Elrick B.

Ok thanks

After changing that option in text mode. It fixed both issues (Calls ending, static when resuming call on hold). Was that listed in the change logs? I didn't see it.

Sorry for taking so long to get back to you. Inbound calls are the same, it rings once quickly then ends the call. I will try your solution now.

No longer have the pcap from the system as I uninstalled and reinstalled the PBX since then, but I do have Invite and Response from the Telnyx side: INVITE sip:+1813xxxxxxx@sip.telnyx.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 34.x.x.x:55514;branch=z9hG4bK-2efef9a6c828658f720b980265180c38;rport From: <sip:+1656xxxxxxx@sxxxxx.xxxxxxcloud.com;user=phone>;tag=909344563 To: <sip:+1813xxxxxxx@sip.telnyx.com;user=phone> Call-ID: 0gpsc8s@pbx CSeq: 9594 INVITE Max-Forwards: 70 Contact: <sip:xxxxxxxxx@34.x.x.x:55514;transport=tls> Allow-Events: refer Supported: 100rel, replaces, norefersub Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Vodia-PBX/69.1.1 Remote-Party-ID: <sip:+1656xxxxxxx@sxxxxx.xxxxxxcloud.com;user=phone> Proxy-Authorization: Digest realm="sip.telnyx.com",nonce="29c73b43-ae74-4f53-9f86-3cef85b7aabe",response="51404a08861a08388b0e023d2d6dfbcd",username="stxxxxxxx",uri="sip:+1813xxxxxxx@sip.telnyx.com;user=phone",qop=auth,nc=00000001,cnonce="373284d9",opaque="fbe7a55b-5951-449e-a422-bc203fdedf9f/10.13.37.24",algorithm=MD5 Content-Type: application/sdp Content-Length: 372 v=0 o=- 2103907024 2103907024 IN IP4 34.x.x.x s=- c=IN IP4 34.x.x.x t=0 0 m=audio 56512 RTP/AVP 0 8 109 9 101 c=IN IP4 34.x.x.x a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1 a=fmtp:101 0-15 a=rtpmap:109 opus/48000/2 a=rtpmap:9 G722/8000/1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000/1 a=sendrecv SIP/2.0 180 Ringing Via: SIP/2.0/TLS 34.x.x.x:55514;received=34.x.x.x;branch=z9hG4bK-2efef9a6c828658f720b980265180c38;rport=55514 Record-Route: <sip:10.255.0.1;transport=tcp;r2=on;lr;ftag=909344563> Record-Route: <sip:192.76.120.10:5061;transport=tls;r2=on;lr;ftag=909344563> From: <sip:+1656xxxxxxx@sxxxxx.xxxxxxcloud.com;user=phone>;tag=909344563 To: <sip:+1813xxxxxxx@sip.telnyx.com;user=phone>;tag=ep3yQgQ8mmKZa Call-ID: 0gpsc8s@pbx CSeq: 9594 INVITE Contact: <sip:+1813xxxxxxx@10.13.37.24:5070;transport=tcp> Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: path Allow-Events: talk, hold, conference, refer Content-Length: 0 X-Telnyx-Session-ID: 3b30dd96-74f7-11ee-bb75-02420a0de568 X-Telnyx-Leg-ID: 3b30d468-74f7-11ee-8f87-02420a0de568

Not to stray too far from the OPs issue. When using our Yealink phones on the latest build with Telnyx on TLS, calls are not completing. It tries to ring then fail, and after checking the PCAP on the system and on Telnyx, it gave a "Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION", as if they are not able to negotiate codecs. I am guessing something happened between version 68.0.32 and 69.1.2 that doesn't play nice with Telnyx. I also have an issue when putting incoming calls on hold, once resumed its pure static. That issue is not present when placing outbound calls on hold though. Both issues are resolved if I roll back to v68.0.32.