Re: [Linphone-users] connecting linphone to asterisk

[Rajesh Narayanan]

Hi,

I had the same problem trying to connect two linphones together.
The version is 0.12.2 . I am not sure how to comment out the GSM
and PCMU codec. When I tried to remove it from the config file
it  gave me the following message:

Message:Found /dev/dsp.
Message:Found ALSA device: Intel 82801CA-ICH3
| INFO1 | <osipmanager.c: 148> port already listened

| INFO1 | <osipmanager.c: 148> port already listened

Message:Adding new codec PCMU/8000
Message:Adding new codec GSM/8000
Ready.

Not sure whats happening. I am unable to have two linphones talk to each
other. Have been debugging 0.12.2 on my system for a while. I am running
Suse 8.1.

Thanks,
Rajesh.

Mariusz Bożewicz wrote:

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?