[asterisk-users] feeling n00b again

[John Kiniston]

It's a configuration issue with the peer (or the peer it's matching) in
sip.conf

You have configured the peer to use avpf but your phone is not attempting
to do avpf.

Either configure the phone to match the entry in sip.conf or change
sip.conf to match the phone.


On Mon, Aug 27, 2018 at 4:56 PM <asterisk at a-domani.nl> wrote:

> Retrying, falling of the list some how :-(
>
>
> -------- Original Message --------
> Subject: feeling n00b again
> Date: 2018-08-20 09:51
>  From: asterisk at a-domani.nl
> To: asterisk-users at lists.digium.com
>
> Hi all,
>
> Long time ago, I followed a Asterisk training, and both at work and at
> home, was able to deploy Asterisk,
> make all sorts of internal call (hard/soft voip-phones,
> incoming/outgoing, ISDN-BA/PRA, SIP, IAX2, ata's, realtime, voicemail,
> load-balancing crypto; almost anything.
> But the days of the 1.2-release is long gone.
>
> Now, I'm trying to pick it up again, but even the most simple config
> seems to fail.
> Hardly anything seems to works.
>
>
> Situation:
> Asterisk 15.2.2 on a x86_64 running Linux on 2018-03-06 15:11:19 UTC
> phone1: Grandsteam2000
> phone2: Siemens DECT
> phone are all on same subnet (no nat involved here)
>
> 1) Echo functionality on phone 1: OK (using alaw codec)
> 2) Echo functionality on phone 2: OK (using alaw codec)
> 3) Call from  phone2 to phone1: OK (both using alaw)
> 4) Call from phone1 to phone2: immediate disconnect after answering
> (might not be related) console says:
>
> [Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10434
> process_sdp: Received AVP profile in audio answer but AVPF is enabled:
> audio 7200 RTP/AVP 8 101
> [Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10819
> process_sdp: Failing due to no acceptable offer found
>
> I enabled debug on the IP of the dect-phone (full log attached), but it
> does not make me any wiser...
>
>
> set_destination: Parsing <sip:dect at 192.168.0.27:5060> for address/port
> to send to
> set_destination: set destination to 192.168.0.27:5060
> Reliably Transmitting (no NAT) to 192.168.0.27:5060:
> BYE sip:dect at 192.168.0.27:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK239cc5d8
> Max-Forwards: 70
>  From: "fam-witvliet eerste verdiep"
> <sip:voip4-01 at 192.168.0.25>;tag=as112dbb55
> To: <sip:dect at 192.168.0.27:5060>;tag=1813732733
> Call-ID: 78d92db820b4926879361f7d4968444a at 192.168.0.25:5060
> CSeq: 103 BYE
> User-Agent: Asterisk PBX 15.2.2
> ===> X-Asterisk-HangupCause: Bearer capability not available     <===
> ===> X-Asterisk-HangupCauseCode: 58                              <===
> Content-Length: 0
>
>
>
> Anyone around to give some pointers/clues?--
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