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

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Hey,

The callee server is complaining too loud “Call from ‘2765’ to extension ‘*
1166:password*’ rejected because *extension not found*.”
Try changing the Dial string as DIAL(SIP/asterisk-callee/${EXTEN}) or w/e
extension you require in place of ${EXTEN}
Let me know what changes.

Also this is a good read:
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/

Wed, Sep 14, 2011 at 12:37 PM, 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@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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