asterisk users - Sep 2011 - secret=pw in sip.conf affecting inter-asterisk sip call

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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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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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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> chan_sip does not support specification of the password to be used for
authentication in the dial string itself; 
> your ":password" suffix is just being sent to the target system
and it
is trying to find a matching extension in the dialplan (and failing).

Thanks Kevin.  This is what I reckon from the tests that I did.  
I think I will have to remove all secret= from all my SIP entries.

However, this is contrary to what the Asterisk books say.

P.S.  I have got problem receiving emails from asterisk-user mailing
list.
I could only see it from the web mail archive.
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