I believe that with firmware 2_24_2, our P320 don’t always dial longer numbers anymore.
Sometimes a full 11 digit number goes through, sometimes even 5 digits don’t. Shorter numbers always work.
When that happens, absolutely nothing appears in asterisk -rvvvv.
Am I on a hiding to nothing?
Hey @kombi2
Welcome to the forum. No known complaints/feedback as such, we will give a try locally.
Thanks
Kapil
Hi @kombi2 During the issue, you need to enable the SIP debug log on Asterisk to confirm whether it is reaching Asterisk. If the request is failing at the SIP level, it will not show on Asterisk cli without debug enabled.
if the problematic extension is chan_sip then run “sip set debug on” else run “pjsip set logger on”
on asterisk cli to enable the sip logs.
when a call goes through (one in 10) it look like this:
[2024-11-18 17:32:00] VERBOSE[2141] res_pjsip_logger.c: <--- Received SIP request (1789 bytes) from UDP:94.79.167.202:1195 --->
INVITE sip:[email protected]:5060;transport=udp SIP/2.0^M
Via: SIP/2.0/UDP 94.79.167.202:1195;rport;branch=z9hG4bKPjJP4GUACFt5zL06rIZ2t14z1hH-0v8pAX^M
Max-Forwards: 70^M
From: <sip:[email protected]>;tag=jpJwk463CgNO0p0nxUs5ZHIxObG2uDaF^M
To: sip:[email protected]^M
Contact: <sip:[email protected]:1195;ob>^M
Call-ID: I-miUtIoUWDCDLOOsiAse5tCPlGg3B-f^M
CSeq: 31890 INVITE^M
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS^M
Supported: replaces, 100rel, timer, norefersub^M
Session-Expires: 1800^M
Min-SE: 90^M
User-Agent: Sangoma P320 4_24_2 000FD3D220A3^M
Accept: application/sdp, application/X-Digium-Event, text/*^M
Authorization: Digest username="11", realm="asterisk", nonce="1731947520/9bda3eb9fb1af3a2936e6a23f7e608db", uri="sip:[email protected]:5060;transport=udp", response="dbe0642e0d3005849a94781bf4f527ab", algorithm=MD5, cnonce="zu9olY9VIyOmzbgPplKrJvc56HjSW0", opaque="7ef96d543b174711", qop=auth, nc=00000001^M
Content-Type: application/sdp^M
Content-Length: 753^M
when it doesn’t there is just this:
[2024-11-18 17:39:02] VERBOSE[2141] res_pjsip_logger.c: <--- Received SIP request (852 bytes) from UDP:94.79.167.202:1195 --->
REGISTER sip:[email protected]:5060;transport=udp SIP/2.0^M
Via: SIP/2.0/UDP 94.79.167.202:1195;rport;branch=z9hG4bKPjPv9DgkOs1a-WQeeUgnBjQ2Pjua2jH7yT^M
Max-Forwards: 70^M
From: <sip:[email protected]>;tag=y7ZIESLQ1CkyOdkuVnqXtPzfqHH9mSOG^M
To: <sip:[email protected]>^M
Call-ID: T0ye83-HEjPwmbSpEK71U2GBa83LwoDR^M
CSeq: 8515 REGISTER^M
User-Agent: Sangoma P320 4_24_2 000FD3D220A3^M
Contact: <sip:[email protected]:1195;ob>^M
Expires: 300^M
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS^M
Authorization: Digest username="11", realm="asterisk", nonce="1731947942/50a49808f28bc494ec31f1a30de2ca40", uri="sip:[email protected]:5060;transport=udp", response="a4bd8b12c02c781b94f91c029b06a23e", algorithm=MD5, cnonce="jyeRl8Szd89bLo655r3qi5pzeo0GBS13", opaque="0609d891736e193b", qop=auth, nc=00000001^M
Content-Length: 0
Not being very familiar with sip traffic I am guessing the latter is a register event where “from” and “to” are the same extension whereas the former is an “invite” that establishes a call. Those happen about every thenth try.
I can’t find a pattern in the behaviour. Tried different extensions and different P320s, one of them brandnew. P320s most of the times just don’t dial out anymore with no trace in the logs. All other phones do that, even on the same extensions. I am out of ideas.