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Summary:ASTERISK-14804: BroadVoice With Asterisk
Reporter:vijay goyal (vijay85_ace)Labels:
Date Opened:2009-09-09 08:23:06Date Closed:2011-06-07 14:01:02
Priority:MajorRegression?No
Status:Closed/CompleteComponents: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.