I am having a difficult time connecting an Asterisk box to a
Metaswitch. I looked at the page at
http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+to+Metaswitch
but was not able to make much progress. If someone could direct in what
direction to start troubleshooting this problem I would be very
appreciative.
When I try to dial out now. I just get dead air (no ring, no dial tone,
or recording). I can dial between phones that are connected to the
asterisk box with no problem.
Asterisk version 1.2.9.1
In the following files, I changed the phone numbers/extensions and ip
address.
sip.conf
[metaswitch]
type=friend
context=Internal
host=1.2.3.4
fromdomain=1.2.3.4
qualify=900
username=1234567890
canreinvite=no
allow=ulaw
dtmfmode=inband
jb-enable=yes
jb-max-size=300
nat=yes
[1234567890]
type=friend
username=1234567890
secret=1234
context=Internal
callerid="cisco 1" 1234567890
host=dynamic
canreinvite=no
mailbox=3000
extensions.conf
[Internal]
exten => 1234567890,1,Dial(SIP/1234567890,20)
exten => 1234567890,2,Voicemail(u3000)
exten => 1234567890,102,Voicemail(b3000)
exten => 798,1,Set,CALLERID(num)=1234567890
exten => 798,2,Dial(SIP/311@metaswitch|20)
exten => 798,3,Congestion
exten => _XXXXXXX,1,Dial(SIP/${EXTEN}@66.170.55.210);
exten => _XXXXXXX,3,Congestion
exten => _1XXXXXXXXXX,1,Dial(SIP/${EXTEN}@66.170.55.210);
exten => _1XXXXXXXXXX,3,Congestion
exten => _X11,1,Dial(SIP/${EXTEN}@66.170.55.210);
Watkins, Bradley
2006-Jul-24 05:51 UTC
[asterisk-users] Connecting Asterisk to a Metaswitch
I recently got this going, and had a similar experience. In my case,
the solution was to set the RPID to the expected number assigned to the
account. YMMV, but it's worth a try.
Regards,
- Brad
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of kharris
Sent: Monday, July 24, 2006 8:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Connecting Asterisk to a Metaswitch
I am having a difficult time connecting an Asterisk box to a Metaswitch.
I looked at the page at
http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+to+Metaswitch
but was not able to make much progress. If someone could direct in what
direction to start troubleshooting this problem I would be very
appreciative.
When I try to dial out now. I just get dead air (no ring, no dial tone,
or recording). I can dial between phones that are connected to the
asterisk box with no problem.
Asterisk version 1.2.9.1
In the following files, I changed the phone numbers/extensions and ip
address.
sip.conf
[metaswitch]
type=friend
context=Internal
host=1.2.3.4
fromdomain=1.2.3.4
qualify=900
username=1234567890
canreinvite=no
allow=ulaw
dtmfmode=inband
jb-enable=yes
jb-max-size=300
nat=yes
[1234567890]
type=friend
username=1234567890
secret=1234
context=Internal
callerid="cisco 1" 1234567890
host=dynamic
canreinvite=no
mailbox=3000
extensions.conf
[Internal]
exten => 1234567890,1,Dial(SIP/1234567890,20)
exten => 1234567890,2,Voicemail(u3000)
exten => 1234567890,102,Voicemail(b3000)
exten => 798,1,Set,CALLERID(num)=1234567890
exten => 798,2,Dial(SIP/311@metaswitch|20) exten => 798,3,Congestion
exten => _XXXXXXX,1,Dial(SIP/${EXTEN}@66.170.55.210);
exten => _XXXXXXX,3,Congestion
exten => _1XXXXXXXXXX,1,Dial(SIP/${EXTEN}@66.170.55.210);
exten => _1XXXXXXXXXX,3,Congestion
exten => _X11,1,Dial(SIP/${EXTEN}@66.170.55.210);
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