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

[Kevin P. Fleming]

On 09/14/2011 02:37 AM, Lee, John (Sydney) wrote:
> 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.

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).

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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