FS does not honour sip-force-contact. Advice?
disable session timers on both sides of the call, they are interfering with the dialog once it's established. On Thu, Oct 21, 2010 at 10:35 AM, Victor Chukalovskiy <victor.chukalovskiy@utoronto.ca> wrote:
Dear Gurus,
We have a user behind NAT. Currently we make him work by using <variable name="sip-force-contact" value="sip:1234@171.134.62.106:5060"/> It works since users IP is static and there is a port forwarding setup for port 5060 on the users side.
The problem is that after 15 minutes of a call FreeSWITCH attempts re-INVITE (RFC 4028 I guess) and does not honour port specified above. Instead, it uses some 40,000-ish UDP destination port. Obviously, this never reaches users phone and FS hangups with "expiry on timer" cause.
Is the syntax of sip-force-contact above valid?
Is it normal for re-INVITEs no to honour sip-force-contact?
I'd try NDLB-tls-connectile-dysfunction but I need clarification: -will it be honoured by re-INVITEs? -will it default to port 5060 or should I specify the port?
Thank you, Victor
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-- 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
Anthony, Thank you. Disabling timers worked out. Since session timers is a per-profile setting, what do we loose by disabling them? Thank you, Victor On 21/10/10 03:56 PM, Anthony Minessale wrote:
disable session timers on both sides of the call, they are interfering with the dialog once it's established.
On Thu, Oct 21, 2010 at 10:35 AM, Victor Chukalovskiy <victor.chukalovskiy@utoronto.ca> wrote:
Dear Gurus,
We have a user behind NAT. Currently we make him work by using <variable name="sip-force-contact" value="sip:1234@171.134.62.106:5060"/> It works since users IP is static and there is a port forwarding setup for port 5060 on the users side.
The problem is that after 15 minutes of a call FreeSWITCH attempts re-INVITE (RFC 4028 I guess) and does not honour port specified above. Instead, it uses some 40,000-ish UDP destination port. Obviously, this never reaches users phone and FS hangups with "expiry on timer" cause.
Is the syntax of sip-force-contact above valid?
Is it normal for re-INVITEs no to honour sip-force-contact?
I'd try NDLB-tls-connectile-dysfunction but I need clarification: -will it be honoured by re-INVITEs? -will it default to port 5060 or should I specify the port?
Thank you, Victor
_______________________________________________ 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
the session timers are used to make sure a call is still alive by sending a periodic re-invite to make sure the server is still there. If the devices you are interacting with can't do it properly there is not much to lose from disabling it. On Fri, Oct 22, 2010 at 1:27 PM, Victor Chukalovskiy <victor.chukalovskiy@utoronto.ca> wrote:
Anthony, Thank you. Disabling timers worked out. Since session timers is a per-profile setting, what do we loose by disabling them? Thank you, Victor
On 21/10/10 03:56 PM, Anthony Minessale wrote:
disable session timers on both sides of the call, they are interfering with the dialog once it's established.
On Thu, Oct 21, 2010 at 10:35 AM, Victor Chukalovskiy <victor.chukalovskiy@utoronto.ca> wrote:
Dear Gurus,
We have a user behind NAT. Currently we make him work by using <variable name="sip-force-contact" value="sip:1234@171.134.62.106:5060"/> It works since users IP is static and there is a port forwarding setup for port 5060 on the users side.
The problem is that after 15 minutes of a call FreeSWITCH attempts re-INVITE (RFC 4028 I guess) and does not honour port specified above. Instead, it uses some 40,000-ish UDP destination port. Obviously, this never reaches users phone and FS hangups with "expiry on timer" cause.
Is the syntax of sip-force-contact above valid?
Is it normal for re-INVITEs no to honour sip-force-contact?
I'd try NDLB-tls-connectile-dysfunction but I need clarification: -will it be honoured by re-INVITEs? -will it default to port 5060 or should I specify the port?
Thank you, Victor
_______________________________________________ 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
_______________________________________________ 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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Victor Chukalovskiy