RE: Dial-peer question

Robert,

If you're still having problems, I would suggest a couple more debugs:
'debug ccsip messages'
'debug voice translation'

I get the following with my tests:

Oct 28 08:58:26.930: //-1/E3B05A7584F4/RXRULE/sed_subst: Successful
substitution; pattern=4949 matchPattern=.... replacePattern=6001 replaced
pattern=6001

Sent:
INVITE sip:6001_at_192.168.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.x.x:5060;branch=z9hG4bK227222A6
Remote-Party-ID:
<sip:703XXXXXXX_at_192.168.x.x>;party=calling;screen=yes;privacy=off
From: <sip:703XXXXXXX_at_192.168.x.x>;tag=4F21E1D8-D3A
To: sip:6001_at_192.168.x.x

Then typical 100, 180, 200 messages follow.

-ryan

From: Robert Kimble [mailto:hoodooman21_at_gmail.com]
Sent: Wednesday, October 27, 2010 12:50 PM
To: Ryan West
Cc: Cisco certification
Subject: Re: Dial-peer question

Ryan,

Here is the output from sh run | s voice service voip:

rtr218-CP#sh run | s voice service voip
voice service voip
 sip
  bind control source-interface GigabitEthernet0/0.4
  bind media source-interface GigabitEthernet0/0.4
  rel1xx disable

g0/0.4 is the default gateway for the voice vlan.

As far as dtmf, I didn't have anything configured, but I just ran through all
four options and they all rang busy.

If I dial and existing ephone-dn the call goes right through to voice mail and
if I set the E&M circuit to plar with the AA number all calls make it to the
AA. I think I need to step away from this one for an hour or so. Probably been
looking at it too long.

Any ideas are very welcome.

-Bobby

On Wed, Oct 27, 2010 at 11:29 AM, Ryan West
<rwest_at_zyedge.com<mailto:rwest_at_zyedge.com>> wrote:
Bobby,

Have you verified the AA functionality? Your config is on the money, I tested
with a lab system and can dial in through a PRI link just fine. I tried most
explicit and then worked my way out to .... on both the dial-peer and
translation-rule's and each time it was fine.

I tested your translation-rule's as well, no issue there either. What are you
doing for DMTF relay (not that I think it's related to this issue). Also, if
you're able to dial into the CUE/AA from an IP phone, can you post the output
from 'show run | s voice service voip'

-ryan

From: Robert Kimble
[mailto:hoodooman21_at_gmail.com<mailto:hoodooman21_at_gmail.com>]
Sent: Wednesday, October 27, 2010 12:03 PM
To: Ryan West
Cc: Cisco certification
Subject: Re: Dial-peer question

Thanks Ryan,

I'm not sure where I'm going wrong though....

Here's the info:

Pilot number: 2101
non-existing extension: 2004
CUE address: 10.218.4.100
Circuit: E&M wink with 4 digits being passed off

Configs:

voice translation-rule 1
 rule 1 /..../ /2101/
!
!
voice translation-profile to_AA
 translate called 1

dial-peer voice 2 voip
 translation-profile outgoing to_AA
 destination-pattern ....
 session target ipv4:10.218.4.100
 codec g711ulaw
 no vad

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Received on Thu Oct 28 2010 - 13:00:58 ART