Summary: | ASTERISK-14804: BroadVoice With Asterisk | ||
Reporter: | vijay goyal (vijay85_ace) | Labels: | |
Date Opened: | 2009-09-09 08:23:06 | Date Closed: | 2011-06-07 14:01:02 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Applications/NewFeature |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | Hi All, 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 =>3017039676@sip.broadvoice.com:XXXXXXXXXX:3017039676@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/${EXTEN},50,t) exten=_3XX,n,GotoIF($["${DIALSTATUS}"="BUSY"]?busy:un) exten=_3XX,n(un),VoiceMail(${EXTEN}@default,u) exten=_3XX,n,Hangup() exten=_3XX,n(busy),VoiceMail(${EXTEN}@default,b) exten=_3XX,n,Hangup exten=3017039676,1,dial(SIP/301) exten=_9.,1,dial(SIP/${EXTEN:1}@sip.broadvoice.com,50) exten=_9.,n,Hangup Thanks in advance Vijay Goyal ****** ADDITIONAL INFORMATION ****** hi | ||
Comments: | By: Leif Madsen (lmadsen) 2009-09-09 12:02:41 This is a support issue. You should use the asterisk-users mailing lists, or the #asterisk IRC channel on irc.freenode.net. |