I have a BE6K running CUCM 11.5 in the Middle East. We have found that we are unable to Call Forward externally and I think this is down to the fact that the invite is coming the external party calling in. So for example Phone A is forwarded to Mobile B, Mobile C from outside the company calls Phone A and instead of forwarding to Mobile B I receive a message from the SIP provider that the number we are trying to reach is not available. The invite to the Mobile B is from Mobile C and not Phone A, so I think the SIP provider may not trust this as it wants to see calls originating from numbers it knows about.
I have seen some links regarding similar issues such as this one https://community.cisco.com/t5/collaboration-voice-and-video/configure-and-troubleshoot-call-forward-to-the-pstn-using-sip/ta-p/3118287
however the first option which I would prefer to try and use, or through reconfiguring CUCM rather than the cube, failed to make any difference.
I have included a picture of our Outbound Call settings if this is useful. I also have the logs from a call if anyone requires them? I have tried changing the 'Calling Party Selection' to 'Last Redirect Number (external)' as suggested in the link above however this made no change.
Any help much appreciated!
The call is actually getting out to the PSTN/SIP Provider, it is their recorded message stating that the number dialled is not valid, however it is valid and sent to them in a valid format that when simply dialled from a desk phone then the call connects. I believe the issue is that the call appears to the SIP provider to be coming 'from' the mobile that called in initially rather than the phone that is forwarding on the call. So the call appears to be from an unknown number to the SIP provider so they stop the call connecting.
I need to know how to configure CUCM so that the invite comes from the phone forwarding the call and not the external caller calling in.
I had a similar issue and i fixed it with translation profile,
voice translation-rule 1
rule 5 /^.*/ /your masternumber of sip trunk/
voice translation-profile profile1
translate calling 1
apply this profile to outgoing dialpeer