I’m in the process of changing over to FreePBX 17 - I’m currently using FreePBX 16. With that process I’m changing my SIP trunks to PJSIP however my inbound DID’s now show as an ‘s’ rather than the DID line this is with two different Trunks (sipgate and Andrews & Arnold)
I’ve tried enabling ‘Send Connected Line’ and ‘Trust RPID/PAI’ which I read somewhere but that didn’t work.
Any ideas how to fix this please ?
If there is only one DID on a trunk, set Contact User to the DID number.
Otherwise, try setting Context to from-pstn-toheader
Many thanks Stewart that seemed to do the trick.
Setting ‘Contact User’ to the trunk DID number. I do some further testing and mark this thread solved if it’s ok, the caller ID now has the added DID truck in brackets on one of the trunks.
Out of curiosity what does the ‘s’ mean or represent ?
“s” is for “start” as in where to start if no dialed number is known.
Thanks Josh, That explains a lot, I assumed it was a provided issue at first. Do you know what Stewart means with ‘Context to from-pstn-toheader’ - I have context set to ‘from-pstn’. Maybe he’ll answer. I just wanted to learn a bit more and about more than one DID per trunk.
Within SIP it’s supposed to be that the request URI contains the dialed phone number. This is where Asterisk looks. Some providers don’t do that, though, and instead put it in the To header. Asterisk doesn’t look at that header for such information so instead FreePBX has a “from-pstn-toheader” context which will get the dialed phone number from there instead.
Ah ok got it. You live and learn - Thanks again !
(next project is to try again to get Localphone working PJSIP from SIP)
To add to the above, this is because the way that SIP is designed, each DID should register separately, when using registration, but many providers treat the address of record in the registration as an account name and forward all traffic for that account. As the request URI should come from the registering device, that leaves them with nowhere for the DID.
I’ve used the term DID in the VoIP industry sense, however it is not a term used in the SIP specifications.
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