Bad P-Asserted-Identity header from FS conference.
Hello, I am not sure what triggers this but I get a bad P-Asserted-Identity header from Free-SWITCH in an UPDATE request when I create a conference and send an INVITE to that conference. Here is the offending Request : UPDATE sip:502@192.168.5.75:6060;x-sipX-nonat SIP/2.0 Via: SIP/2.0/UDP 192.168.5.75:15060;rport;branch=z9hG4bK2ZtpZ94m4Zg7F Route: <sip:192.168.5.75:5060;lr;sipXecs-rs=*auth~.*from~OUQ1MENEMEQtODAzNjU5QkU%60.900_ntap*id~MTQ0MDMtOQ%60%60!34e982a809955a84b401940d3c785d61> Max-Forwards: 70 From: <sip:502@sipxtest.sipxtest.net;user=phone>;tag=2ecF2jHpjjH6c To: "user1" <sip:user1@sipxtest.sipxtest.net>;tag=9D50CD0D-803659BE Call-ID: 41745e29-669e33ca-619711cf@192.168.5.242 CSeq: 6399341 UPDATE Contact: <sip:502@192.168.5.75:15060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-2 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 297 P-Asserted-Identity: "conference" <conference> p-asserted-identity should specify a SIP URI and it does not. Is this indicative of a misconfiguration somewhere or is it a bug? Where is the P-A-I header constructed from? Thanks Regards, Ranga -- M. Ranganathan
What is the client? Those should only be sent to supported devices which are stupid phones who actually want it that way. There is some code to detect it based on the user agent. On Mon, Dec 27, 2010 at 11:35 AM, M. Ranganathan <mranga@gmail.com> wrote:
Hello,
I am not sure what triggers this but I get a bad P-Asserted-Identity header from Free-SWITCH in an UPDATE request when I create a conference and send an INVITE to that conference. Here is the offending Request :
UPDATE sip:502@192.168.5.75:6060;x-sipX-nonat SIP/2.0 Via: SIP/2.0/UDP 192.168.5.75:15060;rport;branch=z9hG4bK2ZtpZ94m4Zg7F Route: <sip:192.168.5.75:5060;lr;sipXecs-rs=*auth~.*from~OUQ1MENEMEQtODAzNjU5QkU%60.900_ntap*id~MTQ0MDMtOQ%60%60!34e982a809955a84b401940d3c785d61> Max-Forwards: 70 From: <sip:502@sipxtest.sipxtest.net;user=phone>;tag=2ecF2jHpjjH6c To: "user1" <sip:user1@sipxtest.sipxtest.net>;tag=9D50CD0D-803659BE Call-ID: 41745e29-669e33ca-619711cf@192.168.5.242 CSeq: 6399341 UPDATE Contact: <sip:502@192.168.5.75:15060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-2 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 297 P-Asserted-Identity: "conference" <conference>
p-asserted-identity should specify a SIP URI and it does not.
Is this indicative of a misconfiguration somewhere or is it a bug? Where is the P-A-I header constructed from?
Thanks
Regards,
Ranga
-- M. Ranganathan
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
The user agent header for the Phone that sends the INVITE in this case is User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.3.0.1098 Thanks Ranga On Mon, Dec 27, 2010 at 12:53 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
What is the client? Those should only be sent to supported devices which are stupid phones who actually want it that way. There is some code to detect it based on the user agent.
On Mon, Dec 27, 2010 at 11:35 AM, M. Ranganathan <mranga@gmail.com> wrote:
Hello,
I am not sure what triggers this but I get a bad P-Asserted-Identity header from Free-SWITCH in an UPDATE request when I create a conference and send an INVITE to that conference. Here is the offending Request :
UPDATE sip:502@192.168.5.75:6060;x-sipX-nonat SIP/2.0 Via: SIP/2.0/UDP 192.168.5.75:15060;rport;branch=z9hG4bK2ZtpZ94m4Zg7F Route: <sip:192.168.5.75:5060;lr;sipXecs-rs=*auth~.*from~OUQ1MENEMEQtODAzNjU5QkU%60.900_ntap*id~MTQ0MDMtOQ%60%60!34e982a809955a84b401940d3c785d61> Max-Forwards: 70 From: <sip:502@sipxtest.sipxtest.net;user=phone>;tag=2ecF2jHpjjH6c To: "user1" <sip:user1@sipxtest.sipxtest.net>;tag=9D50CD0D-803659BE Call-ID: 41745e29-669e33ca-619711cf@192.168.5.242 CSeq: 6399341 UPDATE Contact: <sip:502@192.168.5.75:15060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-2 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 297 P-Asserted-Identity: "conference" <conference>
p-asserted-identity should specify a SIP URI and it does not.
Is this indicative of a misconfiguration somewhere or is it a bug? Where is the P-A-I header constructed from?
Thanks
Regards,
Ranga
-- M. Ranganathan
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- M. Ranganathan
This is exactly how polycom expects the header. anything else is incorrectly displayed on the screen. We actually force the packet to look this way when user agent says polycom. Judging the contents of the header when acting as a proxy is a bit over-zealous. On Mon, Dec 27, 2010 at 12:10 PM, M. Ranganathan <mranga@gmail.com> wrote:
The user agent header for the Phone that sends the INVITE in this case is
User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.3.0.1098
Thanks
Ranga
On Mon, Dec 27, 2010 at 12:53 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
What is the client? Those should only be sent to supported devices which are stupid phones who actually want it that way. There is some code to detect it based on the user agent.
On Mon, Dec 27, 2010 at 11:35 AM, M. Ranganathan <mranga@gmail.com> wrote:
Hello,
I am not sure what triggers this but I get a bad P-Asserted-Identity header from Free-SWITCH in an UPDATE request when I create a conference and send an INVITE to that conference. Here is the offending Request :
UPDATE sip:502@192.168.5.75:6060;x-sipX-nonat SIP/2.0 Via: SIP/2.0/UDP 192.168.5.75:15060;rport;branch=z9hG4bK2ZtpZ94m4Zg7F Route: <sip:192.168.5.75:5060;lr;sipXecs-rs=*auth~.*from~OUQ1MENEMEQtODAzNjU5QkU%60.900_ntap*id~MTQ0MDMtOQ%60%60!34e982a809955a84b401940d3c785d61> Max-Forwards: 70 From: <sip:502@sipxtest.sipxtest.net;user=phone>;tag=2ecF2jHpjjH6c To: "user1" <sip:user1@sipxtest.sipxtest.net>;tag=9D50CD0D-803659BE Call-ID: 41745e29-669e33ca-619711cf@192.168.5.242 CSeq: 6399341 UPDATE Contact: <sip:502@192.168.5.75:15060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-2 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 297 P-Asserted-Identity: "conference" <conference>
p-asserted-identity should specify a SIP URI and it does not.
Is this indicative of a misconfiguration somewhere or is it a bug? Where is the P-A-I header constructed from?
Thanks
Regards,
Ranga
-- M. Ranganathan
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- M. Ranganathan
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
participants (2)
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Anthony Minessale
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M. Ranganathan