[Asterisk-Users] cisco dial-peer voip

[Brian Wilkins]

Why not just have the Asterisk server act as a SIP/H323 gateway instead of the 
cisco router? You can then send incoming calls to registered Asterisk users 
via the cisco router and outgoing calls from Asterisk users to the PSTN via 
the cisco router. You can still use your same config below, but send the VoIP 
sessions through Asterisk and let it parse out where the calls need to go and 
send it to the cisco if you want to terminate traffic.

On Tuesday 30 November 2004 01:35 pm, Jan Baggen wrote:
> I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over
> pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y
> So far so good.
>
> But I want to setup VOIP sessions with local carrier. I added dial-peer
> 40 for this. Session target x.x.x.x But calls will always get routed to
> the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried.
>
> My situation:
> PSTN -> CISCO -> ASTERISK  OK
> ASTERISK -> CISCO -> PSTN  OK
> ASTERISK -> CISCO -> VOIP  NOT OK (only needs outbound calls)
>
>
> SIP01#sh dial-peer voice summary
> dial-peer hunt 0
> TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET
> STAT PORT
> 10     pots  up   up                                0 down 1/0/0
> 20     pots  up   up                                0 down 1/0/1
> 30     voip  up   up             2012345..          0  syst
> ipv4:y.y.y.y:5060
> 40     voip  up   up             .+                 0  syst
> ipv4:x.x.x.x:5060
> 50     pots  up   up             .+                 5 up   1/0/0
> 60     pots  up   up             .+                 5 up   1/0/1
>
>
>
> dial-peer voice 10 pots
>  description INBOUND CALLS PSTN BRI0
>  incoming called-number 2012345..
>  no digit-strip
>  direct-inward-dial
>  port 1/0/0
> !
> dial-peer voice 20 pots
>  description INBOUND CALLS PSTN BRI1
>  incoming called-number 2012345..
>  no digit-strip
>  direct-inward-dial
>  port 1/0/1
> !
> dial-peer voice 30 voip
>  description INBOUND CALLS VOIP ASTERISK
>  destination-pattern 2051860..
>  session protocol sipv2
>  session target ipv4:y.y.y.y:5060
>  session transport udp
>  dtmf-relay sip-notify
>  codec g711alaw
>  no vad
> !
> dial-peer voice 40 voip
>  description OUTBOUND CALLS VOIP CARRIER
>  destination-pattern .+
>  session protocol sipv2
>  session target ipv4:x.x.x.x:5060
>  session transport tcp
>  dtmf-relay sip-notify
>  codec g711alaw
>  no vad
> !
> dial-peer voice 50 pots
>  tone ringback alert-no-PI
>  description OUTBOUND CALLS PSTN BRI0
>  preference 5
>  destination-pattern .+
>  no digit-strip
>  port 1/0/0
> !
> dial-peer voice 60 pots
>  tone ringback alert-no-PI
>  description OUTBOUND CALLS PSTN BRI1
>  preference 5
>  destination-pattern .+
>  no digit-strip
>  port 1/0/1
>
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-- 
Brian Wilkins
Software Engineer
brian at hcc.net

Heritage Communications Corporation
  Melbourne, FL     USA     32935
321.308.4000 x33
http://www.hcc.net