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dial-peer session target sip-server

MeisnerC
Beginner
Beginner

I'm implementing a Multi-tenant configuration on a SIP router that already has a SIP trunk configured on it. On the inbound dial-peer for the current SIP provider it is using "session target sip-server" command to identify the IP of the provider the calls are coming from. I believe this command pulls the IP address listed in the "sip-ua" section of the running config. I have (2) questions. 1) Is what I've said correct? 2) When I move to the multi-tenant configuration can I replace this "session target sip-server" line with "incoming uri via" command? So that I can remove the global configuration of the sip-server from the "sip-ua" section so that the dial-peers will use the Tenant config information to control the dial-peers? 

 

Existing dial-peer

dial-peer voice 1 voip
description Inbound peer match FROM Provider 1
session protocol sipv2
session target sip-server
session transport udp
incoming called-number .T
voice-class codec 1
dtmf-relay rtp-nte
no vad

***********************************

I'll add a voice class uri 10 for example

voice class uri 10

host ipv4:XXX.XXX.XXX

 

 

Tenant config

dial-peer voice 1 voip
description Inbound peer match FROM Provider 1
session protocol sipv2
incoming uri via 10 ---then call it out here
session transport udp
incoming called-number .T
voice-class codec 1
dtmf-relay rtp-nte
no vad

 

Would this be correct?

 

Thanks

 

1 Accepted Solution

Accepted Solutions

Scott Leport
Rising star
Rising star

Hi,

 

1. If that’s what’s configured under sip-ua, yes that’s correct. 
2. Yes and remove the incoming called-number line from your inbound dial-peer. 

Only other consideration might be application of a voice translation profile and applied on this dial-peer to translate the numbers the ITSP is sending you into whatever your extension range is. 

View solution in original post

7 Replies 7

Scott Leport
Rising star
Rising star

Hi,

 

1. If that’s what’s configured under sip-ua, yes that’s correct. 
2. Yes and remove the incoming called-number line from your inbound dial-peer. 

Only other consideration might be application of a voice translation profile and applied on this dial-peer to translate the numbers the ITSP is sending you into whatever your extension range is. 

b.winter
Advocate
Advocate

But the "session target" is not used as an dial-peer matching option, especially not on inbound dial-peers.

It is used to define the target, where to route the call to, on outbound dial-peers.

Hi there, 

 

I completely agree with you , but from his first post I understood that he's going to remove the session target from his inbound dial-peer and replace it with incoming uri via 10.  

It already doesn't have any effect either, since it's not a dial-peer matching option.

He can remove or not. This command is "useless" on inbound dial-peers.

Roger Kallberg
VIP Expert VIP Expert
VIP Expert

SIP session target is an outbound dial peer function and incoming uri via is an inbound dial peer function. As such they are not related to each other. To know more about how to use multi tenants and session target sip-server please have a look at this excellent document.

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



Response Signature


MeisnerC
Beginner
Beginner

Thank you all for your feedback. For my own understanding since the "session target sip-server" command is useless on this dial-peer. This dial-peer is grabbing any call that hits the router matching .T and is passing it on? I will remove the "session target" command and replace it with an "incoming uri via" command which then will specify the source IP address of the inbound calls? As for the voice translation rule. Since this SIP trunk is set up to allow all digits into CUCM would I necessarily need to have one? Thanks

Advice you to remove "incoming called-number .T" from your dial peer once you add the "incoming uri via" as it's redundant and could cause confusion down the line.

About your question about needing a translation or not, that's not a question that anyone that does not have an insight into your CM config can answer. At least not with the information given. If you gave your DNs in CM in +E.164 format and you get the called and calling number from your ITSP in this format as well you would not need any translations.



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