asterisk users - Jul 2005 - 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. 
> 
> 
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Please file a bug report with a full SIP DEBUG output file. Set debug to
4, verbosity to 4 and turn on SIP debugging. Upload that file as an
attachment to the bug report and place the bug report in the SIP category.

Thanks!

/O

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@hcltech.com>
To: <asterisk-users@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>>
> > 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.
> >
> >
>

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