Re: [Linphone-users] connecting linphone to asterisk

[Mariusz Bożewicz]

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?

-- 
regards
Mariusz Bozewicz