micke@party.pp.se
2004-Jul-14 04:27 UTC
[Asterisk-Users] Questing regardning dialplans on a Cisco 5350
Hi. If I use a Cisco as a PSTN termination GW and need to route all incoming isdn calls to my asterisk and all outgoing calls from asterisk via the cisco out to pstn, how do I do that ? in the cisco I have this: dial-peer voice 1 pots destination-pattern [0-9]T no digit-strip direct-inward-dial port 3/0:D ! dial-peer voice 50 voip destination-pattern [0-9] voice-class codec 1 session protocol sipv2 session target sip-server no vad dtmf-relay rtp-nte ! ------- But theese to dialplans seem to interrupt each other. When an incoming call from PSTN goes through this the pattern can be matched by the first, and then be routed ot on the PSTN again, creating a loop. How do I do this in the smartest and easiest way ? /Mike
Glen Hinkle
2004-Jul-14 07:58 UTC
[Asterisk-Users] Questing regardning dialplans on a Cisco 5350
The call is inbound on the pots dial-peer, so you should use incoming called-number, as opposed to destination-pattern. dial-peer voice 1 pots incoming-called number [0-9]T no digit-strip direct-inward-dial port 3/0:D I'm not familiar with the [0-9] syntax, but if it works, ok. I usually use "." Also, you can specify the sip destination directly in the dial-peer, which makes using sip with the cisco's more flexible unless you're using a separate sip proxy. session protocol sipv2 session target ipv4:5.5.5.5 -g On Wed, 2004-07-14 at 07:27, micke@party.pp.se wrote:> > Hi. > > > If I use a Cisco as a PSTN termination GW and need to route all incoming > isdn calls to my asterisk and all outgoing calls from asterisk via the > cisco out to pstn, how do I do that ? > > > in the cisco I have this: > > dial-peer voice 1 pots > destination-pattern [0-9]T > no digit-strip > direct-inward-dial > port 3/0:D > ! > dial-peer voice 50 voip > destination-pattern [0-9] > voice-class codec 1 > session protocol sipv2 > session target sip-server > no vad > dtmf-relay rtp-nte > ! > > > ------- > > But theese to dialplans seem to interrupt each other. > > When an incoming call from PSTN goes through this the pattern can be > matched by the first, and then be routed ot on the PSTN again, creating > a loop. > > How do I do this in the smartest and easiest way ? > > /Mike > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@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