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