Lee, John (Sydney)
2011-Sep-14 06:56 UTC
[asterisk-users] 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? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110914/3b88db26/attachment.htm>
Lee, John (Sydney)
2011-Sep-14 07:23 UTC
[asterisk-users] 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? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110914/85944d0c/attachment.htm>
Lee, John (Sydney)
2011-Sep-14 07:37 UTC
[asterisk-users] 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? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110914/10531d66/attachment.htm>
Lee, John (Sydney)
2011-Sep-15 01:04 UTC
[asterisk-users] secret=pw in sip.conf affecting inter-asterisk sip call
> chan_sip does not support specification of the password to be used forauthentication in the dial string itself;> your ":password" suffix is just being sent to the target system and itis 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110915/a54725a1/attachment.htm>