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gkuri at ieee.org Guest
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Posted: Fri Nov 28, 2008 4:06 pm Post subject: [Freeswitch-users] SIP INVITE timeout |
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I have a phone that is registered to FS but is no longer available
(Internet connection down, phone turned off, etc.). The registration
still exists in the sip_registrations table (not expired yet), but the
phone is not reachable on the network.
According to my dialplan, if the bridge to the phone fails after 20
seconds, the call should be forwarded to a different box for handling
(see dialplan below).
<extension name="My Extension">
<condition field="destination_number" expression="1213XXXXXXX">
<action application="info"/>
<action application="set" data="originate_timeout=20"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge"
data="${sofia_contact(default/1213XXXXXXX@mydomain.net"/>
<action application="bridge"
data="sofia/default/1213XXXXXXX@box.mydomain.net"/>
</condition>
</extension>
If the phone is down and not responding to the INVITEs, it appears my
carrier is canceling the SIP INVITE to FreeSWITCH after about 10
seconds. My timeout is 20 seconds. Is there anyway to deal with this
situation, without going back to my carrier and asking them to increase
their timeout on an INVITE?
Call Progress:
Carrier -> FS (INVITE)
FS -> Carrier (100 Trying)
<10 seconds pass while FS is attempting to contact the phone>
Carrier -> FS (CANCEL)
FS -> Carrier (200 OK)
FS -> Carrier (487 Request Terminated)
Carrier -> FS (ACK)
Thanks ...
Gabe
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brian at freeswitch.org Guest
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Posted: Fri Nov 28, 2008 5:38 pm Post subject: [Freeswitch-users] SIP INVITE timeout |
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Try pre_answer before bridge.
/b
Sent from my iPhone
On Nov 28, 2008, at 3:03 PM, Gabriel Kuri <gkuri@ieee.org> wrote:
Quote: | I have a phone that is registered to FS but is no longer available
(Internet connection down, phone turned off, etc.). The registration
still exists in the sip_registrations table (not expired yet), but the
phone is not reachable on the network.
According to my dialplan, if the bridge to the phone fails after 20
seconds, the call should be forwarded to a different box for handling
(see dialplan below).
<extension name="My Extension">
<condition field="destination_number" expression="1213XXXXXXX">
<action application="info"/>
<action application="set" data="originate_timeout=20"/>
<action application="set"
data="hangup_after_bridge=true"/>
<action application="bridge"
data="${sofia_contact(default/1213XXXXXXX@mydomain.net"/>
<action application="bridge"
data="sofia/default/1213XXXXXXX@box.mydomain.net"/>
</condition>
</extension>
If the phone is down and not responding to the INVITEs, it appears my
carrier is canceling the SIP INVITE to FreeSWITCH after about 10
seconds. My timeout is 20 seconds. Is there anyway to deal with this
situation, without going back to my carrier and asking them to
increase
their timeout on an INVITE?
Call Progress:
Carrier -> FS (INVITE)
FS -> Carrier (100 Trying)
<10 seconds pass while FS is attempting to contact the phone>
Carrier -> FS (CANCEL)
FS -> Carrier (200 OK)
FS -> Carrier (487 Request Terminated)
Carrier -> FS (ACK)
Thanks ...
Gabe
_______________________________________________
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
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gkuri at ieee.org Guest
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Posted: Fri Nov 28, 2008 6:46 pm Post subject: [Freeswitch-users] SIP INVITE timeout |
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that works, but the calling party doesn't get ringback when the phone is
actually on and ringing, is that normal? I don't think I can add
application ringback, because it's using g729.
Gabe
Brian West wrote:
Quote: | Try pre_answer before bridge.
/b
Sent from my iPhone
On Nov 28, 2008, at 3:03 PM, Gabriel Kuri <gkuri@ieee.org> wrote:
Quote: | I have a phone that is registered to FS but is no longer available
(Internet connection down, phone turned off, etc.). The registration
still exists in the sip_registrations table (not expired yet), but the
phone is not reachable on the network.
According to my dialplan, if the bridge to the phone fails after 20
seconds, the call should be forwarded to a different box for handling
(see dialplan below).
<extension name="My Extension">
<condition field="destination_number" expression="1213XXXXXXX">
<action application="info"/>
<action application="set" data="originate_timeout=20"/>
<action application="set"
data="hangup_after_bridge=true"/>
<action application="bridge"
data="${sofia_contact(default/1213XXXXXXX@mydomain.net"/>
<action application="bridge"
data="sofia/default/1213XXXXXXX@box.mydomain.net"/>
</condition>
</extension>
If the phone is down and not responding to the INVITEs, it appears my
carrier is canceling the SIP INVITE to FreeSWITCH after about 10
seconds. My timeout is 20 seconds. Is there anyway to deal with this
situation, without going back to my carrier and asking them to
increase
their timeout on an INVITE?
Call Progress:
Carrier -> FS (INVITE)
FS -> Carrier (100 Trying)
<10 seconds pass while FS is attempting to contact the phone>
Carrier -> FS (CANCEL)
FS -> Carrier (200 OK)
FS -> Carrier (487 Request Terminated)
Carrier -> FS (ACK)
Thanks ...
Gabe
_______________________________________________
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
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_______________________________________________
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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
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_______________________________________________
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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gkuri at ieee.org Guest
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Posted: Mon Dec 01, 2008 2:39 pm Post subject: [Freeswitch-users] SIP INVITE timeout |
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Brian,
Will setting progress_timeout = 8 and originate_timeout = 30 help me out
in this situation without using pre_answer?
Basically I'd like to timeout the INVITE to the phone in 8 seconds if it
doesn't respond to the INVITE (phone is not on the network) and send the
call to voicemail, but if the phone is actually ringing and no one picks
up in 30 seconds, send it to voicemail?
Thanks
Gabe
Brian West wrote:
Quote: | Try pre_answer before bridge.
/b
Sent from my iPhone
On Nov 28, 2008, at 3:03 PM, Gabriel Kuri <gkuri@ieee.org> wrote:
Quote: | I have a phone that is registered to FS but is no longer available
(Internet connection down, phone turned off, etc.). The registration
still exists in the sip_registrations table (not expired yet), but the
phone is not reachable on the network.
According to my dialplan, if the bridge to the phone fails after 20
seconds, the call should be forwarded to a different box for handling
(see dialplan below).
<extension name="My Extension">
<condition field="destination_number" expression="1213XXXXXXX">
<action application="info"/>
<action application="set" data="originate_timeout=20"/>
<action application="set"
data="hangup_after_bridge=true"/>
<action application="bridge"
data="${sofia_contact(default/1213XXXXXXX@mydomain.net"/>
<action application="bridge"
data="sofia/default/1213XXXXXXX@box.mydomain.net"/>
</condition>
</extension>
If the phone is down and not responding to the INVITEs, it appears my
carrier is canceling the SIP INVITE to FreeSWITCH after about 10
seconds. My timeout is 20 seconds. Is there anyway to deal with this
situation, without going back to my carrier and asking them to
increase
their timeout on an INVITE?
Call Progress:
Carrier -> FS (INVITE)
FS -> Carrier (100 Trying)
<10 seconds pass while FS is attempting to contact the phone>
Carrier -> FS (CANCEL)
FS -> Carrier (200 OK)
FS -> Carrier (487 Request Terminated)
Carrier -> FS (ACK)
Thanks ...
Gabe
_______________________________________________
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
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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
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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