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On 05/12/14 16:46, Olli Heiskanen wrote:<br>
<blockquote
cite="mid:CALu7wuZt53W+MfQYus0D2_2+ZhKKdv5unRBpiGHQZPWgByrz0Q@mail.gmail.com"
type="cite">
<div>INVITE that Asterisk (at port 5070) receives:</div>
<div>PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046</div>
<div><span class="" style="white-space:pre"> </span>INVITE <a
moz-do-not-send="true" href="mailto:sip%3A660@testers.com">sip:660@testers.com</a>;transport=UDP
SIP/2.0</div>
<div><span class="" style="white-space:pre"> </span>Record-Route:
<sip:PU.BL.IC.IP;lr=on;ftag=41030177></div>
<div><span class="" style="white-space:pre"> </span>Via:
SIP/2.0/UDP
PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0</div>
<div><span class="" style="white-space:pre"> </span>Via:
SIP/2.0/UDP
AST.ER.ISK.IP:38699;rport=38699;branch=z9hG4bK-d8754z-bd00e9fd46368417-1---d8754z-</div>
<div><span class="" style="white-space:pre"> </span>Max-Forwards:
69</div>
<div><span class="" style="white-space:pre"> </span>Contact:
<a class="moz-txt-link-rfc2396E" href="mailto:sip:771@AST.ER.ISK.IP:38699;transport=UDP"><sip:771@AST.ER.ISK.IP:38699;transport=UDP></a></div>
<div><span class="" style="white-space:pre"> </span>To: <<a
moz-do-not-send="true" href="mailto:sip%3A660@testers.com">sip:660@testers.com</a>;transport=UDP></div>
<div><span class="" style="white-space:pre"> </span>From:
"771"<<a moz-do-not-send="true"
href="mailto:sip%3A771@testers.com">sip:771@testers.com</a>;transport=UDP>;tag=41030177</div>
<div><span class="" style="white-space:pre"> </span>Call-ID:
YWYwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk.</div>
<div><span class="" style="white-space:pre"> </span>CSeq: 2
INVITE</div>
<div><span class="" style="white-space:pre"> </span>Allow:
INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE</div>
<div><span class="" style="white-space:pre"> </span>Content-Type:
application/sdp</div>
<div><span class="" style="white-space:pre"> </span>Supported:
replaces, norefersub, extended-refer, timer, X-cisco-serviceuri</div>
<div><span class="" style="white-space:pre"> </span>User-Agent: Z
3.2.21357 r21367</div>
<div><span class="" style="white-space:pre"> </span>Allow-Events:
presence, kpml</div>
<div><span class="" style="white-space:pre"> </span>Content-Length:
239</div>
<div><br>
</div>
<div><span class="" style="white-space:pre"> </span>v=0</div>
<div><span class="" style="white-space:pre"> </span>o=Z 0 0 IN
IP4 AST.ER.ISK.IP</div>
<div><span class="" style="white-space:pre"> </span>s=Z</div>
<div><span class="" style="white-space:pre"> </span>c=IN IP4
AST.ER.ISK.IP</div>
<div><span class="" style="white-space:pre"> </span>t=0 0</div>
<div><span class="" style="white-space:pre"> </span>m=audio 8000
RTP/AVP 3 110 8 0 98 101</div>
<div><span class="" style="white-space:pre"> </span>a=rtpmap:110
speex/8000</div>
<div><span class="" style="white-space:pre"> </span>a=rtpmap:98
iLBC/8000</div>
<div><span class="" style="white-space:pre"> </span>a=fmtp:98
mode=20</div>
<div><span class="" style="white-space:pre"> </span>a=rtpmap:101
telephone-event/8000</div>
<div><span class="" style="white-space:pre"> </span>a=fmtp:101
0-15</div>
<div><span class="" style="white-space:pre"> </span>a=sendrecv</div>
</blockquote>
<br>
This client is saying it only supports speex and iLBC and would
prefer them in that order.<br>
Your sip.conf appears to only permit alaw, ulaw and gsm so there is
no mutual supported codec and hence the call fails.<br>
<br>
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