Cisco Voice Gateway with SIP Digium Switchvox interoperability!

[GASHAW TURA]

Project scenario to replace the existing ISDN/PRI PBX with Digium switchvox SIP PBX.

1) **Existing working system is PRI PBX --------Cisco IOS Gateway -------MPLS-----VZB SBC.

 New scenario:

2) SIP PBX Digium Switchvox--------- Cisco IOS Gateway ----------MPLS -------VZB SBC.

Change made to replace the PRI dial-peer:

===================Dial-peer to VZB side=====================

dial-peer voice 100 voip

 description OUTBOUND Voice SIP calls

 translation-profile outgoing DIGITSTRIP-9

 destination-pattern 9T

 rtp payload-type nte 98

 session protocol sipv2

 session target sip-server

 voice-class codec 1  

 voice-class sip early-offer forced

 voice-class sip bind control source-interface GigabitEthernet0/1

 voice-class sip bind media source-interface GigabitEthernet0/1

 dtmf-relay rtp-nte

 no vad

!

dial-peer voice 400 voip

 description voip dial peer - from proxy

 rtp payload-type nte 98

 session protocol sipv2

 session target sip-server

 incoming called-number .T

 voice-class codec 1  

 voice-class sip bind control source-interface GigabitEthernet0/1

 voice-class sip bind media source-interface GigabitEthernet0/1

 dtmf-relay rtp-nte

 ip qos dscp cs5 media

 ip qos dscp cs3 signaling

 no vad

=========================== Dial-peer pointing to Digium Switchvox side =========================

dial-peer voice 500 voip

 description switchvox

 preference 1

 destination-pattern 526788….

 session protocol sipv2

 session target ipv4:10.x.x.x

 session transport tcp

 voice-class codec 1  

 voice-class sip bind control source-interface GigabitEthernet0/2

 voice-class sip bind media source-interface GigabitEthernet0/2

 dtmf-relay rtp-nte

!

dial-peer voice 550 voip

 description switchvox

 preference 1

 session protocol sipv2

 session target ipv4:10.x.x.x

 session transport tcp

 incoming called-number 526788….

 voice-class codec 1  

 voice-class sip bind control source-interface GigabitEthernet0/1

 voice-class sip bind media source-interface GigabitEthernet0/1

 dtmf-relay rtp-nte

 no vad

Result after the change-

1) Out going call works successfully without issues both signaling and media.

Problem-

2) Incoming call fails on the Digium switchvox after 407 challenge from the digium switchvox. The digium log shows the failure reason is unknown source that tells it was failing the failing during 407 challenge.

Question- Can we configure the Cisco IOS gateway (C2911) to receive the 407 challenge and replay with another re-invite with nounce value or blank?

[Ayodeji Okanlawon]

Can you post a debug ccsip messages from the CUBE? Include calling and called number. Can you also attach the sip log from the PBX?

[GASHAW TURA]

I will do more test tonight and update this post. Digium mightn't allow us disabling the digest authentication for security reason. I am thinking also to include another change on the sip configuration on Cisco gateway "header-passing error-pass through" to send the 407 challenge to the service provider and hope they will respond with another re-invite.

[GASHAW TURA]

Ayodeji- For some reason the ccsip debug all/calls wasn't showing the call flow. I have tried to configure "header-passing error-pass through" but no luck because it isn't supported on C2911 IOS 15.0. I will do more research at this point or work directly with the provider to bypass this gateway as it is with limited capabilities.

[Vivek Batra]

Is it working if you disable digest authentication in Digium?

Thanks

Vivek