[Asterisk-Users] Cisco PRI Gateway Problems
Peder Angvall
peder at angvall.com
Fri Oct 29 13:16:43 MST 2004
That was it. I knew it wasn't in there, but I was just trying to call
into the PRI to * and not from * out, so I didn't think it would matter.
Another goofy Cisco trick I guess.
Bruce Komito wrote:
> I think you are missing a dial-peer voice xxx pots entry. E.g.:
>
> dial-peer voice 200 pots
> description Match all inbound POTS calls
> incoming called-number T
> direct-inward-dial
>
> I don't think the PRI will pick up the call unless the called number
> matches a number in one of the pots dial-peers.
>
> Bruce Komito
> High Sierra Networks, Inc.
> www.servers-r-us.com
> (775) 236-5815
>
>
> On Fri, 29 Oct 2004, Peder Angvall wrote:
>
>
>>I am trying to get a Cisco PRI gateway to send calls to * and it doesn't
>>appear to be working. It is a 2610 running 12.3 IP+. I've got the
>>config in there, I can see calls come into the Cisco using debugs, but I
>>never see it try to connect to *. When I do debugs, I see the called #
>>as the 10 digit # and I see the calling # as my #, but I never see
>>anything on *. Both devices can ping each other and neither is behind a
>>firewall. If I do a "sip show registry" on the * box, the router is NOT
>>registered, but I never see any error messages either, so it looks like
>>it isn't even trying to register with *. Anybody have any ideas?
>>
>>Here is the relevant config from the 2610. We are being passed a 10
>>digit # (I replaced the real #'s with 123456 below).
>>
>>voice service voip
>> signaling forward unconditional
>> sip
>>
>>controller T1 1/0
>> framing esf
>> linecode b8zs
>> pri-group timeslots 1-24
>>
>>interface Serial1/0:23
>> no ip address
>> isdn switch-type primary-ni
>> isdn incoming-voice voice
>> no cdp enable
>>
>>voice-port 1/0:23
>>!
>>dial-peer voice 1 voip
>> destination-pattern 123456....
>> session protocol sipv2
>> session target ipv4:192.168.1.2:5060
>> session transport udp
>> dtmf-relay rtp-nte
>> codec g711ulaw
>> no vad
>>!
>>sip-ua
>> retry invite 3
>> retry response 3
>> retry bye 3
>> retry cancel 3
>> timers trying 1000
>> sip-server ipv4:192.168.1.2
>>
>>Here is my sip.conf:
>>
>>[general]
>>port=5060
>>bindaddr=192.168.1.2
>>disallow=all
>>allow=ulaw
>>
>>[192.168.1.1]
>>context=pstn-incoming
>>type=friend
>>host=192.168.1.1
>>dtmfmode=rfc2833
>>disallow=all
>>allow=ulaw
>>
>>[3200]
>>context=local-phones
>>type=friend
>>username=3200
>>secret=3200
>>host=dynamic
>>mailbox=3200
>>
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>This message has been categorized as "Legitimate" by Bayesian Analyzer.
>>If you do not agree, please click on the link below to train the Analyzer.
>>http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-10-29%5C5bc66d662f1440aba60e35c11252071d&C=2
>>
>>--
>>-----------------------------------------------------------------------
>>This message has been inspected by DynaComm i:mail
>>-----------------------------------------------------------------------
>>
>>
>
>
>
>
More information about the asterisk-users
mailing list