[Asterisk-Users] Regarding Call Hold

[Aarthy G - CTD, Chennai.]

> 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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