Good Afternoon All!

I’m hoping someone may have some better insight and possibly suggestions for me.

I’m integrating my FreePBX 12 with Geneys Pure Cloud to allow me to take advantage of some of their features. I’ll be passing my inbound calls to their platform via a sip trunk (“tie line”), and then after the caller goes through an IVR on that platform it will send the call back to my FreePBX box via a SIP REFER. I’ve never worked with SIP REFER in this manner and am running into some difficulties.

Essentially, what is happening is that I see the SIP REFER coming in from GPC, and then a response of 603 Declined (Non sip: uri). The transfer then completes but doesn’t do the refer so it is tying up two ports.

The Refer-To field is simply tel:+1XXXXXXXXXX and not a URI. From reading the RFC, it seems like that is not allowed, but from other Google searches it appears it is.

The Contact URI is my valid sip:[email protected]:5060;transport=udp.

The From is the valid from as it is coming from GPC with the full sip uri containing the DNIS I passed into GPC on the originating call.

The To is the full sip uri with the ANI from the originating caller @mypbxip

I hope this is enough information to get started.

My trunk only has the peer configured.

Thank you for any assistance!

I’m not 100% if current versions of Asterisk support the tel: format for this or not. I’m going with not. However, you’re on FreePBX v12 which means that your versions of Asterisk are unsupported and probably haven’t been touched in at least over a year or more (and won’t be).

Bottom line, your current PBX cannot support using the tel: format. Asterisk will require the REFER to be a valid SIP URI.

Thank you! I had a feeling that was the issue!

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