I have several branch offices, each with their own Asterisk server
(version 1.4.22.1) handling their PBX functions. All of these offices
need to talk to each other. In sip.conf I created a peer entry for each
office with a username of branch-user and a friend entry for every
branch-user with the username being just the branch, for example:
[Office2]
username=Office1-user
host=10.10.80.253
type=peer
context=internal
secret=sipptrunk
dtmfmode=rfc2833
disallow=all
allow=ulaw
call-limit=20
[Office1-user]
username=Office2
host=10.10.60.253
type=user
context=internal
secret=sipptrunk
dtmfmode=rfc2833
disallow=all
allow=ulaw
call-limit=20
This mostly works. I am able to place calls between all of the branches
about 90% of the time. What is happening is that if no call has been
placed between two of the branches for a while then the next call that
is made may get a "busy/congested" message like follows:
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/21-00a16c70",
"SIP/[EMAIL PROTECTED]||Tt") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/Office1-009f49b0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:2] Hangup("SIP/21-00a16c70", "") in new
stack
If I try a few more times, the call will eventually go through.
I have tried setting qualify=yes in sip.conf, and also setting up a
constant ping to make sure there were no connectivity issues between the
two sites. Any idea what could be causing the problem?
Thanks,
Brent Davidson
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
