Re: [Asterisk-Users] 403 FORBIDDEN Help!

[Bartosz Jozwiak]

Some more details.... When I am dailing an extension on Asterixsk PBX     Maybe it will help some how....     66.178.36.15 -> 66.178.36.220   SIP/2.0 66.178.36.15 -> 66.178.36.220   SIP/2.0/UDP 66.178.36.15:5060;branch=2b0b0ed72ad04c5615dcab707e0fbe4a.4 66.178.36.15 -> 66.178.36.220   SIP/2.0/UDP 66.178.36.15:5060;branch=49cec9c54d88703154f83cb6acc7b397.2 66.178.36.15 -> 66.178.36.220   SIP/2.0/UDP 66.178.37.103:5060 :V:#.Af.8.6."0.P`> f.8.6..P`:  udp i4 (DF) E@@iB$B$qINVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137 From: "[EMAIL PROTECTED]" <sip:[EMAIL PROTECTED]>;tag=as4046c5f2 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 238   v=0 o=root 18960 18960 IN IP4 66.178.36.220 s=session c=IN IP4 66.178.36.220 t=0 0 m=audio 15100 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 66.178.36.220 -> 66.178.36.15   SIP/2.0 66.178.36.220 -> 66.178.36.15   SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137 :V:$."Qf.8.6."0.P`> f.8.6..P`:  udp i4 (DF) E@@iB$B$qINVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137 From: "[EMAIL PROTECTED]" <sip:[EMAIL PROTECTED]>;tag=as4046c5f2 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 238   v=0 o=root 18960 18960 IN IP4 66.178.36.220 s=session c=IN IP4 66.178.36.220 t=0 0 m=audio 15100 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 66.178.36.220 -> 66.178.36.15   SIP/2.0 66.178.36.220 -> 66.178.36.15   SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137 :V:%.wPIf.8.6..P`> f.8.6."0.P`:  udp 02 (DF) EJ@@kTB$B$6SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137 To: <sip:[EMAIL PROTECTED]>;tag=2ff28aef From: "[EMAIL PROTECTED]" <sip:[EMAIL PROTECTED]>;tag=as4046c5f2 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0   66.178.36.15 -> 66.178.36.220   SIP/2.0 403 Forbidden 66.178.36.15 -> 66.178.36.220   SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137 :V:%.wiCf.8.6."0.P`> f.8.6..P`:  udp 81 (DF) E@@kB$B$vACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137 From: "[EMAIL PROTECTED]" <sip:[EMAIL PROTECTED]>;tag=as4046c5f2 To: <sip:[EMAIL PROTECTED]>;tag=2ff28aef Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0   ----- Original Message ----- From: Bartosz Jozwiak To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 11:50 AM Subject: Re: [Asterisk-Users] 403 FORBIDDEN Help!   Asterix PBX is loggin to Vocal and the extension number is also loggin on the same vocal server. I cannot make it work.... :(   ----- Original Message ----- From: Josh Roberson To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 11:43 AM Subject: Re: [Asterisk-Users] 403 FORBIDDEN Help! I'm new too, but alot of my 403 forbidden messages when adding extensions were due to context rules..   make sure that the client dialing the extension is included in the same context your extension is in.    just my thoughts on it, as it resolved a lot of 403 errors for me.     ----- Original Message ----- From: Bartosz Jozwiak To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 9:31 AM Subject: [Asterisk-Users] 403 FORBIDDEN Help! Hello,   I have a question. I set up an extension to 1234   exten => 1234,1,Dial(SIP/[EMAIL PROTECTED]:5060)   And when I dial that extension I got in SIP message "403 FORBIDDEN" Can somebody tell me why I cannot call that extension? When I am not using Asterisk I can call that extension without any problems. My SIP proxy is VOCAL. I am new here so I do not know a lot yet.   Thank you in advance.   Bartosz Jozwiak