[Freeswitch-users] Manipulate the sip: string in the INVITE header on an outgoing Bridge

[Joseph Waite]

Hi Guys

Were having issues with Asterisk boxes registering to a trunk on our FreeSwitch box.

We are sending the DID number called down the trunk in the to: field, however Asterisk doesn’t seem to like this.

Is there anyway to manipulate the INVITE to be INVITE sip:DID@ <sip:DID@> instead of what is currently INVITE sip:user@ <sip:user@>

Im hoping there is something like sip_invite_to_uri, however google is failing me.

Regards
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