1. Brekeke Product Name and Version: Brekeke SIP Server, Version 3.3.8.1
2. Java version: 7 Update 17
3. OS type and the version: Windows 7 64 bit
4. UA (phone), gateway or other hardware/software involved: X-Lite soft phone
5. Your problem:
I'm trying to setup failover/redundant dial plans to my call manager & SRST router. I've been tinkering with the .net.sip.timeout.inviting & the .net.sip.timeout.ringing settings to no avail on my deploy patterns.
Is there a way to setup a dial plan (whether it's a single dial plan for outgoing with multiple TO servers, or multiple dial plans) so that if the first one does not respond to INVITE, that it will roll to the next outgoing dial plan or to the next TO in this dial plan?
Failover/Redundant Dialplan?
Moderator: Brekeke Support Team
Have you checked this page?
http://wiki.brekeke.com/wiki/How-to-mak ... -Dial-Plan
http://wiki.brekeke.com/wiki/How-to-mak ... -Dial-Plan
I have now.
Still no luck in SRST/failover. Current Deploy Pattern:
To = sip:%1@10.101.3.20
$session = failover sip:%2@10.101.2.250
&net.sip.timeout.inviting = 100
$auth = false
$transport = udp
the 10.110.2.250 does not see any requests coming in when I try dialing. Have set that timeout.inviting to 5, 20 100. I get pretty much immediate tri tone call failure. Delete the timeout.inviting setting to let it default, and it takes about a minute and then goes to tri tone call failure.
Have also tried Deploy Pattern (just reordered, not sure if sequence matters):
To = sip:%1@10.101.3.20
$auth = false
$transport = udp
$session = failover sip:%2@10.101.2.250
&net.sip.timeout.inviting = 100
Both Deploy patterns work with Call Manager online (10.101.3.20), but fail when in SRST (10.101.2.250 online, but not 10.101.3.20). I can place calls from Call Manager & SRST devices to the Brekeke server in either circumstance. Can only call from Brekeke to Call Manager when 10.101.3.20 is responding.
Still no luck in SRST/failover. Current Deploy Pattern:
To = sip:%1@10.101.3.20
$session = failover sip:%2@10.101.2.250
&net.sip.timeout.inviting = 100
$auth = false
$transport = udp
the 10.110.2.250 does not see any requests coming in when I try dialing. Have set that timeout.inviting to 5, 20 100. I get pretty much immediate tri tone call failure. Delete the timeout.inviting setting to let it default, and it takes about a minute and then goes to tri tone call failure.
Have also tried Deploy Pattern (just reordered, not sure if sequence matters):
To = sip:%1@10.101.3.20
$auth = false
$transport = udp
$session = failover sip:%2@10.101.2.250
&net.sip.timeout.inviting = 100
Both Deploy patterns work with Call Manager online (10.101.3.20), but fail when in SRST (10.101.2.250 online, but not 10.101.3.20). I can place calls from Call Manager & SRST devices to the Brekeke server in either circumstance. Can only call from Brekeke to Call Manager when 10.101.3.20 is responding.
I did try using what appears to be either old or fat fingered documentation from the link that says instead of
&net.sip.timeout.inviting = 30
to use
&failover.timer.inviting = 30
That seems to be an unrecognized variable, does not show as a green or blue as a valid variable usually does, and the phone takes about a minute to timeout on INVITE before the softphone does a tri tone call failure.
&net.sip.timeout.inviting = 30
to use
&failover.timer.inviting = 30
That seems to be an unrecognized variable, does not show as a green or blue as a valid variable usually does, and the phone takes about a minute to timeout on INVITE before the softphone does a tri tone call failure.