Wrong ptime on single trunk

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:[email protected]:5060 SIP/2.0
            PBX:5060             TRUNK:5060 │Via: SIP/2.0/UDP PBX:5060;rport;branch=z9hG4bKPj3d56d099-1649-4242-a90b-d31c0cacd4bd
          ──────────┬─────────          ──────────┬─────────│From: <sip:[email protected]>;tag=d83a3ab4-f41a-4679-b59a-55a2f0970178
                    │        INVITE (SDP)         │         │To: <sip:[email protected]>
  14:51:35.815154   │ ──────────────────────────> │         │Contact: <sip:[email protected]: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

Just to let you know, we’ve changed the trunk provider (the trunk is a voice gateway, we’ve replaced the SIM inside) and the problems are gone.

Now my question is: how do you get if problem is generated by remote provider or local gateway or your phone device?

Thanks, BR

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