[Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6

[Vice President - Lamsre]

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." <aarthyg at hcltech.com>
To: <asterisk-users at lists.digium.com>
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>>
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> >
>


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