[cisco-voip] FW: Cisco CUBE Sip to Sip Question

[Granger, Simon]

Thank you for your reply Paul

Good spot it was a UK number (we are only testing it in the UK before we install it in Brussels.) I thought the @ was for Call Manager, i think the time it is waiting is on the 9T on the router itself, but i will now need to go and verify this.

The reason we were wanting to keep the 9 was that most of our offices use H323 gateways, and when dialling the 9 it does not strip them when going out the ISDN lines, so we were trying to mimic this for the sip setup.

Thanks

________________________________________
From: Paul [asobihoudai at yahoo.com]
Sent: 17 June 2011 16:22
To: Granger, Simon; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] FW: Cisco CUBE Sip to Sip Question

You can also install a UK dial plan and use the @ macro to bypass the T302 (interdigit timeout) timer. I'm not sure why you want the 9 on the phone to not drop but from what I've seen this is default behavior. Perhaps if you altered to Called Party Transform Mask you can make the number appear as if it isn't dropped? I haven't tested this out.

If you install the UK dialplan and use the @ macro with route filters, this does require a reboot of your servers but I found the macro to be indispensable when deploying phones to an office in India.


________________________________
From: "Granger, Simon" <simon.granger at fmglobal.com>
To: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Sent: Friday, June 17, 2011 7:43 AM
Subject: [cisco-voip] FW: Cisco CUBE Sip to Sip Question

Hi All

Hopefully an easy couple of question,

In Communications Manager we have created a SIP trunk to our CUBE router. This SIP trunk is part in a route list for route pattern 9.01753123123

On the CUBE Router we have the following Dial Peer and respective voice translation profiles.

voice translation-rule 4
rule 1 /^9\(.*\)/ /\1/

voice translation-profile SIP_OUTGOING
translate calling 3
translate called 4


dial-peer voice 9 voip
translation-profile outgoing SIP_OUTGOING
destination-pattern 9T
session protocol sipv2
session target ipv4:82.12.241.168
dtmf-relay rtp-nte
codec g711ulaw
!

The First question, is that as we are using the Route Pattern 9.T we are having to wait for the interdigit timeout to complete before it sends the call out the SIP trunk, As there is no voice port associated with this dial peer, is there a way to reduce the interdigit timeout so it starts the call quicker?

The second question I have is that the translation rule is adjusting the display on the Cisco IP phone, so for example:

The phone displays 901753123123
As soon as it matches the dial peer then phone adjusts to 01752123123 (i.e it drops the 9) Is there a way to stop this?

Thanks

Simon



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