07-02-2017 10:24 PM - edited 03-18-2019 12:18 PM
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
07-03-2017 08:56 AM
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:+914038123789@10.17.127.113:5060 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.
07-03-2017 09:06 AM
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.
HTH
Regards
Abhay
Kindly rate all helpful posts !!!
07-04-2017 04:51 AM
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....
05-06-2018 12:55 AM
Hi,
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
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.
07-03-2017 03:04 AM
05-04-2018 03:07 PM - edited 05-04-2018 03:09 PM
Hello,
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?
Thanks.
VP
05-05-2018 07:40 AM
You can do it either way, on UCM or GW. Translations on UCM can take effect for internal, inbound or outbound calls.
05-05-2018 08:26 PM
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?
05-06-2018 12:57 AM
i had same issue
i solved by num-exp
apply below command
num-exp 914038123750 750
num-exp 914038123XXX XXX
Saiful Islam
05-06-2018 01:25 AM
number expansion: Seriously!!
05-06-2018 03:15 AM
05-06-2018 03:13 AM
05-06-2018 04:30 AM
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?
05-06-2018 04:58 AM
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.
05-06-2018 07:19 AM
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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