hi,
i have registered with Asterisk, After that when i try to call my other
extension, i am getting an "403 Forbidden" response.
Below is the dump of SIP messages.
Is there any SIP message is in conflict with the Asterisk rules..?
I need your help....
>>
SENT:
SOURCE [0.0.0.0:1035] <-> DESTINATION [10.100.12.201:5060]
INVITE sip:6442161001@10.100.12.201:5060 SIP/2.0
Via: SIP/2.0/UDP
10.100.12.209:5060;rport;branch=z9hG4bKce95dd81f969e71b577c8037f9bbdd63
From:
"Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4
To: sip:6442161001@10.100.12.201
Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209
CSeq: 24891 INVITE
Contact: "Phone1"<sip:6442091001@10.100.12.209>
Content-Length: 287
Content-Type: application/sdp
Expires: 180
Max-Forwards: 70
Organization: PMC-SIERRA
Proxy-Authorization: DIGEST
username="6442091001",realm="asterisk",nonce="51afa269",uri="sip:6442161001@
10.100.12.201",response="4e3255ca53d7cd894c98fac21132a991",algorithm=MD5
User-Agent: Stein
Authorization: DIGEST
username="6442091001",realm="asterisk",nonce="51afa269",uri="sip:6442161001@
10.100.12.201",response="4e3255ca53d7cd894c98fac21132a991",algorithm=MD5
v=0
o=UserName 584103545 0 IN IP4 10.100.12.209
s=Voice Session
c=IN IP4 10.100.12.209
t=0 0
m=audio 8002 RTP/AVP 0 4 8 18
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=sendrecv
a=rtcp: 8003
<<
RECEIVED:
SOURCE [0.0.0.0:1036] <-> DESTINATION [10.100.12.201:5060]
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.100.12.209:5060;branch=z9hG4bK39757cbf1864977ffda3e87bf0072942
From:
"Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4
To: sip:6442161001@10.100.12.201;tag=as7f9a42c9
Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209
CSeq: 24892 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:6442161001@10.100.12.201>
Content-Length: 0
<<
RECEIVED:
SOURCE [0.0.0.0:1036] <-> DESTINATION [10.100.12.201:5060]
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
10.100.12.209:5060;branch=z9hG4bK39757cbf1864977ffda3e87bf0072942
From:
"Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4
To: sip:6442161001@10.100.12.201;tag=as7f9a42c9
Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209
CSeq: 24892 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:6442161001@10.100.12.201>
Content-Length: 0
<<
RECEIVED:
SOURCE [0.0.0.0:1036] <-> DESTINATION [10.100.12.201:5060]
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
10.100.12.209:5060;branch=z9hG4bK39757cbf1864977ffda3e87bf0072942
From:
"Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4
To: sip:6442161001@10.100.12.201;tag=as7f9a42c9
Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209
CSeq: 24892 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:6442161001@10.100.12.201>
Content-Length: 0
>>
SENT:
SOURCE [0.0.0.0:1036] <-> DESTINATION [10.100.12.201:5060]
ACK sip:6442161001@10.100.12.201:5060 SIP/2.0
Via: SIP/2.0/UDP
10.100.12.209:5060;rport;branch=z9hG4bK39757cbf1864977ffda3e87bf0072942
From:
"Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4
To: sip:6442161001@10.100.12.201;tag=as7f9a42c9
Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209
CSeq: 24892 ACK
Content-Length: 0
Organization: PMC-SIERRA
User-Agent: Stein
Thanks,
Subashini
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