Calls are not working on Sip relocation.

[waqas sardar]

Hi

Please find the attached debug file and advise what could be the issue.

We have relocated the voice gateway with sip trunk , call is landing on the gateway but it is not going through the isp.

Thanks

[b.winter]

You honestly posting the output of a SIP-call without enabling SIP traces?

Please do it again and use these debugs enabled:

debug voice ccapi ind 1

debug voice ccapi ind 2

debug voice ccapi ind 74

debug ccsip messages

And also post the config (without any sensitive information).

[waqas sardar]

Please find the attached requested logs

calling ext is 082.

Called no : 900 966569607143

[Roger Kallberg]

Not directly related to your issue as such, but I would recommend you to make these changes to your configuration to clean it up and simplify it.

no voice class h323 1
!
voice service voip
 address-hiding
 mode border-element licence capacity 1
!
voice class uri CUCM sip
 host ipv4:xxx.xxx.xxx.xxx ;CPE Publisher IP
 host ipv4:yyy.yyy.yyy.yyy ;CPE Subscriber IP
!
voice class uri PSTN sip
 host ipv4:10.138.120.249 ;siptrunkIP
!
voice class e164-pattern-map 1
 description E164 Pattern Map for called number to CUCM
  e164 [01]..
  e164 333
!
voice class server-group 1
 ipv4 yyy.yyy.yyy.yyy preference 1
 ipv4 xxx.xxx.xxx.xxx preference 2
 description Inbound calls from PSTN to CUCM
!
voice class sip-options-keepalive 1
 description Used for Server Group SIP OPTIONS PING
!

dial-peer voice 201 voip
 description For Incoming Calls form SIP Provider
 no session target sip-server
 no incoming called-number +966115124...
 incoming uri via PSTN
!
dial-peer voice 100 voip
 no answer-address [01]..$
 no destination-pattern [01]..
 no session target ipv4:
 incoming uri via CUCM
!
dial-peer voice 101 voip
 description For Outbound calls to CUCM
 no answer-address [01]..$
 no destination-pattern [01]..
 no session target ipv4:
 destination e164-pattern-map 1
 session server-group 1
 voice-class sip options-keepalive profile 1
!
no dial-peer voice 102 voip
!
no dial-peer voice 103 voip

You can simplify your setup for outbound dial peers to the PSTN as well, but I did not include that for simplicity. In  general it would be to use an e164 pattern map for this as well and then you could collapse these into one instead of multiple as you have now.

[Hudaifa shakkir Nattukallingal]

Hi Waqas.

From the debug I can see that you are sending multiple INVITES to the service provider but there is no response. Please check the service provider.

Thanks

Shakkir

[b.winter]

As @Hudaifa shakkir Nattukallingal already wrote, the CUBE is sending multiple INVITEs, but doesn't get any responses from the service provider.

This can have multiple reasons, but only the service provider can tell you that.

What has been done during the "relocation" exactly?

[waqas sardar]

We have change the voice gateway interface IP and in cucm updated the H323 gateway IP only during the relocation.

[b.winter]

I hope, you have changed the IP in the SIP trunk in CUCM (and not in the H.323 gateway). You don't have any H.323 dial-peers configured on the CUBE.

As per logs, you are sending the called number from CUBE to the provider with leading zero:

Sent: 
INVITE sip:0569607143@ims.mobily.com.sa:5060 SIP/2.0
Via: SIP/2.0/UDP 10.180.7.14:5060;branch=z9hG4bK11A57
Remote-Party-ID: "Nidhi Savla" <sip:+966115124283@ims.mobily.com.sa>;party=calling;screen=yes;privacy=off
From: "Nidhi Savla" <sip:+966115124283@ims.mobily.com.sa>;tag=1AE8F90-1024
To: <sip:0569607143@ims.mobily.com.sa>

Have you tried to send it in +E.164 format? Maybe the provider isn't accepting it with leading zero.

[Hudaifa shakkir Nattukallingal]

Hi,

Are you getting incoming calls,? is your SIP trunk is registered? Can you share the output of show sip-ua register status.

Thanks

[b.winter]

From the config, it seems to be a static SIP trunk towards provider. So, no registration needed.