[Asterisk-Users] Questing regardning dialplans on a Cisco 5350

[Glen Hinkle]

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 at 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
> 
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