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 DISCLAIMER This message and any attachment(s) contained here are information that is confidential, proprietary to HCL Technologies and its customers. Contents may be privileged or otherwise protected by law. The information is solely intended for the individual or the entity it is addressed to. If you are not the intended recipient of this message, you are not authorized to read, forward, print, retain, copy or disseminate this message or any part of it. If you have received this e-mail in error, please notify the sender immediately by return e-mail and delete it from your computer
Hi Subashini,
unfortunately I don't thing is much we can help you here since the negative reply is generated by the Asterisk - maybe you should try to increase the debug level of Asterisk and look into the logs - maybe it will help you understand why you call is denied.
regards, bogdan
PS: I think your inquiry is not openser related at all, but more asterisk related ;)
Subashini C V - CTD, Chennai wrote:
hi,
i have registered with Asterisk, After that when i try to call myother 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
*DISCLAIMER: This message and any attachment(s) contained here are information that is confidential, proprietary to HCL Technologies and its customers. Contents may be privileged or otherwise protected by law. The information is solely intended for the individual or the entity it is addressed to. If you are not the intended recipient of this message, you are not authorized to read, forward, print, retain, copy or disseminate this message or any part of it. If you have received this e-mail in error, please notify the sender immediately by return e-mail and delete it from your computer *
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Bogdan-Andrei Iancu wrote:
Hi Subashini,
unfortunately I don't thing is much we can help you here since the negative reply is generated by the Asterisk - maybe you should try to increase the debug level of Asterisk and look into the logs - maybe it will help you understand why you call is denied.
Check the passwords on both sides.
/Olle Asterisk bugfixer :-)
Hi all,
I have an Uac wich register each time with a different port As a result he is registered 2 or 2 times in the location table with differents ports...
Ser seems to have a problem to manage that, impossible to call this Uac from another one And also seems to have quality problems...
Olivier
Hi!
You should drop this client :-)
or try one of these: http://www.openser.org/docs/modules/0.10.x/usrloc.html#AEN206 http://www.openser.org/docs/modules/0.10.x/registrar.html#AEN160
there are several parameters in these modules which might be interesting for you (max number of locations, ....)
klaus
Olivier Taylor wrote:
Hi all,
I have an Uac wich register each time with a different port As a result he is registered 2 or 2 times in the location table with differents ports...
Ser seems to have a problem to manage that, impossible to call this Uac from another one And also seems to have quality problems...
Olivier
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
-
Bogdan-Andrei Iancu -
Klaus Darilion -
Olivier Taylor -
Olle E. Johansson -
Subashini C V - CTD, Chennai