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CUBE VOIP Inbound Dial-peer Bind Commands

fedor.solovev
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Hello, dear support community.

I get used to configure separate inbound and outbound dial-peers. GW is connected to more the one CUCMs.

 

1.

I would like to find out if a bind command works for incoming dial-peers ?

( I would like to separate dial-peer matching for incoming calls from different CUCMs).

I would probably try some tests with debug voip dialpeer; however, I really wonder if it is configurable.


GW#sh run | s 4050
voice class codec 4050
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 ilbc
codec preference 4 g729r8


dial-peer voice 4050 voip
description -= Incoming from CUCM =-
shutdown
session protocol sipv2
incoming called-number 8T

voice-class codec 4050
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 0 fallback pass-through g711alaw
no vad

 

2. 

Maybe we may bind incoming called-number with IP as well in order to match incoming dial-peers based on Different sources ?

2 Accepted Solutions

Accepted Solutions

Chris Deren
Hall of Fame
Hall of Fame

What IOS version are you using?

Easiest way assuming recent IOS is to use voice class uri to match IP address of your CUCM nodes and then have separate dial-peers matching based on it, example:

 

voice class uri 10 sip
host ipv4:x.x.x.y 
host ipv4:1x.x.x.x

 

dial-peer voice 10 voip
description CUCM dial-peer - Inbound
session protocol sipv2
incoming uri via 10
voice-class codec 1
dtmf-relay rtp-nte sip-kpml
fax-relay ecm disable
fax nsf 000000
ip qos dscp cs3 signaling
no vad

View solution in original post

URI matching has preference, see

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/cube-dp.html

Preference for Dial-Peer Matching

The following is the order in which inbound dial-peer is matched for SIP call-legs: 

  •  

    voice class uri URI-class-identifier with incoming uri {viaURI-class-identifier

     

  •  voice class uri URI-class-identifier with incoming uri {requestURI-class-identifier

     

  •  voice class uri URI-class-identifier with incoming uri {toURI-class-identifier

     

  •  voice class uri URI-class-identifier with incoming uri {fromURI-class-identifier

     

  •  incoming called-number DNIS-string

     

  •  answer-address ANI-string

View solution in original post

7 Replies 7

Chris Deren
Hall of Fame
Hall of Fame

What IOS version are you using?

Easiest way assuming recent IOS is to use voice class uri to match IP address of your CUCM nodes and then have separate dial-peers matching based on it, example:

 

voice class uri 10 sip
host ipv4:x.x.x.y 
host ipv4:1x.x.x.x

 

dial-peer voice 10 voip
description CUCM dial-peer - Inbound
session protocol sipv2
incoming uri via 10
voice-class codec 1
dtmf-relay rtp-nte sip-kpml
fax-relay ecm disable
fax nsf 000000
ip qos dscp cs3 signaling
no vad

Hello, Chris.
Thank you for quick response.
this is 29xx GW, flash0:c2900-universalk9-mz.SPA.154-3.M.bin

 

Yes, I have heard about this way of "binding".

It looks even more precisely in comparison with bind command. I think I will use it.

 

But I would like to match exact digits as well along with matching IP addresses.
It probably will be needed to produce some digit translations.

 

Does bind command should work for incoming calls ?

This has nothing to do with binding, if you are using different interfaces for ingress traffic from different CUCM clusters (not sure what you would do that) and need to bind the ingress traffic to specific interface you can add bind statements on those ingress dial-peers.

Typically binding is required for outbound traffic, i.e. separate interfaces for different carriers in which case you add the bindings on egress (towards ITSP) dial peers.

Hello, Chris.

(this is some middle step to switch from remote CUCM to local one).

Anyway, I think I got your point.
In addition to this, I am not sure, what GW matchs against first: incoming uri via 10 or incoming called-number ?

 

That is, if GW has small dial-peer and incoming called-number verifies first:

dial-peer voice 1 voip
incoming called-number .
voice-class h323 1
dtmf-relay h245-alphanumeric
codec g711ulaw

I am thinking that uri wouldn't help, and it is necessary to use more precise incoming called-number 
or remake all incoming dial-peers to match uri.

URI matching has preference, see

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/cube-dp.html

Preference for Dial-Peer Matching

The following is the order in which inbound dial-peer is matched for SIP call-legs: 

  •  

    voice class uri URI-class-identifier with incoming uri {viaURI-class-identifier

     

  •  voice class uri URI-class-identifier with incoming uri {requestURI-class-identifier

     

  •  voice class uri URI-class-identifier with incoming uri {toURI-class-identifier

     

  •  voice class uri URI-class-identifier with incoming uri {fromURI-class-identifier

     

  •  incoming called-number DNIS-string

     

  •  answer-address ANI-string

Hello, Chris.
Thank you for the actual link.

Hi Experts
I have a problem between my Gateway (ISR 4431 Remote Site) and a Call Manager, I have it registered by sip Trunk to my call Manager, but from one moment to the next it loses connection unexpectedly, therefore the calls in the remote site fall, they are seconds or up to 1 minute losing connection to the CM, then it registers normal.
I opened the case with Cisco and indicated that the Gateway loses connection with the CM and indicated me to apply the commands that are below, this would make it feel the ISR with the Call Manager and the SIP TRUNK would not fall.

Apply these commands and a Dial-peer that I have configured for calls to cellular numbers through a Licea with chips did not work. I thought that I could have duplicity in the Dial Peer but it was not like that, I had to remove the configuration so that the cellular calls work.

Exactly these commands for what are they?

dial-peer voice 50 voip
 description to keep sip trunk up
 session protocol sipv2
 incoming uri via 50
 voice-class sip bind control source-interface GigabitEthernet0 / 0/0
 voice-class sip bind media source-interface GigabitEthernet0 / 0/0


voice class uri 50 sip
 host ipv4: x.x.x.x (IP CUCM)
 
Regards.
Carlos P.
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