10-25-2018 09:40 AM
Hi all,
I have some doubts about matching criterias in inbound voip dial-peers, specifically when I need to configure two different inbound voip dial-peers for receiving messages from different SIP servers.
I have the following configuration (summarized) in a voice gateway (not SBC):
voice class server-group 1
ipv4 CUCMIP1 port 5060 preference 1
ipv4 CUCMIP2 port 5060 preference 2
!
voice class server-group 2
ipv4 SBCIP1 port 5060 preference 1
ipv4 SBCIP2 port 5060 preference 2
!
dial-peer voice 1000 voip
translation-profile incoming IncomingCalllegHCS
session protocol sipv2
incoming called-number .T
session server-group 1
!
dial-peer voice 1100 voip
session protocol sipv2
incoming called-number .T
session server-group 2
!
I am not sure if I can use the "session server-group" command to match the IP servers from which the SIP messages must be received, to associate the incoming call-leg to the correct dial-peer (take into account that I have to apply a translation-profile in dial-peer 1000, but I haven't in dial-peer 1100).
According some info resources, "session server-group" command is used in outbound dial-peers, and the correct way to match IP servers in an inbound dial-peer is creating a "voice class uri <tag>" and apply it to the dial-peer ("incoming uri via <tag>").
But according other resources, if I apply a voice class to an inbound voip dial-peer, I lost number information and I couldn't apply a translation-profile to change calling or called number. The basis for saying that is the matching priority criteria for an inbound voip dial-peer:
1. Match based on URI of an incoming INVITE message.
2. Match based on Called Number
3. Match based on Calling Number
4. Default Dial-Peer = 0
Which would be the correct configuration, really?
Thank you very much. Best regards.
10-30-2018 11:55 AM
The correct way is to use 'voice class uri'.
See this post and test the dial-peer with voice translation rule. Should work.
I'va a question. Is the traffic from different IPs received on the same interface?
Regards.
10-31-2018 02:26 AM
Thank you Daniele.
No, the SIP traffic from different IPs is received on different interfaces.
Best regards,
Patricia
10-31-2018 11:17 AM
Have you try 'voice calss uri'?
Is it working?
Is this a voice gateway with TDM interfaces (isdn, fxs, fxo)?
Or is it a ip2ip gateway?
Have you configure 'allow-connections sip to sip?
In the past I used a trink in an ip2ip gateway to apply PBR and translation rule to a specific traffic received from a specific interface. I don't know if you can use this trick but you can study it on this link:
Regards.
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