On 2007-01-19, Craig Matsuura <[EMAIL PROTECTED]> wrote:
> Seems the 1.2.0 version was working better than the 1.4.0 - 1.5.1 version of 
> linphone.  
I think I used some of this versions and it worked well.

>
> When using asterisk as the exchange, the calls appear to route but now
> sound.  
Try to switch re-INVITE functionality on Asterisk (canreinvite=no) and
on linphone to select alaw or ulaw format. I had problem with speex, becasue
Asterisk tried to convert the stream and it was a lot of bad.

-- Petr


> On Thursday 18 January 2007 1:01 pm, Craig Matsuura wrote:
>> I looking for some of the basic setup for linphonec to connect (use) a
>> local asterisk server.
>>
>> My understanding is I have to setup a proxy server (proxy add) to my
>> asterisk server?  I'm unsure if I am doing it correctly as I can not call
>> the other sip phones in my network.  I had success with minisip, but prefer
>> to use linphone as it is much smaller.
>>
>> I can dial directly from one linphone to another, but rather go via the
>> asterisk server.
>>
>> Examples of a working configuration of asterisk and linphonec would be
>> greatly appreciated, as well as usage (assuming I using linphonec
>> incorrectly).
>>
>> Thanks,
>> Craig
>>
>>
>> _______________________________________________
>> Linphone-users mailing list
>> Linphone-users@nongnu.org
>> http://lists.nongnu.org/mailman/listinfo/linphone-users



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