[asterisk-users] secret=pw in sip.conf affecting inter-asterisk sip call

[Lee, John (Sydney)]

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 at 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 at 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 at incoming:1] Dial("SIP/1166-09d81668",
"SIP/1166:password at asterisk-callee") in new stack

    -- Called 1166:password at 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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