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@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@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@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@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@10.100.12.209

CSeq: 24892 ACK

Content-Length: 0

Organization: PMC-SIERRA

User-Agent: Stein

Thanks,

Subashini

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