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sampath9614
Beginner

SIP Provider incoming dial-peer

Hello experts,

This is the first time am configuring SIP trunking (to provider), i have done so many sip trunks to CUCM and other routers but not to the provider.

The problem i am facing is: i have pilot number +91xxxxx700 (pstn numbers 700 - 799)

i need to configure inbound dial-peers so that incoming calls from PSTN will hit the voip extensions.

for ex: +91xxxx750 should ring 750 extension

Which is not happening. If i give +91xxx750 in a translation rule it's not accepting, how should i configure this to work?

voice translation-rule 10
 rule 1 /914038123750/ /750/
 rule 2 /914038123789/ /750/
voice translation-profile 10
 translate called 10

RTR#sh run | s dial-peer
dial-peer voice 901 voip
dial-peer voice 9 voip
 description **Star Code to SIP Trunk**
 destination-pattern 0T
 session protocol sipv2
 session target sip-server
 voice-class sip dtmf-relay force rtp-nte
 voice-class sip outbound-proxy dns:bangalore.relianceims.in
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
dial-peer voice 1 voip
 description **Incoming Call from SIP Trunk**
 translation-profile incoming CUE_Voicemail/AutoAttendant
 session protocol sipv2
 incoming called-number .%
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 no vad
dial-peer voice 10 voip
 description test
 translation-profile incoming 10
 incoming called-number 914038123750
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

36 REPLIES 36

Yes CUBE is required.

 

Saiful Islam

What if I configure the 4331 as an MGCP gateway? Provider SIP to 4331, 4331 MGCP to CUCM. Is there any design I can use that does not require CUBE? The only aspect I cannot control is provider SIP.

 

Addendum... looks like MGCP endpoints can't be used with SIP. Correct?

How do I verify a router is CUBE capable? From the FAQ, CUBE licenses are only on paper. If so, how is a router CUBE-capable?

 

Addendum 2: I figured that one out.

nmx-voicegw1#show cube status
Cisco Unified Border Element (CUBE) Application is not enabled

Ok, so you need to understand the difference first. CUBE is not a separate device. It is just a feature/component running on the IOS on your ISR-4k. If you have an IP to IP termination, you would call that as a CUBE. Now to your other question - MGCP is used to control end points like T1/FXS etc. You cannot use MGCP to control SIP.

You need to add the following to see the CUBE status and have the router act as a CUBE -

voice service voip
mode border-element
allow-connections sip to sip

This requires a reload to take effect.

Nipun Singh Raghav
"We cannot solve our problems with the same thinking we used when we created them"

Thank you Nipun. I found this document and am working through it.

https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/117300-configure-cube-00.html

It doesn't have the provider side configs but I will figure it out.

Thanks for your help.

Vincent

You won't find provider configs on the internet. You need to talk and find out what they need you to configure on the router.
Having said that, you can refer to the following link and see if your ITSP is listed -

https://www.cisco.com/c/en/us/solutions/enterprise/interoperability-portal/networking_solutions_products_genericcontent0900aecd805bd13d.html

Nipun Singh Raghav
"We cannot solve our problems with the same thinking we used when we created them"

Hello everyone... as it turns out you don't need CUBE for SIP to SIP terminating on the same device. I was able to get this working without enabling the 4331 as a border element.

Thank you for all the responses.

CUBE is IP to IP. SIP to SIP is an IP to IP call flow. The device b/w both the legs is called a CUBE. So you are using a CUBE and you will always need one.

Nipun Singh Raghav
"We cannot solve our problems with the same thinking we used when we created them"
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