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SIP Trunk port setup on cucm for 2 service provider

CUCM ---> SIPTrunk 5060 ------>CUBE -------> Service provider 1

CUCM-----SIP Trunk 5060---->Right Fax

CUCM------>SIP Trunk 5060----Same CUBE---->Service Provider 2

Will this feasible with different dest IP address or need to assign different sip port for each one?

It would be grateful if you provide your thoughts on followings:

  • •1. Can we change sip trunk port on CUCM to 5070 or (5062 - 5069) and if yes how it could be? Currently we set to 5060 and if we try change to 5062-5070, it’s not working as getting fast busy tone and calls are not reaching upto CUBE itself.
  • •2. We have set up of 1st sip trunk with one service provider with 5060 port for both inbound/outbound and we have another SIP trunk for Right fax with 5060. Will this Config of same port for 2 different setup feasible for future perspective.
  • •3. In future we will have 2nd SIP Trunk from another service provider, during those scenario can we place same port 5060 for 2nd SIP trunk which will be directing to same CUBE which we are using now for 1st SIP trunk.
  • •4. What could be the advantages/Disadvantages if we use same port for two sip trunk and Right fax?

Thanks

Regards,

Sam

1 ACCEPTED SOLUTION

Accepted Solutions
Cisco Employee

SIP Trunk port setup on cucm for 2 service provider

Hi Sam,

Yes, I believe two different SIP listen ports is not possible in CUBE.

You can bind the Control & Media interfaces to be different in dial-peer basis.

In Cisco Unified Sip Proxy (CUSP) it is possible to have different sip listen ports for different networks.

Regards,

Senthilkumar.

4 REPLIES
Cisco Employee

SIP Trunk port setup on cucm for 2 service provider

Hi Sam,

Once you have changed the SIP Port as 5070 in CUCM, You need to config you CUBE also to listen the sip traffic on the same port number.

You can achieve this by giving the below command in your CUBE

voice service voip

     sip

          listen-port non-secure 5070.

This will be used for both incoming and outgoing sip calls. Also we cannot give this on dial-peer basis.

You could use the below link for more clarification and restrictions for changing this sip-listen port.

http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-sipsip.html#wp1389955

Regards,

Senthil

New Member

SIP Trunk port setup on cucm for 2 service provider

Hi Senthilkumar,

Thanks for reply, Yes that setting we did it during testing.

I am wondering to find out to see the answer for the following,

We have Right Fax server with SIP Trunk port 5060, today we tested SIP Trunk with CUBE with 5060 and both are working. In future we have another SIP Trunk from 2nd service provider. In that case port 5060 will work for all scenario?

I know for 2 service provider, it's feasilbe and we can seggregate the traffice for two service provider by priorities the traffic on dial peers, but just wanted to confirm about the port 5060?

dial-peer voice 1 voip

description calls to SP 1st service provider

"bind to Loopback0"

dial-peer voice 2 voip

description calls to SP 2nd service provider

"bind to Gi 0/0"

and voice-class sip bind control source-interface GigabitEthernet0/0 and L0

Thanks

Regards,

Sam

Cisco Employee

SIP Trunk port setup on cucm for 2 service provider

Hi Sam,

Yes, I believe two different SIP listen ports is not possible in CUBE.

You can bind the Control & Media interfaces to be different in dial-peer basis.

In Cisco Unified Sip Proxy (CUSP) it is possible to have different sip listen ports for different networks.

Regards,

Senthilkumar.

New Member

Re: SIP Trunk port setup on cucm for 2 service provider

Hi Senthikumar,

Thanks for the information. I have another issue for fax send/receive through Right fax from cube.

Right fax ---SIP 5060 -----CUCM 8.0.3-------- SIP Trunk 5060---CUBE----SP

I made config for fax T.38 as follows on cube, and getting calls for fax on cube but unable to send/receive faxes. Status - Fax number busy

if I try to send/receive fax through Mgcp gateway from Right fax, it's working fine.

voice service voip

mode border-element

allow-connections sip to sip

redirect ip2ip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw

sip

early-offer forced

midcall-signaling passthru

sip-profiles 1

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