search for: sip_hangup

....1-rc30) on our pbx but since no voice was passing I decided to go back to old version (0.3.1-rc23). Last friday everything seemed to work fine but now every incoming call drops after 3-4 seconds while Asterisk console is showing these messages: Apr 23 12:42:39 DEBUG[7625]: chan_sip.c:2426 sip_hangup: update_call_counter(3) - decrement call limit counter Apr 23 12:42:39 DEBUG[7625]: chan_sip.c:1379 __sip_ack: Acked pending invite 102 Apr 23 12:42:39 DEBUG[7625]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '1fd7824840123666030e29a70d1d7739@192.168.1.200' of Request 102: Mat...

...grab lock, trying again... 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:7417 sipsock_read: Failed to grab lock, trying again... 2004-08-09 17:36:29 DEBUG[245775]: channel.c:733 ast_hangup: Hanging up channel 'SIP/65.67.76.30-0814e4f0' 2004-08-09 17:36:29 DEBUG[245775]: chan_sip.c:1717 sip_hangup: sip_hangup(SIP/65.67.76.30-0814e4f0) 2004-08-09 17:36:29 DEBUG[245775]: chan_sip.c:1732 sip_hangup: update_user_counter() - decrement inUse counter 2004-08-09 17:36:29 DEBUG[245775]: chan_sip.c:1605 update_user_counter: is not a local user 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:817...

Hi I have noticed that asterisk returns 'SIP 603' when the called party does not answer. My test setup is simple: two SIP phones (extensions: 100 and 111) registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to 111 (expected) and a '603 Decline' response to 100 (unexpected &

...26 15:54:17 DEBUG[-1260983376]: pbx.c:1827 ast_pbx_run: Spawn extension (from-sip,0085221120000,3) exitednon-zero on 'SIP/2000-e12c' Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:733 ast_hangup: Hanging up channel 'SIP/2000-e12c' Aug 26 15:54:17 DEBUG[-1260983376]: chan_sip.c:1717 sip_hangup: sip_hangup (SIP/2000-e12c) Aug 26 15:54:17 DEBUG[-1260983376]: chan_sip.c:1732 sip_hangup: update_user_counter(2000) - decrement inUse counter Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:817 __sip_ack: Stopping retransmission on '95AB5805-C94F-4C15-AC5A-6DFE5F58D644@192.168.1.101' of...

...annel.c, Line 947 (ast_settimeout): Scheduling timer at 0 sample intervals WARNING[294927]: File app_voicemail2.c, Line 2567 (vm_execmain): Couldn't read u sername == Spawn extension (extensions, 1001, 1) exited non-zero on 'SIP/xirak-259d' DEBUG[294927]: File chan_sip.c, Line 980 (sip_hangup): find_user(xirak)

...17 19:20:02 DEBUG[1605650]: File chan_zap.c, Line 1960 (zt_setoption): Set option TDD MODE, value: OFF(0) on Zap/1-1 Oct 17 19:20:02 DEBUG[1605650]: File chan_zap.c, Line 992 (update_conf): Updated conferencing on 1, with 0 conference users Oct 17 19:20:02 DEBUG[1605650]: File chan_sip.c, Line 985 (sip_hangup): find_user(atasuporte) Oct 17 19:20:04 DEBUG[147466]: File chan_zap.c, Line 1033 (zt_disable_ec): disabled echo cancellation on channel 1 ============================================ debug when I hangup the ATA186 =========================== Oct 17 19:40:25 DEBUG[278546]: File dsp.c, Line 1212 (a...

...0 sample intervals Sep 13 22:18:12 WARNING[13167]: app_voicemail.c:4890 vm_authenticate: Couldn't r ead username Sep 13 22:18:12 DEBUG[13167]: pbx.c:2262 __ast_pbx_run: Extension 8500, priority 1 returned normally even though call was hung up Sep 13 22:18:12 DEBUG[13167]: chan_sip.c:2315 sip_hangup: update_user_counter(Ph one3) - decrement inUse counter Sep 13 22:18:16 DEBUG[13167]: chan_sip.c:6350 check_user_full: Setting NAT on RT P to 0 Sep 13 22:18:16 DEBUG[13167]: chan_sip.c:9413 handle_request_invite: Checking SI P call limits for device Phone3 Sep 13 22:18:16 DEBUG[13167]: chan_sip....

...ng up IAX[12.37.165.130:5036]/7 now... -- Hungup 'IAX[12.37.165.130:5036]/7' == No one is available to answer at this time WARNING[1209269552]: File pbx.c, Line 1810 (ast_pbx_run): Timeout, but no rule 't' in context 'sip' DEBUG[1209269552]: File chan_sip.c, Line 1025 (sip_hangup): find_user(phone1) - decrement inUse counter DEBUG[1133735216]: File chan_sip.c, Line 548 (__sip_ack): Stopping retransmission on '3c29efbbc5b1-diw483wrl88j@10-1-2-24' of Response 1: Found On FWD I get the following DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT...

...: Got it eventually... chan_sip.c line 1453 (sip_alloc): Got it eventually... ..... chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 4980 (do_monitor): Got it eventually... chan_sip.c line 948 (sip_hangup): Error obtaining mutex: Device or resource busy channel.c line 370 (ast_queue_frame): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resou...

...Jan 7 15:37:01 DEBUG[1234379840]: File cdr_addon_mysql.c, Line 123 (mysql_log): Jan 7 15:37:01 asterisk_pbx[5038]: Jan 7 15:37:01 DEBUG[1234379840]: File cdr_addon_mysql.c, Line 130 (mysql_log): Jan 7 15:37:01 asterisk_pbx[5038]: Jan 7 15:37:01 DEBUG[1234379840]: File chan_sip.c, Line 1081 (sip_hangup): Festival's info is very minimal, but seems to indicate success: # cat festival_server.log "Load server start ./festival_server.scm" festival port=1314 wrapper Wed Jan 7 15:36:40 EST 2004 : USING DEFAULT CONFIGURATION wrapper Wed Jan 7 15:36:41 EST 2004 : waiting server Wed Jan...

...munity and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Allow users to specify a port for DUNDi peers. (Closes issue #17056. Reported, patched by klaus3000) * Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. (Closes issue #16815. Reported, patched by rain) * First caller into a dynamic conference new enters the pin once. (Closes issue #15878. Reported, patched by pabelanger) * Send AgentComplete manager events in the event of blind and attended...

...munity and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Allow users to specify a port for DUNDI peers. (Closes issue #17056. Reported, patched by klaus3000) * Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. (Closes issue #16815. Reported, patched by rain) * If there is realtime configuration, it does not get re-read on reload unless the config file also changes. (Closes issue #16982. Reported, patched by dmitri) * Send AgentComplete manager...

...ot RTCP report of 84 bytes May 15 22:31:17 DEBUG[4792]: Didn't get a frame from channel: SIP/2463-2f7a May 15 22:31:17 DEBUG[4792]: Bridge stops bridging channels SIP/2433-9716 and SIP/2463-2f7a May 15 22:31:17 DEBUG[4792]: Hanging up channel 'SIP/2463-2f7a' May 15 22:31:17 DEBUG[4792]: sip_hangup(SIP/2463-2f7a) May 15 22:31:17 DEBUG[4792]: update_user_counter(2463) - decrement outUse counter May 15 22:31:17 DEBUG[4792]: Exiting with DIALSTATUS=ANSWER. May 15 22:31:17 DEBUG[4792]: Exiting with ANSWERTIME=7. May 15 22:31:17 DEBUG[4792]: Spawn extension (macro-stdexten,s,4) exited non-zero on...

...xecuting Hangup("SIP/nyct-901-539f", "") in new stack Apr 21 12:35:18 WARNING[16815]: pbx.c:5548 pbx_builtin_hangup: chan->hangupcause=(null) == Spawn extension (nyct, 9218, 2) exited non-zero on 'SIP/nyct-901-539f' Apr 21 12:35:18 WARNING[16815]: chan_sip.c:2471 sip_hangup: ast->hangupcause=16 res=(null) This is all on Asterisk 1.2.7.1. Your line numbers may vary since there were some ast_log lines added. Hopefully this makes some sense to someone. Thanks for any help or input. -- New York Connect Technical Support Staff Eri...

...munity and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Allow users to specify a port for DUNDi peers. (Closes issue #17056. Reported, patched by klaus3000) * Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. (Closes issue #16815. Reported, patched by rain) * First caller into a dynamic conference new enters the pin once. (Closes issue #15878. Reported, patched by pabelanger) * Send AgentComplete manager events in the event of blind and attended...

...munity and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Allow users to specify a port for DUNDI peers. (Closes issue #17056. Reported, patched by klaus3000) * Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. (Closes issue #16815. Reported, patched by rain) * If there is realtime configuration, it does not get re-read on reload unless the config file also changes. (Closes issue #16982. Reported, patched by dmitri) * Send AgentComplete manager...

...e: Bridge stops because we're zombie or need a soft hangup: c0=OH323/R27469, c1=SIP/xlite1-89a7, flags: No,Yes,No,No 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]: chan_sip.c:1729 sip_hangup: update_user_counter(xlite1) - decrement outUse counter Aug 13 10:19:05 DEBUG[524304]: app_dial.c:965 dial_exec: Exiting with DIALSTATUS=ANSWER. == Spawn extension (default, 224, 1) exited non-zero on 'OH323/R27469' -- H.323 call 'ip$IPofNetmeeting:1082/27469' cleared, reason 1 (...

...zap.c:3080 zt_handle_event: Got event Ring/Answered(2) on channel 1 (index 0) May 27 18:08:12 WARNING[1224]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 6400b01c3dca805e76adf78264e9f598@10.203.10.101 for seqno 102 (Critical Request) May 27 18:08:12 DEBUG[1224]: chan_sip.c:1716 sip_hangup: update_user_counter(7011) - decrement outUse counter == No one is available to answer at this time May 27 18:08:12 DEBUG[1224]: app_dial.c:1037 dial_exec: Exiting with DIALSTATUS=NOANSWER. May 27 18:08:12 DEBUG[1224]: chan_zap.c:3768 __zt_exception: Exception on 15, channel 1 May 27 18:08:1...

...cancellation on channel 1 -- Hungup 'Zap/1-1' Oct 4 00:54:04 DEBUG[1146877376]: app_dial.c:975 dial_exec: Exiting with DIALSTATUS=ANSWER. == Spawn extension (6568543197, 6591596323, 2) exited non-zero on 'SIP/6568543197-86c2' Oct 4 00:54:04 DEBUG[1146877376]: chan_sip.c:1749 sip_hangup: update_user_counter(6568543197) - decrement inUse counter

...e: Contact hop: <sip:1747xxxxxxx@192.168.0.100:51332> Nov 6 01:50:14 WARNING[4101]: File chan_sip.c, Line 456 (retrans_pkt): Maximum retries exceeded on call 93161d54-1d74-7515-1fba-b6c8db18e466@192.168.0.100 for seqno 32120 (Response) Nov 6 01:50:14 DEBUG[9226]: File chan_sip.c, Line 1062 (sip_hangup): find_user(xxxxx) - decrement inUse counter Nov 6 01:50:14 DEBUG[4101]: File chan_sip.c, Line 559 (__sip_ack): Stopping retransmission on '93161d54-1d74-7515-1fba-b6c8db18e466@192.168.0.100' of Request 102: Found Nov 6 01:50:14 DEBUG[4101]: File chan_sip.c, Line 886 (__sip_destroy): Dest...