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:1002@sip.greentone.com SIP/2.0
Via: SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137
From: "1002@sip.greentone.com"
<sip:asterisk@66.178.36.220>;tag=as4046c5f2
To: <sip:1002@sip.greentone.com>
Contact: <sip:asterisk@66.178.36.220>
Call-ID: 0d94bac54d736a42709e590a6dee22d8@66.178.36.220
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$qINVITE sip:1002@sip.greentone.com SIP/2.0
Via: SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137
From: "1002@sip.greentone.com"
<sip:asterisk@66.178.36.220>;tag=as4046c5f2
To: <sip:1002@sip.greentone.com>
Contact: <sip:asterisk@66.178.36.220>
Call-ID: 0d94bac54d736a42709e590a6dee22d8@66.178.36.220
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 18960 18960 IN IP4 66.178.36.220
s=session
c=IN IP4 66.178.36.220
t=0 0
m=audio 15100 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
66.178.36.220 -> 66.178.36.15 SIP/2.0
66.178.36.220 -> 66.178.36.15 SIP/2.0/UDP
66.178.36.220:5060;branch=z9hG4bK44889137
:V:%.wPIf.8.6..P`> f.8.6."0.P`: udp 02 (DF)
EJ@@kTB$B$6SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137
To: <sip:1002@sip.greentone.com>;tag=2ff28aef
From: "1002@sip.greentone.com"
<sip:asterisk@66.178.36.220>;tag=as4046c5f2
Call-ID: 0d94bac54d736a42709e590a6dee22d8@66.178.36.220
CSeq: 102 INVITE
Content-Length: 0
66.178.36.15 -> 66.178.36.220 SIP/2.0 403 Forbidden
66.178.36.15 -> 66.178.36.220 SIP/2.0/UDP
66.178.36.220:5060;branch=z9hG4bK44889137
:V:%.wiCf.8.6."0.P`> f.8.6..P`: udp 81 (DF)
E@@kB$B$vACK sip:1002@sip.greentone.com SIP/2.0
Via: SIP/2.0/UDP 66.178.36.220:5060;branch=z9hG4bK44889137
From: "1002@sip.greentone.com"
<sip:asterisk@66.178.36.220>;tag=as4046c5f2
To: <sip:1002@sip.greentone.com>;tag=2ff28aef
Contact: <sip:asterisk@66.178.36.220>
Call-ID: 0d94bac54d736a42709e590a6dee22d8@66.178.36.220
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
----- Original Message -----
From: Bartosz Jozwiak
To: asterisk-users@lists.digium.com
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: asterisk-users@lists.digium.com
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: Asterisk-Users@lists.digium.com
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/1234@sip.greentone.com: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
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