[sip-comm] Re: Hi Emcho


#1

Hello Florian,

There seem to be many problems with the sip-commmunicator when connecting to asterisk currently. We'll soon have a local copy installed here and will do some testing. Until then - any fixes are welcome

Cheers
Emil

P.S. Please send this kind of questions to the sip-communicator mailing lists.

Florian Buzin wrote:

···

Hi Emcho,

i am trying your sip communicator for several hours now, but i cannot find the correct setting to run it with asterisk.

I if i use the direct sound device i get one way sound after about 30 secs. but the sound is really bad and the asterisk console shows
dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833

do you have any advice for me?

Best regards
Florian Buzin

---------------------------------------------------------------------
To unsubscribe, e-mail: users-unsubscribe@sip-communicator.dev.java.net
For additional commands, e-mail: users-help@sip-communicator.dev.java.net