Hi asterisk users, I have a inssue with incoming calls with wcfxo card, while receiving a call, I?ve configured my dialplan to forward the call to all mi home voip extensions and that works just fine, but while in the call, after a few seconds, the pbx starts the simple switch once more and keeps ringing the voip extensions log as follows: ---------------------------------------------------------------------------- ---------------------------------------------------------------------------- - -- Starting simple switch on 'Zap/1-1' Feb 3 12:11:17 NOTICE[6424]: chan_zap.c:5363 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial("Zap/1-1", "SIP/2001&SIP/2002&IAX2/2003&IAX2/2101&SIP/2102&SIP/2103|10") in new stack Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'IAX2' -- Called 2101 Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' -- Call accepted by 192.168.200.100 (format gsm) -- Format for call is gsm -- IAX2/2101/1 is ringing -- IAX2/2101/1 answered Zap/1-1 Feb 3 12:11:31 NOTICE[2025]: chan_iax2.c:1371 iax2_destroy: Avoiding IAX destroy deadlock -- Hungup 'IAX2/2101/1' == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Feb 3 12:11:54 WARNING[6428]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Dial("Zap/1-1", "SIP/2001&SIP/2002&IAX2/2003&IAX2/2101&SIP/2102&SIP/2103|10") in new stack Feb 3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create channel of type 'IAX2' -- Call accepted by 192.168.200.100 (format gsm) -- Format for call is gsm -- Called 2101 Feb 3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' -- IAX2/2101/3 is ringing -- Hungup 'IAX2/2101/3' == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' Hungup 'Zap/1-1' ---------------------------------------------------------------------------- ---------------------------------------------------------------------------- - Also if someone tooks the physical PSTN phone line and make an outgoing call I receive exactly the same behavior Here?s what I have in my Zapata.conf and extensions.conf, please any advice to correct/handle this problem will be awesome! Thanks in advance! ;-------------------- zapata.conf ---------------------- [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 rxgain=10.5 txgain=-1.0 group=1 callgroup=1 pickupgroup=1 immediate=no signalling=fxs_ks context=default ;-------------------- extensions.conf ---------------------- [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp TRUNK=Zap/1 PACO=IAX2/2101&SIP/2102&SIP/2103 CASA=SIP/2001&SIP/2002&IAX2/2003&${PACO} ;----------------------- COMMON SERVICES ------------------------- [default] ; context used for general services exten => s,1,Dial(${CASA},10) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050203/871c6d5d/attachment.htm
Hi asterisk users, I have a inssue with incoming calls with wcfxo card, while receiving a call, I?ve configured my dialplan to forward the call to all mi home voip extensions and that works just fine, but while in the call, after a few seconds, the pbx starts the simple switch once more and keeps ringing the voip extensions log as follows: ---------------------------------------------------------------------------- ---------------------------------------------------------------------------- - -- Starting simple switch on 'Zap/1-1' Feb 3 12:11:17 NOTICE[6424]: chan_zap.c:5363 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial("Zap/1-1", "SIP/2001&SIP/2002&IAX2/2003&IAX2/2101&SIP/2102&SIP/2103|10") in new stack Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'IAX2' -- Called 2101 Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' -- Call accepted by 192.168.200.100 (format gsm) -- Format for call is gsm -- IAX2/2101/1 is ringing -- IAX2/2101/1 answered Zap/1-1 Feb 3 12:11:31 NOTICE[2025]: chan_iax2.c:1371 iax2_destroy: Avoiding IAX destroy deadlock -- Hungup 'IAX2/2101/1' == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Feb 3 12:11:54 WARNING[6428]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Dial("Zap/1-1", "SIP/2001&SIP/2002&IAX2/2003&IAX2/2101&SIP/2102&SIP/2103|10") in new stack Feb 3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create channel of type 'IAX2' -- Call accepted by 192.168.200.100 (format gsm) -- Format for call is gsm -- Called 2101 Feb 3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:54 NOTICE[6428]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' -- IAX2/2101/3 is ringing -- Hungup 'IAX2/2101/3' == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' Hungup 'Zap/1-1' ---------------------------------------------------------------------------- ---------------------------------------------------------------------------- - Also if someone tooks the physical PSTN phone line and make an outgoing call I receive exactly the same behavior Here?s what I have in my Zapata.conf and extensions.conf, please any advice to correct/handle this problem will be awesome! Thanks in advance! ;-------------------- zapata.conf ---------------------- [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 rxgain=10.5 txgain=-1.0 group=1 callgroup=1 pickupgroup=1 immediate=no signalling=fxs_ks context=default ;-------------------- extensions.conf ---------------------- [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp TRUNK=Zap/1 PACO=IAX2/2101&SIP/2102&SIP/2103 CASA=SIP/2001&SIP/2002&IAX2/2003&${PACO} ;----------------------- COMMON SERVICES ------------------------- [default] ; context used for general services exten => s,1,Dial(${CASA},10) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050203/3cecf386/attachment.htm