hi All I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb ram, with g729 for i686 , (fedora 1).
my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen otherparty realtime voice , but other party geting sip party's voice 1 sec later (not realtime). please some help me to solve this issu, last one month i am tring different different way to solve this issu. is it codec problem or something else. thanks bashir ----- Original Message ----- From: "Aarthy G - CTD, Chennai." <[EMAIL PROTECTED]> To: <[email protected]> Sent: Wednesday, July 27, 2005 1:12 AM Subject: [Asterisk-Users] Regarding Call Hold > > Hi All, > > > > We are using asterisk for testing our home gateway setup. > > We have implemented Call Hold feature in our application. > > In our Application we have written code in such a way that for an INVITE > > for > > putting a SIP phone on HOLD will contain connection address "0.0.0.0" in > > the SDP message. > > We expect the same connection address i.e "0.0.0.0" in the 200 OK response > > for the INVITE that is sent. > > This feature works when we tested without involving Asterisk. > > Now after configuring Asterisk as our Registrar and OutBound Proxy, we > > find that Call hold is not getting through. But we are getting a 200 0K > > with connection address as the host ip of Asterisk. We see that the this > > ReInvite is not getting forwarded to the appropriate detsination from the > > asterisk. We are not looking for music on hold feature. > > Output of sip debug and the two configuration files sip.conf and > > extensions.conf > > have been attached in this mail. > > Lines where we send "0.0.0.0" in the connection address field of SDP > > message and the 200 OK Response in which we get host ip of Asterisk in > > connection Address have > > been highlighted in RED in the attached word document. > > Please go through the configuration files and the debug output and suggest > > us the necessary changes that have to be done by us. > > We also do not want music_on_hold feature. > > Can somebody here please tell us about how to configure asterisk to > > disable music on hold > > and get 0.0.0.0 in the 200 OK response for the Re-Invite Sent? > > > > thanks, > > Aarthy G. > > <<Call-Hold.zip>> > > DISCLAIMER > > This message and any attachment(s) contained here are information that is > > confidential, proprietary to HCL Technologies and its customers. Contents > > may be privileged or otherwise protected by law. The information is solely > > intended for the individual or the entity it is addressed to. If you are > > not the intended recipient of this message, you are not authorized to > > read, forward, print, retain, copy or disseminate this message or any part > > of it. If you have received this e-mail in error, please notify the sender > > immediately by return e-mail and delete it from your computer. > > > > > ---------------------------------------------------------------------------- ---- > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
