vijay.goyal at alliance-infotech.com
2009-Sep-10 05:49 UTC
[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
=>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
Thanks & Regards
Vijay Goyal
Software Engineer - VOIP
Alliance Infotech Private Limited
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
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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-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
vijay.goyal at alliance-infotech.com
Sent: Thursday, September 10, 2009 12:50 AM
To: asterisk-users at 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
=>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
<mailto:1%7d at 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
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Maybe try removing the /301 from your register line and either use nothing or try /sip.broadvoice.com vijay.goyal at alliance-infotech.com wrote:> 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 > =>3017039676 at sip.broadvoice.com > <mailto:3017039676 at sip.broadvoice.com>:XXXXXXXXXX:3017039676 at sip.broadvoice.com > <mailto: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 > <mailto:1%7D at 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-- MARK. Hulber Technologies Read my blog : http://mark.hulber.com Follow @hulber on Twitter: http://twitter.com/hulber