[cisco-voip] Cisco dial-peer question
OConnell, Ryan
Ryan.OConnell at nexinnovations.com
Thu May 18 16:44:39 EDT 2006
You should try to be more exact on your dial-patterns instead of using
the . wildcard. I'm really not sure what you're trying to accomplish
from your explanation so perhaps you can clarify or take a peek at this
doc it will clarify how peers get matched.
http://www.cisco.com/en/US/partner/products/sw/iosswrel/ps1835/products_
configuration_guide_chapter09186a0080080aec.html
Ryan O'Connell
Senior Consultant CCIE Voice (13382)
NexInnovations Inc.
3636 - 23rd Street N.E.
Calgary, AB. T2E 8Z5
Desk (403) 291-7293
Cell: (403) 990-1275
Fax: (403) 291-7281
Ryan.OConnell at nexinnovations.com
http://www.nexinnovations.com
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jarrod Baumann
Sent: May 18, 2006 12:52 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cisco dial-peer question
I am currently attempting to use a Cisco router with PRI cards and a
Voice
IOS image. The problem I am having is with the dial-peer configuration
and
accomplishing the inbound/outbound call flow where I match all calls
coming
from all PRIs and forwarding to my softswitch, with some specific DIDs
being
forwarded to other SIP servers, and a match-all for all voip calls from
the
softswitch being routed through a specific LD PRI, with some specific
matching for sending local calls out a different PRI. Should I be able
to
accomplish this?
Here is a crude example from memory:
dial-peer voice 1 pots
incoming-number .
direct-inward-dial
dial-peer voice 2 pots
destination-pattern 713.......
no digit-strip
port 1/0:23
dial-peer voice 3 pots
destination-pattern .
no digit-strip
port 1/1:23
dial-peer voice 4 voip
destination-pattern .
session protocol sipv2
session target ipv4:10.10.10.2
codec g711ulaw
dial-peer voice 5 voip
destination-pattern 7135551212
session protocol sipv2
session target ipv4:10.10.10.3
codec g711ulaw
When I have this the calls from voip to pstn seem to work correctly, but
the
calls coming from pstn which are supposed to be forward to the
softswitch
seem to be matching other pots dial-peers.
Any suggestions are much appreciated.
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