Hi there,
I've recently set up a Gradwell SIP trunk to receive outgoing and incoming calls. Outgoing all register fine however when trying to ring our DID it doesn't come through and seems to get rejected straight away by 3CX. Please see below, any assistance would be great!
Unidentified Incoming Call. Review INVITE and adjust source identification: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 195.74.60.254;branch=z9hG4bKbd8.9c3e650ba09f60145431747ed9fc7005.0 Via: SIP/2.0/UDP 172.24.1.184:5060;received=172.24.1.184;rport=5060;branch=z9hG4bKPj3d6a99da-e12b-43e4-8252-072612a08668 Max-Forwards: 69 Record-Route: <sip:195.74.60.254:5060;transport=udp;ftag=c258dea8-b347-4274-a205-85e2a94a2a15;did=ec6.eac2;lr=on> Contact: <sip:[email protected]:5060> To: <sip:[email protected]> From: <sip:[email protected]>;tag=c258dea8-b347-4274-a205-85e2a94a2a15 Call-ID: 4b0546de-5f5e-42d4-8728-8bc1cee3a0c2 CSeq: 18288 INVITE Session-Expires: 1800 Min-SE: 90 Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Content-Type: application/sdp Supported: 100rel, timer, replaces, norefersub, histinfo User-Agent: Asterisk PBX 18.22.0-gw1.0.4 X-Inbound-Call-ID: [email protected] Session-ID: 1724336434.d794ac51047d06a3 X-Platform-Type: Hosted SIP Gradwell-Originator: extension=7603753 Gradwell-DDI: 01133200541 Content-Length: 291 v=0 o=- 551617280 551617280 IN IP4 195.74.60.103 s=Asterisk c=IN IP4 195.74.60.103 t=0 0 m=audio 19798 RTP/AVP 8 107 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:107 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:60 a=sendrecv
I've recently set up a Gradwell SIP trunk to receive outgoing and incoming calls. Outgoing all register fine however when trying to ring our DID it doesn't come through and seems to get rejected straight away by 3CX. Please see below, any assistance would be great!
Unidentified Incoming Call. Review INVITE and adjust source identification: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 195.74.60.254;branch=z9hG4bKbd8.9c3e650ba09f60145431747ed9fc7005.0 Via: SIP/2.0/UDP 172.24.1.184:5060;received=172.24.1.184;rport=5060;branch=z9hG4bKPj3d6a99da-e12b-43e4-8252-072612a08668 Max-Forwards: 69 Record-Route: <sip:195.74.60.254:5060;transport=udp;ftag=c258dea8-b347-4274-a205-85e2a94a2a15;did=ec6.eac2;lr=on> Contact: <sip:[email protected]:5060> To: <sip:[email protected]> From: <sip:[email protected]>;tag=c258dea8-b347-4274-a205-85e2a94a2a15 Call-ID: 4b0546de-5f5e-42d4-8728-8bc1cee3a0c2 CSeq: 18288 INVITE Session-Expires: 1800 Min-SE: 90 Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Content-Type: application/sdp Supported: 100rel, timer, replaces, norefersub, histinfo User-Agent: Asterisk PBX 18.22.0-gw1.0.4 X-Inbound-Call-ID: [email protected] Session-ID: 1724336434.d794ac51047d06a3 X-Platform-Type: Hosted SIP Gradwell-Originator: extension=7603753 Gradwell-DDI: 01133200541 Content-Length: 291 v=0 o=- 551617280 551617280 IN IP4 195.74.60.103 s=Asterisk c=IN IP4 195.74.60.103 t=0 0 m=audio 19798 RTP/AVP 8 107 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:107 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:60 a=sendrecv
