24 Jul
2005
24 Jul
'05
10:37 a.m.
Aarthy G - CTD, Chennai. wrote:
Without seeing a SIP debug output it's very hard to diagnose your
problem. Surely Asterisk supports on hold by putting 0.0.0.0 in the SDP,
but it seems like your system is doing something else.
/O
Hi,
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.
>Can somebody here please tell us about how to configure asterisk for this
>to work?
>