On 05/12/14 16:46, Olli Heiskanen wrote:
INVITE that Asterisk (at port 5070) receives:
PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046
INVITE sip:6...@testers.com
<mailto:sip%3a...@testers.com>;transport=UDP SIP/2.0
Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177>
Via: SIP/2.0/UDP
PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0
Via: SIP/2.0/UDP
AST.ER.ISK.IP:38699;rport=38699;branch=z9hG4bK-d8754z-bd00e9fd46368417-1---d8754z-
Max-Forwards: 69
Contact: <sip:7...@ast.er.isk.ip:38699;transport=UDP>
To: <sip:6...@testers.com <mailto:sip%3a...@testers.com>;transport=UDP>
From: "771"<sip:7...@testers.com
<mailto:sip%3a...@testers.com>;transport=UDP>;tag=41030177
Call-ID: YWYwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Allow-Events: presence, kpml
Content-Length: 239
v=0
o=Z 0 0 IN IP4 AST.ER.ISK.IP
s=Z
c=IN IP4 AST.ER.ISK.IP
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
This client is saying it only supports speex and iLBC and would prefer
them in that order.
Your sip.conf appears to only permit alaw, ulaw and gsm so there is no
mutual supported codec and hence the call fails.
--
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