<div dir="ltr"><div><div>It's a configuration issue with the peer (or the peer it's matching) in sip.conf<br><br></div>You have configured the peer to use avpf but your phone is not attempting to do avpf.<br><br></div>Either configure the phone to match the entry in sip.conf or change sip.conf to match the phone.<br><br></div><br><div class="gmail_quote"><div dir="ltr">On Mon, Aug 27, 2018 at 4:56 PM <<a href="mailto:asterisk@a-domani.nl">asterisk@a-domani.nl</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Retrying, falling of the list some how :-(<br>
<br>
<br>
-------- Original Message --------<br>
Subject: feeling n00b again<br>
Date: 2018-08-20 09:51<br>
From: <a href="mailto:asterisk@a-domani.nl" target="_blank">asterisk@a-domani.nl</a><br>
To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>
<br>
Hi all,<br>
<br>
Long time ago, I followed a Asterisk training, and both at work and at <br>
home, was able to deploy Asterisk,<br>
make all sorts of internal call (hard/soft voip-phones, <br>
incoming/outgoing, ISDN-BA/PRA, SIP, IAX2, ata's, realtime, voicemail, <br>
load-balancing crypto; almost anything.<br>
But the days of the 1.2-release is long gone.<br>
<br>
Now, I'm trying to pick it up again, but even the most simple config <br>
seems to fail.<br>
Hardly anything seems to works.<br>
<br>
<br>
Situation:<br>
Asterisk 15.2.2 on a x86_64 running Linux on 2018-03-06 15:11:19 UTC<br>
phone1: Grandsteam2000<br>
phone2: Siemens DECT<br>
phone are all on same subnet (no nat involved here)<br>
<br>
1) Echo functionality on phone 1: OK (using alaw codec)<br>
2) Echo functionality on phone 2: OK (using alaw codec)<br>
3) Call from phone2 to phone1: OK (both using alaw)<br>
4) Call from phone1 to phone2: immediate disconnect after answering<br>
(might not be related) console says:<br>
<br>
[Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10434 <br>
process_sdp: Received AVP profile in audio answer but AVPF is enabled: <br>
audio 7200 RTP/AVP 8 101<br>
[Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10819 <br>
process_sdp: Failing due to no acceptable offer found<br>
<br>
I enabled debug on the IP of the dect-phone (full log attached), but it <br>
does not make me any wiser...<br>
<br>
<br>
set_destination: Parsing <<a href="http://sip:dect@192.168.0.27:5060" rel="noreferrer" target="_blank">sip:dect@192.168.0.27:5060</a>> for address/port <br>
to send to<br>
set_destination: set destination to <a href="http://192.168.0.27:5060" rel="noreferrer" target="_blank">192.168.0.27:5060</a><br>
Reliably Transmitting (no NAT) to <a href="http://192.168.0.27:5060" rel="noreferrer" target="_blank">192.168.0.27:5060</a>:<br>
BYE <a href="http://sip:dect@192.168.0.27:5060" rel="noreferrer" target="_blank">sip:dect@192.168.0.27:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK239cc5d8<br>
Max-Forwards: 70<br>
From: "fam-witvliet eerste verdiep" <br>
<<a href="mailto:sip%3Avoip4-01@192.168.0.25" target="_blank">sip:voip4-01@192.168.0.25</a>>;tag=as112dbb55<br>
To: <<a href="http://sip:dect@192.168.0.27:5060" rel="noreferrer" target="_blank">sip:dect@192.168.0.27:5060</a>>;tag=1813732733<br>
Call-ID: <a href="http://78d92db820b4926879361f7d4968444a@192.168.0.25:5060" rel="noreferrer" target="_blank">78d92db820b4926879361f7d4968444a@192.168.0.25:5060</a><br>
CSeq: 103 BYE<br>
User-Agent: Asterisk PBX 15.2.2<br>
===> X-Asterisk-HangupCause: Bearer capability not available <===<br>
===> X-Asterisk-HangupCauseCode: 58 <===<br>
Content-Length: 0<br>
<br>
<br>
<br>
Anyone around to give some pointers/clues?-- <br>
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