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@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.
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).
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