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How to configure two sip trunks

Hi Team,

 

I have two SIP Trunks to configure on One CUBE but the problem is the first SIP Trunk needs SIP Registration and the second one does not. 

when I am placing call through second SIP Trunk the "From address" in Invite MSG from CUBE  carries the IP Address of First SIP Trunk Register Server and due to this my second SIP Trunk provider is rejecting the call by giving error "SIP/2.0 484 Address Incomplete"

 

Thanks in advance.

Regards

Dharmendra

 

1 ACCEPTED SOLUTION

Accepted Solutions

Roger Kallberg
VIP Mentor VIP Mentor
VIP Mentor

As @Scott Leport pointed out you’ll need to use tenant configuration to get around this. Please have a look at outstanding document for more details on various details on this.

In Depth Explanation of Cisco IOS and IOS-XE Call Routing 



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3 REPLIES 3

Scott Leport
Rising star
Rising star

Hi there, 

 

Voice class tenants should sort this for you. The example in the link below is if you have multiple SIP trunks which require registration, but same principle applies. 

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/ios-xe/config/ios-xe-book/m_voi-cube-multi-tenants.html

 

Just make sure that your CUBE is on the right IOS or IOS-XE version:

 https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html#anc28

 

Roger Kallberg
VIP Mentor VIP Mentor
VIP Mentor

As @Scott Leport pointed out you’ll need to use tenant configuration to get around this. Please have a look at outstanding document for more details on various details on this.

In Depth Explanation of Cisco IOS and IOS-XE Call Routing 



Response Signature


TONY SMITH
Collaborator
Collaborator

Voice class tenants will indeed sort out complex configurations.  However for one registered and one unregistered trunk you don't actually need this.  For the dial peer(s) handling the non-registered trunk, just put the destination IP address.  And make sure any proxy or other over rides applied at the "voice service" level are switched off for this dial peer.

Maybe you could share your configuration?

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