Hi,
I have a rule that looks like this:
and it gives calling a go when the call comes in:
but for some reason, the SNOM 320 handset doesn’t actually ring.
Showing the endpoint during the ringing attempt, it seems to know that it isn’t ringing:
[i] Endpoint: 6005/6005 Not in use 1 of inf
InAuth: auth6005/6005
Aor: 6005 1
Contact: 6005/sip:6005@<HANDSET-IP>:44001;line=m5g7ju7r Unknown nan
Channel: PJSIP/6005-00000026/AppDial Down 20
Exten: <EXTERNALNO> CLCID: "<EXTERNALNO>" <<EXTERNALNO>>[/i]
Tracing the PJSIP there are INVITE events going out, though I see nothing in response:
[code][i]<— Transmitting SIP request (920 bytes) to UDP::25825 —>
INVITE sip:6005@:25825;line=zoxehpx5 SIP/2.0
Via: SIP/2.0/UDP :5006;rport;branch=z9hG4bKPjc46e9d72-4e18-43db-af7f-a07d0b384abc
From: “” <sip:@>;tag=f63f6639-c200-4625-854c-b3ce2a89e7f2
To: <sip:6005@;line=zoxehpx5>
Contact: <sip:fadd65a5-ecab-4dab-a1c3-4fd455c4c925@:5006>
Call-ID: 91cff3f8-dbb1-42ae-a930-10921ee17712
CSeq: 19385 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, REGISTER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 241
v=0
o=- 1630659639 1630659639 IN IP4
s=Asterisk
c=IN IP4
t=0 0
m=audio 10402 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv[/i][/code]
Any ideas very much appreciated. I’m new at this game, although admittedly learning fast
Thanks