CFA - no matching outgoing dial peer

[Neil Sheridan]

Hi,

I've been messing with CFA for a few days now and getting nowhere fast. Speaking with the ITSP I can confirm they do not allow P-Asserted methods but they DO allow Diversion modifications to the SIP header.

They are telling me that the issue is on my end, but won't confirm if the diverted call reaches them. Before I get more involved with them I would like to solve the "No matching outgoing dial peer" error.

I've attached a ccsip messages debug and the dial peers are below. Any assistance would be appreciated.

Call flow:

0862431267 dials the extension 0862702734 (DN is 234) which is diverted to 0450633094

The phone can call this number, and it can also be manually transferred to the number. When dialled with the divert on a busy tone is received. Aside from the DP issue the case for the dropped call is Q.850 cause 1 - which I believe is "unallocated number"??

I've also add a screenshot of the SIP Trunk configuration also - i've tried all permutations without success.

The Call Forward All is set to "Configured CSS" which are set to a CSS the number can be called.

Dial peers are:

dial-peer voice 10 voip

description Inbound to/from CUCM

huntstop

destination-pattern 08627027..

session protocol sipv2

session target ipv4:10.1.1.135

incoming called-number .

voice-class codec 1 

dtmf-relay rtp-nte

dial-peer voice 20 voip

description To-From SIP Proxy

translation-profile incoming Extensions

huntstop

destination-pattern .T

session protocol sipv2

session target sip-server

incoming called-number 08627027..

voice-class codec 1 

voice-class sip profiles 1

dtmf-relay rtp-nte

dial-peer voice 11 voip

description inbound to/from CUCM

huntstop

destination-pattern 2..

session protocol sipv2

session target ipv4:10.1.1.135

incoming called-number .

voice-class codec 1 

dtmf-relay rtp-nte

Translation rules: (without this calls cannot be received internally using DiD)

voice translation-rule 1

rule 1 /^08627027/ /2/

SIP Profile: 

voice class sip-profiles 1

request INVITE sip-header Diversion modify "sip:(.*>)" "sip:0862702734@amcomvoice.ipsystems.com.au>"

request REINVITE sip-header Diversion modify "sip:(.*>)" "sip:0862702734@amcomvoice.ipsystems.com.au>"

I really hope someone can help here, it's driving me crazy..

[Neil Sheridan]

Just and update, and another call for assistance :)

I've managed to get the CFA working using the profile below:

voice class sip-profiles 1

request INVITE sip-header Diversion modify "<sip:(.*)@(.*)>" "<sip:0862702734@amcomvoice.ipsystems.com.au>"

request INVITE sip-header Diversion copy "<sip:(.*)@.*" u01

request INVITE sip-header From copy ".*<sip:(.*)@.*" u02

request INVITE sip-header From modify "(.*)<sip:.*@(.*)" "\1<sip:\u01@\2"

request INVITE sip-header From modify "<sip:@" "<sip:\u02@"

request INVITE sip-header Privacy add "Privacy: id"

Whilst this works it has two issues

1, the caller ID of every diversion comes from "0862702734" which is to be expected seeing as it is hard-coded

2, we do not see the original callers caller-id

My question now is....Is there a way to alter this profile to show the original caller id? I've added the last line to the profile to remove caller id completely as every diverted call was seen as coming from my extension, increasing the amount of calls I received :)

Regards