Re: [Linphone-users] connecting linphone to asterisk

[Simon MORLAT]

You should enable linphone to use all codecs by setting connection type 
to "Ethernet or Adsl".
See the informationnal message at the left of the codec config box that 
nearly says "Codecs in red are not usable with respect of your internet 
connection.".
Simon

Rajesh Narayanan wrote:

> 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?
> >
> >  
> ------------------------------------------------------------------------
>
> _______________________________________________
> Linphone-users mailing list
> address@hidden
> http://lists.nongnu.org/mailman/listinfo/linphone-users
>