I was trying to do a SIP call between two Asterisk servers (1.4.21.2)
1) On the caller server, I coded the following in extensions.conf
Dial(SIP/1166:password@asterisk-callee);
2) On the callee server, I coded the following in sip.conf
[1166]
type=friend ; Friends place calls and receive calls
context=incoming ; Context for incoming calls from this
user
host=dynamic ; This peer register with us
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
qualify=yes ; Monitor latency between Asterisk server
and phone
call-limit=99
username=1166 ; Username to use in INVITE until peer
registers
secret=password ; Normally you do NOT need to set this
parameter
mailbox=1166@default ; mailbox 5100 in voicemail context
.default.
callgroup=1
pickupgroup=1
The call was unsuccessful as follows.
1) On the caller machine, this is what we got from the console
-- Executing [1166@incoming:1] Dial("SIP/1166-09d81668",
"SIP/1166:password@asterisk-callee") in new stack
-- Called 1166:password@asterisk-callee
-- SIP/asterisk-callee is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
2) On the callee machine, this is what we got from the console,
[Sep 14 14:34:12] NOTICE[11991]: chan_sip.c:14035 handle_request_invite:
Call from '2765' to extension '1166:password' rejected because extension
not found.
However, I found out that if I remove "secret=.." from the SIP entry and
call without the password, then I will be able to call.
Any thoughts?
--
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