Hello, is there a way to check if a UA is still available? The reason is, that sometimes a channel hangs and is not being terminated because the UA was disconnected due to a network failure. As the media is outside Freeswitch (bypass media), it is difficult for us to determine, if the call is still active. Meanwhile the B leg to PSTN is also not hungup and produces costs. Is there a way Freeswitch can detect this? Or is there a way Freeswitch may send a SIP message to the UA by an external process and wait for an OK which can be captured e.g. by event socket interface? An help is welcome. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#unregister-on-opti... http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#nat-options-ping On Tue, Jun 7, 2011 at 3:36 PM, Peter Steinbach <lists@telefaks.de> wrote:
Hello,
is there a way to check if a UA is still available? The reason is, that sometimes a channel hangs and is not being terminated because the UA was disconnected due to a network failure. As the media is outside Freeswitch (bypass media), it is difficult for us to determine, if the call is still active. Meanwhile the B leg to PSTN is also not hungup and produces costs.
Is there a way Freeswitch can detect this? Or is there a way Freeswitch may send a SIP message to the UA by an external process and wait for an OK which can be captured e.g. by event socket interface?
An help is welcome.
-- With kind regards Peter Steinbach
Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
_______________________________________________ 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
-- Kristian Kielhofner
Thanks Kristian, does that also mean that both legs of the call for this UA will be terminated on deregister? Best regards Peter Am 07.06.2011 22:02, schrieb Kristian Kielhofner:
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#unregister-on-opti... http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#nat-options-ping
On Tue, Jun 7, 2011 at 3:36 PM, Peter Steinbach<lists@telefaks.de> wrote:
Hello,
is there a way to check if a UA is still available? The reason is, that sometimes a channel hangs and is not being terminated because the UA was disconnected due to a network failure. As the media is outside Freeswitch (bypass media), it is difficult for us to determine, if the call is still active. Meanwhile the B leg to PSTN is also not hungup and produces costs.
Is there a way Freeswitch can detect this? Or is there a way Freeswitch may send a SIP message to the UA by an external process and wait for an OK which can be captured e.g. by event socket interface?
An help is welcome.
-- With kind regards Peter Steinbach
Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
_______________________________________________ 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
-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
Peter, No. Registration is separate from any calls. If the device fails to respond to OPTIONS it will be de-registered and inbound calls to that user/device will fail immediately because it isn't registered. Any ongoing calls with or incoming calls from that device will proceed as normal. On Tue, Jun 7, 2011 at 4:46 PM, Peter Steinbach <lists@telefaks.de> wrote:
Thanks Kristian,
does that also mean that both legs of the call for this UA will be terminated on deregister?
Best regards Peter
-- Kristian Kielhofner
What if we want to even hungup the channels associated with those users? does freeswitch helps ? Like, i have a scenario in which , an active call disrupts , once BOTH users gets disconnected from network. Here, both users gets unregistered after option messages but the call sustains. Is there any thing i can handle this scenario? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6... Sent from the freeswitch-users mailing list archive at Nabble.com.
options pings are really just for registered users. a user does not have to be registered to place a call. more useful in your scenario is rtp timers and session timers Ken Sent from my iPad On May 2, 2013, at 8:49, mehroz <mehroz.ashraf85@gmail.com> wrote:
What if we want to even hungup the channels associated with those users? does freeswitch helps ?
Like, i have a scenario in which , an active call disrupts , once BOTH users gets disconnected from network. Here, both users gets unregistered after option messages but the call sustains. Is there any thing i can handle this scenario?
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6... Sent from the freeswitch-users mailing list archive at Nabble.com.
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i cannot use session-timers, disabled due to an issue with video calling.... and rtp timers just does not seems to be working! i have But once network disconnects from both clients (works good, with a single client network outage), FS does not hangs up the call ..... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6... Sent from the freeswitch-users mailing list archive at Nabble.com.
On Thu, May 2, 2013 at 8:13 AM, mehroz <mehroz.ashraf85@gmail.com> wrote:
i cannot use session-timers, disabled due to an issue with video calling....
and rtp timers just does not seems to be working!
Can you elaborate on this? Do you have a full pcap and FS debug log of a call with RTP in both directions and then the incoming RTP stream stopping and yet the call not being disconnected?
i have
But once network disconnects from both clients (works good, with a single client network outage), FS does not hangs up the call .....
Correct. No SIP signaling and no session timers means there's nothing left to tell FS that the call should be torn down, except for the RTP timers mentioned above. That's why you'll need to supply a complete debug w/ pcap in order to find out why RTP timers are not working. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
My configuration of SIP profile are: param name="nat-options-ping" value="true" param name="all-reg-options-ping" value="true" param name="unregister-on-options-fail" value="true" param name="enable-timer" value="false" and this is because, i was experiencing video session starting up within audio call, SDP session update once session timers completes, That was totally unexplained, and i eventually had to disable timer, which ultimately resolved that issue. What left is , RTP timers, and i have param name="rtp-timer-name" value="soft" param name="rtp-timeout-sec" value="15" but it is not helpful. Channel hangs up once only a client becomes unreachable. But if both becomes unreachable, Freeswitch is helpless!! which logs are required to dig it in details and what approach shall be considered? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6... Sent from the freeswitch-users mailing list archive at Nabble.com.
On Fri, May 3, 2013 at 12:40 AM, mehroz <mehroz.ashraf85@gmail.com> wrote:
My configuration of SIP profile are:
param name="nat-options-ping" value="true" param name="all-reg-options-ping" value="true" param name="unregister-on-options-fail" value="true"
param name="enable-timer" value="false" and this is because, i was experiencing video session starting up within audio call, SDP session update once session timers completes, That was totally unexplained, and i eventually had to disable timer, which ultimately resolved that issue.
What left is , RTP timers, and i have param name="rtp-timer-name" value="soft" param name="rtp-timeout-sec" value="15" but it is not helpful. Channel hangs up once only a client becomes unreachable. But if both becomes unreachable, Freeswitch is helpless!!
which logs are required to dig it in details and what approach shall be considered?
Same thing I mentioned before: pcap with both SIP and RTP debug log of call not behaving properly. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
participants (5)
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Ken Rice
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Kristian Kielhofner
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mehroz
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Michael Collins
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Peter Steinbach