SIP trunk Dial-peers incoming and outgoing

[Stephen Keller]

I have always struggled understanding which dial-peers I need when doing VOIP to VOIP dial-peers.

I am pretty sure I need to match an incoming and outgoing dial-peer both on calls coming in from PSTN and calls coming from CUCM.

My users will be using 8 as an outside access code and we will always send 10 digits to SIP provider.

On inbound we will recieve 10 digits and the area code is 517.

below are the dial-peers I plan on using but I sense something is missing.

voice translation-rule 1
rule 1 /8/ //
!
!
voice translation-profile 1
translate called 1
!
dial-peer voice 1 voip
description SIP Trunk to 123.net
translation-profile outgoing 1
destination-pattern 8..........
session protocol sipv2
session target ipv4:216.234.105.74
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 2 voip
description SIP Trunk to CUCM
session protocol sipv2
session target ipv4:172.24.98.35
incoming called-number 517.......
dtmf-relay rtp-nte
codec g711ulaw
no vad
!

will this work for inbound and outbound calls?

call from outside would match 2 on incoming (but what do i need for outgoing leg)

call from inside would match 1 on incoming(b

[nitsinha]

For inbound calls:-

Since are code is 517, it will hit dialpeer 2 as inbound dialpeer. For outboud dialpeer you can still user dialpeer 2, if you add a destination pattern as follows:-

destination-pattern 517.......

For out bound calls:-

Since 8 is an access code, you should add a command called 'forward digits 10' under dialpeer 1. Also include 'incoming called number .' under dialpeer 1 so that the same dialpeer can be used for both inbound and outbound dialpeer match for outbound calls.

Hope that helps

Regards

Nitesh

PS:Please rate helpful posts

[Stephen Keller]

Thank you, much of that sounds like what I was thinking, but was not 100% sure how to implement it.

You cannot use forward-digits command on VOIP dial-peers, that is for POTS only.

however I am using the translation profile to strip off the 8 prior to sending to SIP provider.