I’m on FreePBX 14.0.11
My sip provider restricts the “From:” header in the INVITE to exactly match the “From:” header in the REGISTER message.
In my case, the “From:” headers look like this:
From: <sip: myusername @ myprov. com>;tag = blablabla
From: <sip: myusername @ myprov. com:5160>;tag = as0e796078>; tag = anotherblablabla
As we see here, the asterisk adds “: 5160” to the INVITE.
And my provider declares that this behavior (then “From:” is different in “INVITE” and “REGISTER”) is not an RFC valid and rejects my “INVITES”.
If I switch chan_sip from the default port 5160 to port 5060 (and disable pjsip), then everything works.
But I can’t change the default port for chan_sip, because I already have tons of sip clients configured on port 5160.
I killed a few hours for:
- tried to remove the postfix “:5160” from the header, manipulating the sip parameters for the trunk. Unsuccessful.
- tried to add the postfix “:5160” in the header, manipulating the sip registration string. No luck.
- In the Dial application, there is an option that allows to change the header, but I do not know how I can change it in FreePBX for a particular trunk.
- tried to use pjsip for this provider, but failed. My provider expects a username in the form [email protected], but pjsip does not like it and complains:
Unable to create outbound OPTIONS request to endpoint myprovname as URI ‘sip:[email protected]@[email protected]:5060’ is not valid
- Google, no answers.
I do not want to make changes directly to the asterisk configuration files, as this does not seem to be the right way.
As a workaround, I will create a gateway with a clean asterisk between freepbx and the provider, but I don’t like this solution too.
Can someone provide an easy way to change the header either in the invite or register messages for my case?
Any other help is appreciated.