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.
Received: INVITE sip:+firstname.lastname@example.org:5060 SIP/2.0 Via: SIP/2.0/UDP 10.237.250.225:5060;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 !!!
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 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.
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.
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
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.
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?