HELP!! Voice Connection During A SIP Call

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



On Tue, Jul 29, 2008 at 1:47 AM, Joseph Maiquez <josephmaiquez at yahoo.com>wrote:

> My problem was no voice was transmitted during a phone call that's why I
> can't hear the voice of the person on the other network. I think someone had
> faced this problem before
>
> here is the link..
>
>
>
> http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2008-June/003452.html
>
>
I think that discussed different problem.


>
> Im sorry but i dont have any idea on how to transmit voice.
>


> This is what I did.
>
>
My guess is probably you're missing the call to pjsua_conf_connect().



> Call the pjsua_call_make_call function to begin the invite Sip Call
>
> Then on the other end, call the function pjsua_call_answer then the call is
> successfully established but no audio is enable. How can I enable the audio
> during the call? this is what I wanted to know.. Hope you can help me guys,
> a sample code on how to make it will be very useful..
>
>
But we already have the samples (simple_pjsua, etc.). What's wrong with
them?

 -benny
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080729/01f1243d/attachment.html 


[Index of Archives]     [Asterisk Users]     [Asterisk App Development]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [Linux API]
  Powered by Linux