct 2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF begin on channel (SIP/1000-0a292360) [Oct 2 11:09:20] DEBUG[29533]: channel.c:5150 ast_channel_bridge: Bridge stops bridging channels SIP/1000-0a292360 and SIP/1001-0a026408 [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rt...

...ived on SIP/1000-0a292360 [Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read: DTMF begin passthrough '#' on SIP/1000-0a292360 [Oct 2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF begin on channel (SIP/1000-0a292360) [Oct 2 11:09:20] DEBUG[29533]: channel.c:5150 ast_channel_bridge: Bridge stops bridging channels SIP/1000-0a292360 and SIP/1001-0a026408 [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rt...

...ived on SIP/1000-0a292360 [Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read: DTMF begin passthrough '#' on SIP/1000-0a292360 [Oct 2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF begin on channel (SIP/1000-0a292360) [Oct 2 11:09:20] DEBUG[29533]: channel.c:5150 ast_channel_bridge: Bridge stops bridging channels SIP/1000-0a292360 and SIP/1001-0a026408 [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rt...

...c:5179 socket_read: Ooh, voice format changed to 2 Sep 23 11:49:33 DEBUG[1112759216]: rtp.c:1166 ast_rtp_write: Ooh, format changed from UNKN to GSM Sep 23 11:50:01 DEBUG[1090837424]: chan_iax2.c:5208 socket_read: Ooh, video format changed to 262144 Sep 23 11:50:04 DEBUG[1112759216]: channel.c:2655 ast_channel_bridge: Didn't get a frame from channel: SIP/1102-62b6 Sep 23 11:50:04 DEBUG[1112759216]: channel.c:2725 ast_channel_bridge: Bridge stops bridging channels SIP/1102-62b6 and IAX2/192.168.0.7:4569/2 Sep 23 11:50:04 DEBUG[1112759216]: chan_iax2.c:2337 iax2_hangup: We're hanging up IAX2/192.168.0.7:...

...alow-64 both. The call is terminated with this debug output - Didn't get a frame from channel: MGCP/aaln/1@dlink-1 and is Hanged up. Jan 18 20:32:55 NOTICE[114695]: chan_mgcp.c:2293 handle_response: Terminating on result 406 from aaln/1@dlink--1 Jan 18 20:32:55 DEBUG[294930]: channel.c:2655 ast_channel_bridge: Didn't get a frame from channel: MGCP/aaln/1@dlink-1 Jan 18 20:32:55 DEBUG[294930]: channel.c:2725 ast_channel_bridge: Bridge stops bridging channels MGCP/aaln/1@dlink-1 and H323/192.168.1.36 -- No command found on [dlink] for transaction 19. Ignoring... Jan 18 20:32:55 DEBUG[294930]: app...

...retransmission on '3c26700f6ddd-eeq2u57j5kuf@snom320-00041324ABE7' of Request 112: Match Found P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE] Apr 23 12:42:43 DEBUG[7625]: channel.c:3275 ast_generic_bridge: Didn't get a frame from channel: mISDN/1-1 Apr 23 12:42:43 DEBUG[7625]: channel.c:3550 ast_channel_bridge: Bridge stops bridging channels mISDN/1-1 and SIP/4-5079 Apr 23 12:42:43 DEBUG[7625]: chan_sip.c:2426 sip_hangup: update_call_counter(4) - decrement call limit counter Apr 23 12:42:43 DEBUG[7625]: app_dial.c:1619 dial_exec_full: Exiting with DIALSTATUS=ANSWER. -- AGI Script exten2.py complet...

...er/PLCI/NCCI = 0x301 InfoNumber = 0x18 InfoElement = <80> Oct 22 09:22:40 ERROR[49156]: File chan_capi.c, Line 1111 (find_pipe): unable to find a pipe for PLCI = 0x301 MN = 0x509a Oct 22 09:22:58 DEBUG[901149]: File channel.c, Line 2245 (ast_channel_bridge): Didn't get a frame from channel: H323:6083 Oct 22 09:22:58 DEBUG[901149]: File channel.c, Line 2313 (ast_channel_bridge): Bridge stops bridging channels CAPI[contr1/5427980]/9 and H323:6083 -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # F...