On Fri, 2 Jul 2004 12:45:07 +0100
address@hidden wrote:
Command ? | INFO1 | <udp.c: 295> Sending message:
INVITE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK1765101061
From: <sip:address@hidden>;tag=2495366955
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 20 INVITE
Contact: <sip:address@hidden>
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Type: application/sdp
Content-Length: 242
v=0
o=aa 123456 654321 IN IP4 192.168.10.24
s=A conversation
c=IN IP4 192.168.10.24
t=0 0
m=audio 7078 RTP/AVP 110 115 101
b=AS:8
a=rtpmap:110 speex/8000/1
a=rtpmap:115 1015/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
Above there is Your invite message.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK1765101061
From: <sip:address@hidden>;tag=2495366955
To: <sip:address@hidden>;tag=as3b81e5d4
Call-ID: address@hidden
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:address@hidden>
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 26656 26656 IN IP4 192.168.10.20
s=session
c=IN IP4 192.168.10.20
t=0 0
m=audio 13906 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
Above there is response message from remote machine.
The messages consist of two
parts: SIP body and SDP Body.
Compare SDP bodies from above messages. Take a look on lines "a=". As
you can see "101 telephone-event/8000" exists in those messages.
Linphone trys tu use exactly that codec.
Connected.
MediaStreamer-Message: alsa_set_params: blocksize=512.
Sound was initialized correctly.
MediaStreamer-ERROR **: mediastream.c: No decoder availlable for
payload 101. aborting...
Aborted
There is no needed decoder.
[general]
port=5060 ; Port to bind to
bindaddr=192.168.10.20 ; Address to bind SIP channel to
context=default ; Default context for incoming calls
;srvlookup = yes ; Enable DNS SRV lookups on outbound
calls ;pedantic = yes ; Enable slow, pedantic checking
for Pingtel ;tos=lowdelay ; IP QoS parameter, either
keyword or value
;maxexpirey=3600 ; Max length of incoming registration
we allow ;defaultexpirey=120 ; Default length of
incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in
NOTIFY ;videosupport=yes ; Turn on support for SIP video
;disallow=all ; Disallow all codecs
;allow=gsm
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc
Maybe You should uncomment gsm and ulaw codec?