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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
 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

Exactly bro... i have logged a ticket with cisco on this..

they have told me that, whatever the number is dialed, invite is coming to the above number whereas the actual dialed number is showing in to field.

INVITE sip:+914038123789@ SIP/2.0

Via: SIP/2.0/UDP;branch=z9hG4bK-*2*1f54423880477d7bf42etaN0
To: <tel:+914038123710>
From: <tel:+919642000814>;tag=ztesipsaiS6fSEeoW3fALsksBp1KF*1-1-20481*dfcj.1
Call-ID: -tuOuejUuqcPZhHcchAU0bOo6a8umNbOb3-kOF-2xM1ehbb@zteims
CSeq: 1000 INVITE
Max-Forwards: 65

As the invite is coming to the board number only, none of the other numbers are able to ring. If a call to that board number itself then i can able to ring an ip phone without any issues.

This is compeltely weird for me and don't know how to move forward. The cisco tac engineer tried to modify the SIP Header, but not successful because of the CME functionality (not cube).

Please guide any other ways..

I am thinking of the following:
1. Auto attendant with board number
2. Any ways to sort out SIP Header modification to bring the number in to field to SIP Invite field.

I would suggest to get in touch with the provider as well and check with them in regards to number that they are sending in the SIP header. They should be able to modify at their own end as this is not desired and causing trouble.

Keep this post updated.




Kindly rate all helpful posts !!! 

Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle

Dear Abhay,

I have spoke to the provider, they are not willing to make the changes at their end.

Could you please tell me any other possible ways....


Some providers especially in india send the called number in the To header. All you need to do is copy the To header into the Request-URI.

Here is how to configure inbound sip profiles to achieve this

voice service voip


sip-profiles inboud

sip-profiles 10 inbound


voice class sip-profiles 10

rule 1 request INVITE sip-header To copy "sip:(.*)@.*" u01

rule 2 request INVITE sip-header Request-URI "sip:.*@(.*)" "sip:\u01@1"


Now you can apply any translation rule you want to the called number.

Please rate all useful posts

please find the attached latest logs...



I've never setup SIP trunk to service provider and am curious about your configuration.

Isn't inbound translation done by CUCM under Translation Pattern or do you have to configure it on both CM and the gateway?



You can do it either way, on UCM or GW. Translations on UCM can take effect for internal, inbound or outbound calls.



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

Thank you, Nipun. I have to configure a Cisco 4331 for SIP to provider and already did the SIP to CUCM. I looked at the configuration of the OP and see the session target is sip-server. Where is the sip server configured? Does any have a sample of outbound and inbound SIP to provider configuration?

i had same issue

i solved by num-exp

apply below command

num-exp 914038123750 750

num-exp 914038123XXX XXX


Saiful Islam

number expansion: Seriously!!

Please rate all useful posts

Number expansion are a quick fix but not the best solution for anything for any translations. There is so much more that IOS offers now that there is no need for the use of number expansion nowadays. At least not on CUBE or an H.323/SIP GW in general.

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

You configure your sip server under "sip-ua" sub-configuration. Once defined here, you can then use "session-target sip-server" on your dial-peers. This is usually your ITSP SBC IP.

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

Thank you, everyone.

Using your replies and some other research, I configured the 4331. My biggest question now (and I will be calling Cisco later today) is can I use the 4331 for SIP to provider without CUBE. If the router takes all the commands, do I still need CUBE?

Your 4431 can act as a voice gateway or CUBE. CUBE simply means you have either h323 to sip or sip to sip terminating on same device. If your 4431 is a CCME, then you can simply have a SIP trunk to your ITSP and it will work fine. 

Please rate all useful posts

The 4431 is a gateway that registers to CUCM 10.5 It is not CCME.

SIP provider to 4431 to CUCM. IF SIP from both the provider and CUCM terminate on the 4431, does that require CUBE?

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