Hi,
FREEPBX 16 with Asterisk 18.6.0
PJSIP extension act in different way based on the device.
On a Fanvill x7 I’ve set ptime=40:
- from-internal calls works fine.
- when using a trunk it turn out to use a=ptime:20, the trunk device is set to use 40ms.
On trunk page I cannot see anything like the allow row present in the extension to force : ulaw:40&g729&h264.
On the linphone device:
- from-internal calls works fine.
- when using the trunk device it use 40ms.
If I set ptime 40 on both devices, trunk and fanvil phone, the call is established with a ptime of 20.
If I set ptime 20 on both devices, trunk and fanvil phone, the call is established with a ptime of 20.
It seems that is the PBX that trigger the wrong ptime=20.
Anyway with a Linphone client it switch to the correct value, with fanvill no.
Can you please help me to make the trunk use the same ptime 40?
│INVITE sip:mobile@PBX:5060 SIP/2.0
PBX:5060 TRUNK:5060 │Via: SIP/2.0/UDP PBX:5060;rport;branch=z9hG4bKPj3d56d099-1649-4242-a90b-d31c0cacd4bd
──────────┬───────── ──────────┬─────────│From: <sip:trunknumber@PBX>;tag=d83a3ab4-f41a-4679-b59a-55a2f0970178
│ INVITE (SDP) │ │To: <sip:MOBILEDEST@TRUNK>
14:51:35.815154 │ ──────────────────────────> │ │Contact: <sip:asterisk@PBX:5060>
+0.065099 │ 100 Trying │ │Call-ID: 0da9956c-794f-44dc-abf0-078da30ed46d
14:51:35.880253 │ <────────────────────────── │ │CSeq: 31456 INVITE
+0.096263 │ 180 Ringing │ │Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK
14:51:35.976516 │ <────────────────────────── │ │REFER, MESSAGE
+10.088665 │ 200 OK (SDP) │ │Supported: 100rel, timer, replaces, norefersub, histinfo
14:51:46.065181 │ <────────────────────────── │ │Session-Expires: 1800
+0.000737 │ ACK │ │Min-SE: 90
14:51:46.065918 │ ──────────────────────────> │ │Max-Forwards: 70
+12.195917 │ BYE │ │User-Agent: FPBX-16.0.10.43(18.6.0)
14:51:58.261835 │ <────────────────────────── │ │Content-Type: application/sdp
+0.000448 │ 200 OK │ │Content-Length: 443
14:51:58.262283 │ ──────────────────────────> │ │
│ │ │v=0
│ │ │o=- 1725926302 1725926302 IN IP4 10.7.208.245
│ │ │s=Asterisk
│ │ │c=IN IP4 PBX
│ │ │t=0 0
│ │ │m=audio 12452 RTP/AVP 0 18 8 3 111 9 107 101
│ │ │a=rtpmap:0 PCMU/8000
│ │ │a=rtpmap:18 G729/8000
│ │ │a=fmtp:18 annexb=no
│ │ │a=rtpmap:8 PCMA/8000
│ │ │a=rtpmap:3 GSM/8000
│ │ │a=rtpmap:111 G726-32/8000
│ │ │a=rtpmap:9 G722/8000
│ │ │a=rtpmap:107 opus/48000/2
│ │ │a=fmtp:107 useinbandfec=1
│ │ │a=rtpmap:101 telephone-event/8000
│ │ │a=fmtp:101 0-16
│ │ │a=ptime:20
│ │ │a=maxptime:20
│ │ │a=sendrecv
│ │ │
│ │ │
How can I understand where is the ptime=20 setup in the PBX?
Thanks, BR