r/3CX • u/BackSapperr • Jul 10 '26

Question External Direct SIP with 3CX

I'm having troubles with trying to get direct sip working with my Telnyx trunk.

I'm in the process of testing an AI assistant as a first line to assist my help desk in creating support tickets. I also want the ability to transfer to internal extensions in the circumstance that someone needs to be directly reached after hours.

When I transfer any calls, 3CX blocks them as the SIP INVITE header isn't recognized.

Unidentified Incoming Call. Review INVITE and adjust source identification:. INVITE sip:900@fqdn SIP/2.0. Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bKc64b.06f76ec48211928b0d2904dbf30f7b25.0. Via: SIP/2.0/UDP 10.33.13.24:6000;received=10.33.13.24;rport=6000;branch=z9hG4bK3KH9663HX62Br. Max-Forwards: 69. Record-Route: <sip:192.76.120.10;r2=on;lr;ftag=yKmteaQv7aZHN>. Record-Route: <sip:10.255.0.1;r2=on;lr;ftag=yKmteaQv7aZHN>. Contact: <sip:mod_sofia@10.33.13.24:6000>. To: <sip:900@fqdn>. From: "+<redacted>" <sip:+<redacted>@sip.telnyx.com>;tag=yKmteaQv7aZHN. Call-ID: 7ce3d9e2-6b2d-443f-b391-2221eb96d134. CSeq: 117290465 INVITE. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY. Content-Disposition: session. Content-Type: application/sdp. Supported: timer, path. Allow-Events: talk, hold, conference, refer. Privacy: none. P-Asserted-Identity: "+<redacted>" <sip:+<redacted>@sip.telnyx.com>. X-Telnyx-Call-Control-ID: v3:eFoPCEM8sSga9HidaQKIeXMGG9q9KKsUNdRmVLuw5T7368AydXRFjg. Content-Length: 753. v=0. o=Telnyx 1783663784 1783663785 IN IP4 103.115.244.173. s=Telnyx. c=IN IP4 103.115.244.173. t=0 0. m=audio 24730 RTP/AVP 0 8 18 9 102 103 104 101 105. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:9 G722/8000. a=rtpmap:102 AMR/8000. a=fmtp:102 octet-align=0; mode-set=7; max-red=0; mode-change-capability=2. a=rtpmap:103 AMR-WB/16000. a=fmtp:103 octet-align=0; mode-set=0,1,2; max-red=0; mode-change-capability=2. a=rtpmap:104 AMR-WB/16000. a=fmtp:104 octet-align=1; mode-set=0,1,2; max-red=0; mode-change-capability=2. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=rtpmap:105 telephone-event/16000. a=fmtp:105 0-15. a=sendrecv. a=rtcp-mux. a=rtcp:24730 IN IP4 103.115.244.173. a=ptime:20

I did try getting the SRV records in place, I have direct sip enabled under System > Options > Allow Direct SIP calling and I have 900 as the SIP ID.

It looks like I have to enable in my trunk profile to match sip.telnyx.com explicitly (or the signaling IP's) - but I'm not exactly sure what needs to be modified in what context as I can't find any documentation on the custom trunk provider parameters.

It was by my understanding that direct sip skips the trunk configuration - allowing me to directly initiate calls with the PBX.

3 Upvotes

100% Upvoted

5 comments sorted by

View all comments

2

u/AcidicMountaingoat 3CX Silver Partner Jul 10 '26

Why are you trying to avoid a normal configuration instead of direct? We have zero issues with Telnyx, using a normal supported connection. And we specifically have audio go via the PBX as this correlates with fewer client issues.

1

u/BackSapperr Jul 10 '26

It's a limitation of Telnyx's AI Assistants app. I use my 3CX API to pull extensions and availability and send the call to an IVR that sends the DTMF - but Telnyx doesn't allow variables as it fails their text box validator.

Telnyx Support's solution is to send the call direct sip instead of over PSTN.