Singaling calls during sip round

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

 



Hi all,

 I use your software (version 5) and I have a problem: when I make a
call from a simple Voip telephone to my agent (pjsua)  I don't have any
signalling tone in my phone  so I can't understand if it's online or
busy .... I look into the documentations on web, but I can't find any
solutions!  I use the -play-tone options, but I can connect  a call to
this port only after that the Sip round is terminated! And I wont that
the calling tone is play during the Sip round ! 

 

Can you help me?

 

Thank you

 

Fabio

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20070924/eef07656/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