Perhaps you omitted it for space considerations, but it seems that you don't have any [default] call handling. You would definitely need this for attended calls. Assuming I am incorrect, you should post your CLI output from a couple of failed calls.
_____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vijay.go...@alliance-infotech.com Sent: Thursday, September 10, 2009 12:50 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk With Broadvoice Hi, I am using Asterisk 1.4.25. I have one Broadvoice account. I Integrated this broadvoice account with Asterisk Server. I am Able to Make calls but cannot recieve calls. In Incoming calls, call lands to SIP extension, as I attend the call....It gets hungup......... If i dont transfer this call to extension or I play any file then it works OK. But as I transfer it to SIP Extension it get hungs up. Please Help me....it is very urgent. Kindly find my sip.conf and extension.conf sip.conf:- [general] port=5060 bindaddr=192.168.1.170 pedantic=no allow=all NAT=yes language=en relaxdtmf=yes rtptimeout=60 dtmfmode=auto allow=alaw allow=ulaw allow=gsm allow=g723.1 allow=g729 allow=h264 allow=h263 allow=h323 videosupport=yes context=trusted register =>3017039...@sip.broadvoice.com:XXXXXXXXXX:3017039...@sip.broadvoice.com/301 [301] type=friend secret=301 host=dynamic context=trusted [3017039676] type=friend secret=444 host=dynamic context=trusted [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=3017039676 secret=xxxxxxxxx username=3017039676 authname=3017039676 insecure=very context=trusted dtmfmode=inband dtmf=inband Extensions.conf:- [trusted] exten=_3XX,1,dial(SIP/${EXTEN},50,t) exten=_3XX,n,GotoIF($["${DIALSTATUS}"="BUSY"]?busy:un) exten=_3XX,n(un),VoiceMail(${ext...@default,u) exten=_3XX,n,Hangup() exten=_3XX,n(busy),VoiceMail(${ext...@default,b) exten=_3XX,n,Hangup exten=3017039676,1,dial(SIP/301) exten=_9.,1,dial(SIP/${EXTEN:1...@sip.broadvoice.com <mailto:1...@sip.broadvoice.com> ,50) exten=_9.,n,Hangup Thanks in advance Thanks & Regards Vijay Goyal Software Engineer - VOIP Alliance Infotech Private Limited www.alliance-infotech.com <BLOCKED::BLOCKED::BLOCKED::BLOCKED::http://www.alliance-infotech.com/> (An ISO 9001: 2000 certified company) B 254 Okhla Industrial Area-I, New Delhi 110 020 (India) | Tel: +91 11 2637 1851 | Fax: +91 11 2637 1852, 2981 0953 | Mobile: +91 9811974564
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
