Seems like an ACK timeout to me. What does the log looks like?
On 07/01/2011 07:32 PM, Roman Gelfand wrote:
I am trying to take my configuration a step further with flexi route.
I removed all aliases from sipx and trying to route to different
extension based on sipToUser.
This is what I have done in my config... After approximately half a
minute media is not transmitted. I recall something like this in the
past and if I am not mistaken it had to do with authentication.
function is_my_pbx_routable(profile, ifaceAddr, ifacePort, ds, domain, ts)
{
if (typeof domain != "string")
return false;
//
// Check if the packet came in using the WAN interface
//
if ( ifaceAddr != sip_interface_address[0] ||
ifacePort != sip_interface_port[0])
return false;
//
// Check if the domain is valid
//
if (domain != "66.193.176.35" && domain != "mydomain.com
<http://mydomain.com>")
return false;
//
// route it to our PBX
//
profile.setTargetDomain("mydomain.com <http://mydomain.com>");
profile.setTargetAddress("udp", "sipx.mydomain.com
<http://sipx.mydomain.com>", "5060");
//
// Use the LAN interface to send it out
//
profile.setInterfaceAddress(sip_interface_address[1],
sip_interface_port[1]);
//
// Check if the source address is the ITSP. If yes, bridge it
//
if (profile.sipMessage.getSourceAddress() == "204.11.192.22")
if (ts == "1XXXXXXXXX1")
{
profile.bridge("user", "pass");
profile.sipMessage.setRequestUriUser("201");
}
else if (ts == "1XXXXXXXXXX2")
{
profile.bridge("user", "pass");
profile.sipMessage.setRequestUriUser("203");
}
return true;
}
Thanks in advance
_______________________________________________
sipx-users mailing list
[email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users/
