[asterisk-bugs] [Asterisk 0015863]: BroadVoice With Asterisk

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=15863 
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Reported By:                vijay85_ace
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15863
Category:                   Applications/NewFeature
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.26.1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 no change required
Fixed in Version:           
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Date Submitted:             2009-09-09 08:23 CDT
Last Modified:              2009-09-09 12:02 CDT
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Summary:                    BroadVoice With Asterisk
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 at sip.broadvoice.com:XXXXXXXXXX:3017039676 at 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


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---------------------------------------------------------------------- 
 (0110432) lmadsen (administrator) - 2009-09-09 12:02
 https://issues.asterisk.org/view.php?id=15863#c110432 
---------------------------------------------------------------------- 
This is a support issue. You should use the asterisk-users mailing lists,
or the #asterisk IRC channel on irc.freenode.net. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-09-09 12:02 lmadsen        Note Added: 0110432                          
2009-09-09 12:02 lmadsen        Status                   new => closed       
2009-09-09 12:02 lmadsen        Resolution               open => no change
required
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