I am glad to tell you that I have found the solution. the problem is : A is pjusa, B is another sip phone. A is talking with B, B press hold key, then A start to listen to hold music. After a while, B press unhold key, music stop. Then B can hear A's voice, but A can not hear B's voice. The solution: A also press unhold key( input "v"), then everything is ok. I don't know where the exactly problem is? The Server is asterisk, and set canreinvite=no ?2008-04-04?ofn at keystream.se ??? In the re-invite and corresponding OK there should be a sendrecv attribute for media just check if it's present. Citerar hlzhangxt at 163.com: > The server is asterisk 1.4 providing music on hold. > > You mean that is sip signal problem ? > > How to restart codec ? > > > > > ?2008-04-03?"Olle Frimanson" <olle.frimanson at keystream.se> ??? > > > Hi we use the commercial version of VA G729 and run it on embedded > Linux, and we don?t have this problem. Is A and B connected to a > server or who provides music on hold? > > > > Have you tested the same scenario with another codec? > > Check the exact signalling with wireshark are sendrcv, inactive, > sendonly, rcvonly correct in the re-invite? > > Perhaps the codec?s needs to be restarted after a hold/unhold. > > > > BR/Olle > > > > > > From: hlzhangxt [mailto:hlzhangxt@xxxxxxx] > Sent: den 3 april 2008 14:31 > To:olle.frimanson at keystream.se > Subject: a problem with voiceage g729 > > > > > > > > > > I have successfully integrated voiceage g729 with pjsip. > > But, there is a problem. > > > > A is pjusa, B is another sip phone. > > A is talking with B, B press hold key, then A start to listen to hold music. > > After a while, B press unhold key, music stop. > > Then B can hear A's voice, but A can not hear B's voice. > > > > What's the problem? > > > > > > ? ? ? ? ?? --- ? ? ? ? ? ? ?166 ? ? ? ? ? ? ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080404/7ec24445/attachment.html