Which variable should be set to exit bridge application if the call has not been answered within the specified time? call_timeout variable works if called SIP end point responds with "180 Ringing", but doesn't work if the endpoint responds with early media.
Hi Sergey Have you tried ignore_early_media? It may help. You can find more information from http://wiki.freeswitch.org/wiki/Channel_Variables On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
Which variable should be set to exit bridge application if the call has not been answered within the specified time? call_timeout variable works if called SIP end point responds with "180 Ringing", but doesn't work if the endpoint responds with early media.
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Yes, I tried ignore_early_media. call_timeout doesn't work if early media received. On Sunday 13 June 2010, Code Ghar wrote:
Hi Sergey
Have you tried ignore_early_media? It may help. You can find more information from http://wiki.freeswitch.org/wiki/Channel_Variables
On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin
<sos@sokhapkin.dyndns.org>wrote:
Which variable should be set to exit bridge application if the call has not been answered within the specified time? call_timeout variable works if called SIP end point responds with "180 Ringing", but doesn't work if the endpoint responds with early media.
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Look at leg_progress_timeout or just progress_timeout. On Sunday, June 13, 2010, Sergey Okhapkin <sos@sokhapkin.dyndns.org> wrote:
Yes, I tried ignore_early_media. call_timeout doesn't work if early media received.
On Sunday 13 June 2010, Code Ghar wrote:
Hi Sergey
Have you tried ignore_early_media? It may help. You can find more information from http://wiki.freeswitch.org/wiki/Channel_Variables
On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin
<sos@sokhapkin.dyndns.org>wrote:
Which variable should be set to exit bridge application if the call has not been answered within the specified time? call_timeout variable works if called SIP end point responds with "180 Ringing", but doesn't work if the endpoint responds with early media.
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-- João Mesquita FreeSWITCH™ Solutions t: +1 (646) 4959927
progress_timeout exits the bridge if NO 180 or 183 received, I'm receiving 183. On Sunday 13 June 2010, João Mesquita wrote:
Look at leg_progress_timeout or just progress_timeout.
On Sunday, June 13, 2010, Sergey Okhapkin <sos@sokhapkin.dyndns.org> wrote:
Yes, I tried ignore_early_media. call_timeout doesn't work if early media received.
On Sunday 13 June 2010, Code Ghar wrote:
Hi Sergey
Have you tried ignore_early_media? It may help. You can find more information from http://wiki.freeswitch.org/wiki/Channel_Variables
On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin
<sos@sokhapkin.dyndns.org>wrote:
Which variable should be set to exit bridge application if the call has not been answered within the specified time? call_timeout variable works if called SIP end point responds with "180 Ringing", but doesn't work if the endpoint responds with early media.
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What you want is a way to limit the ringing time isnt it ? I think it's not possible. I needed that some time ago, and I never found out the solution. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 13/06/2010 à 18:25, Sergey Okhapkin a écrit :
progress_timeout exits the bridge if NO 180 or 183 received, I'm receiving 183.
On Sunday 13 June 2010, João Mesquita wrote:
Look at leg_progress_timeout or just progress_timeout.
On Sunday, June 13, 2010, Sergey Okhapkin <sos@sokhapkin.dyndns.org> wrote:
Yes, I tried ignore_early_media. call_timeout doesn't work if early media received.
On Sunday 13 June 2010, Code Ghar wrote:
Hi Sergey
Have you tried ignore_early_media? It may help. You can find more information from http://wiki.freeswitch.org/wiki/Channel_Variables
On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin
<sos@sokhapkin.dyndns.org>wrote:
Which variable should be set to exit bridge application if the call has not been answered within the specified time? call_timeout variable works if called SIP end point responds with "180 Ringing", but doesn't work if the endpoint responds with early media.
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Correct, I need to limit ringing time and continue dialplan execution if no answer within N seconds. It's trivial "find me" service, serial DID forwarding to multiple PSTN numbers. On Sunday 13 June 2010, David Ponzone wrote:
What you want is a way to limit the ringing time isnt it ? I think it's not possible. I needed that some time ago, and I never found out the solution.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 13/06/2010 à 18:25, Sergey Okhapkin a écrit :
progress_timeout exits the bridge if NO 180 or 183 received, I'm receiving 183.
On Sunday 13 June 2010, João Mesquita wrote:
Look at leg_progress_timeout or just progress_timeout.
On Sunday, June 13, 2010, Sergey Okhapkin
<sos@sokhapkin.dyndns.org> wrote:
Yes, I tried ignore_early_media. call_timeout doesn't work if early media received.
On Sunday 13 June 2010, Code Ghar wrote:
Hi Sergey
Have you tried ignore_early_media? It may help. You can find more information from http://wiki.freeswitch.org/wiki/Channel_Variables
On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin
<sos@sokhapkin.dyndns.org>wrote:
Which variable should be set to exit bridge application if the call has not been answered within the specified time? call_timeout variable works if called SIP end point responds with "180 Ringing", but doesn't work if the endpoint responds with early media.
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Sergey, I did this <action application="set" data="continue_on_fail=true"/> <action application="ring_ready"/> <action application="set" data="call_timeout=20"/> before bridge and works for me Franck ----- Original Message ----- From: "Sergey Okhapkin" <sos@sokhapkin.dyndns.org> To: <freeswitch-users@lists.freeswitch.org> Sent: Sunday, June 13, 2010 2:05 PM Subject: Re: [Freeswitch-users] How to limit call attemp time? Correct, I need to limit ringing time and continue dialplan execution if no answer within N seconds. It's trivial "find me" service, serial DID forwarding to multiple PSTN numbers. On Sunday 13 June 2010, David Ponzone wrote:
What you want is a way to limit the ringing time isnt it ? I think it's not possible. I needed that some time ago, and I never found out the solution.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 13/06/2010 à 18:25, Sergey Okhapkin a écrit :
progress_timeout exits the bridge if NO 180 or 183 received, I'm receiving 183.
On Sunday 13 June 2010, João Mesquita wrote:
Look at leg_progress_timeout or just progress_timeout.
On Sunday, June 13, 2010, Sergey Okhapkin
<sos@sokhapkin.dyndns.org> wrote:
Yes, I tried ignore_early_media. call_timeout doesn't work if early media received.
On Sunday 13 June 2010, Code Ghar wrote:
Hi Sergey
Have you tried ignore_early_media? It may help. You can find more information from http://wiki.freeswitch.org/wiki/Channel_Variables
On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin
<sos@sokhapkin.dyndns.org>wrote:
Which variable should be set to exit bridge application if the call has not been answered within the specified time? call_timeout variable works if called SIP end point responds with "180 Ringing", but doesn't work if the endpoint responds with early media.
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Are you getting SIP 180 or SIP 183 from the B leg? Everything works fine when SIP 180 is received, but call_timeout doesn't work when SIP 183 is received. switch_ivr_originate.c checks MEDIA READY condition, but not for ANSWER condition, "media ready" includes early media also :-( On Sunday 13 June 2010, Madovsky wrote:
Sergey,
I did this
<action application="set" data="continue_on_fail=true"/> <action application="ring_ready"/> <action application="set" data="call_timeout=20"/>
before bridge and works for me
Franck
----- Original Message ----- From: "Sergey Okhapkin" <sos@sokhapkin.dyndns.org> To: <freeswitch-users@lists.freeswitch.org> Sent: Sunday, June 13, 2010 2:05 PM Subject: Re: [Freeswitch-users] How to limit call attemp time?
Correct, I need to limit ringing time and continue dialplan execution if no answer within N seconds.
It's trivial "find me" service, serial DID forwarding to multiple PSTN numbers.
On Sunday 13 June 2010, David Ponzone wrote:
What you want is a way to limit the ringing time isnt it ? I think it's not possible. I needed that some time ago, and I never found out the solution.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 13/06/2010 à 18:25, Sergey Okhapkin a écrit :
progress_timeout exits the bridge if NO 180 or 183 received, I'm receiving 183.
On Sunday 13 June 2010, João Mesquita wrote:
Look at leg_progress_timeout or just progress_timeout.
On Sunday, June 13, 2010, Sergey Okhapkin
<sos@sokhapkin.dyndns.org> wrote:
Yes, I tried ignore_early_media. call_timeout doesn't work if early media received.
On Sunday 13 June 2010, Code Ghar wrote:
Hi Sergey
Have you tried ignore_early_media? It may help. You can find more information from http://wiki.freeswitch.org/wiki/Channel_Variables
On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin
<sos@sokhapkin.dyndns.org>wrote: > Which variable should be set to exit bridge application if the > call > has not been answered within the specified time? call_timeout > variable > works if called > SIP end point responds with "180 Ringing", but doesn't work if the > endpoint responds with early media. > > _______________________________________________ > 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- >us e rs http://www.freeswitch.org
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How about leg_timeout on that specific leg of the call. I had an issue where a general call_timeout didn't work if there was no response but leg_timeout did work. Not sure exactly how it works if there is a response though. Worth a shot. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sergey Okhapkin Sent: Sunday, June 13, 2010 1:25 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] How to limit call attemp time? Are you getting SIP 180 or SIP 183 from the B leg? Everything works fine when SIP 180 is received, but call_timeout doesn't work when SIP 183 is received. switch_ivr_originate.c checks MEDIA READY condition, but not for ANSWER condition, "media ready" includes early media also :-( On Sunday 13 June 2010, Madovsky wrote:
Sergey,
I did this
<action application="set" data="continue_on_fail=true"/> <action application="ring_ready"/>repk <action application="set" data="call_timeout=20"/>
before bridge and works for me
Franck
----- Original Message ----- From: "Sergey Okhapkin" <sos@sokhapkin.dyndns.org> To: <freeswitch-users@lists.freeswitch.org> Sent: Sunday, June 13, 2010 2:05 PM Subject: Re: [Freeswitch-users] How to limit call attemp time?
Correct, I need to limit ringing time and continue dialplan execution if no answer within N seconds.
It's trivial "find me" service, serial DID forwarding to multiple PSTN numbers.
On Sunday 13 June 2010, David Ponzone wrote:
What you want is a way to limit the ringing time isnt it ? I think it's not possible. I needed that some time ago, and I never found out the solution.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 13/06/2010 à 18:25, Sergey Okhapkin a écrit :
progress_timeout exits the bridge if NO 180 or 183 received, I'm receiving 183.
On Sunday 13 June 2010, João Mesquita wrote:
Look at leg_progress_timeout or just progress_timeout.
On Sunday, June 13, 2010, Sergey Okhapkin
<sos@sokhapkin.dyndns.org> wrote:
Yes, I tried ignore_early_media. call_timeout doesn't work if early media received.
On Sunday 13 June 2010, Code Ghar wrote:
Hi Sergey
Have you tried ignore_early_media? It may help. You can find more information from http://wiki.freeswitch.org/wiki/Channel_Variables
On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin
<sos@sokhapkin.dyndns.org>wrote: > Which variable should be set to exit bridge application if the > call > has not been answered within the specified time? call_timeout > variable > works if called > SIP end point responds with "180 Ringing", but doesn't work if the > endpoint responds with early media. > > _______________________________________________ > 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- >us e rs http://www.freeswitch.org
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Just tried, leg_timeout doesn't work too. On Sunday 13 June 2010, Peder wrote:
How about leg_timeout on that specific leg of the call. I had an issue where a general call_timeout didn't work if there was no response but leg_timeout did work. Not sure exactly how it works if there is a response though. Worth a shot.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sergey Okhapkin Sent: Sunday, June 13, 2010 1:25 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] How to limit call attemp time?
Are you getting SIP 180 or SIP 183 from the B leg? Everything works fine when SIP 180 is received, but call_timeout doesn't work when SIP 183 is received.
switch_ivr_originate.c checks MEDIA READY condition, but not for ANSWER condition, "media ready" includes early media also :-(
On Sunday 13 June 2010, Madovsky wrote:
Sergey,
I did this
<action application="set" data="continue_on_fail=true"/> <action application="ring_ready"/>repk <action application="set" data="call_timeout=20"/>
before bridge and works for me
Franck
----- Original Message ----- From: "Sergey Okhapkin" <sos@sokhapkin.dyndns.org> To: <freeswitch-users@lists.freeswitch.org> Sent: Sunday, June 13, 2010 2:05 PM Subject: Re: [Freeswitch-users] How to limit call attemp time?
Correct, I need to limit ringing time and continue dialplan execution if
no
answer within N seconds.
It's trivial "find me" service, serial DID forwarding to multiple PSTN numbers.
On Sunday 13 June 2010, David Ponzone wrote:
What you want is a way to limit the ringing time isnt it ? I think it's not possible. I needed that some time ago, and I never found out the solution.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 13/06/2010 à 18:25, Sergey Okhapkin a écrit :
progress_timeout exits the bridge if NO 180 or 183 received, I'm receiving 183.
On Sunday 13 June 2010, João Mesquita wrote:
Look at leg_progress_timeout or just progress_timeout.
On Sunday, June 13, 2010, Sergey Okhapkin
<sos@sokhapkin.dyndns.org> wrote:
Yes, I tried ignore_early_media. call_timeout doesn't work if early media received.
On Sunday 13 June 2010, Code Ghar wrote: > Hi Sergey > > Have you tried ignore_early_media? It may help. You can find more > information from http://wiki.freeswitch.org/wiki/Channel_Variables > > On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin > > <sos@sokhapkin.dyndns.org>wrote: >> Which variable should be set to exit bridge application if the >> call >> has not been answered within the specified time? call_timeout >> variable >> works if called >> SIP end point responds with "180 Ringing", but doesn't work if >> the endpoint responds with early media. >> >> _______________________________________________ >> 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-
>>us e rs http://www.freeswitch.org
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I tested it the following way: <extension name="outside-call"> <condition field="destination_number" expression="^2125550000$"> <action application="set" data="continue_on_fail=true" /> <action application="set" data="ignore_early_media=true" /> <action application="set" data="call_timeout=10" /> <action application="bridge" data="sofia/gateway/gateway1/$1"/> <action application="sleep" data="5000" /> <action application="set" data="call_timeout=15" /> <action application="bridge" data="sofia/gateway/gateway2/$1" /> <action application="set" data="call_timeout=30" /> <action application="bridge" data="sofia/gateway/gateway3/$1" /> </condition> </extension> After 10 seconds, FS sends "cancel" to gateway1 and sends "invite" to gateway2 after sleeping for a while (optional). Once it sends invite to gateway2, after 15 seconds FS sends "cancel" to gateway2 and sends "invite" to gateway3. If after 30 seconds call is not answered it sends "cancel" to gateway3 and then does not call any other gateway. From all three gateways I received "183 Session Progress". If at any point call is answered, then subsequent gateways are not tried. For example, if call is answered through gateway1, then after hangup it doesn't attempt gateway2 and gateway3. Now comes your situation, where you need to try one gateway only. In this situation the following may be of help. <extension name="outside-call"> <condition field="destination_number" expression="^2125550000$"> <action application="set" data="continue_on_fail=true" /> <action application="set" data="ignore_early_media=true" /> <action application="set" data="call_timeout=10" /> <action application="bridge" data="sofia/gateway/gateway1/$1"/> </condition> </extension> As you can see, we have eliminated all gateways except gateway1. I have tested this myself and after 10 seconds FS sends "cancel" to gateway1 and call is disconnected. Hope this helps. I used the examples at following wiki links to create and test the above configurations. Misc. Dialplan Tools bridge http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge Dialplan FollowMe http://wiki.freeswitch.org/wiki/Dialplan_FollowMe On Sun, Jun 13, 2010 at 1:56 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
Just tried, leg_timeout doesn't work too.
On Sunday 13 June 2010, Peder wrote:
How about leg_timeout on that specific leg of the call. I had an issue where a general call_timeout didn't work if there was no response but leg_timeout did work. Not sure exactly how it works if there is a response though. Worth a shot.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sergey Okhapkin Sent: Sunday, June 13, 2010 1:25 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] How to limit call attemp time?
Are you getting SIP 180 or SIP 183 from the B leg? Everything works fine when SIP 180 is received, but call_timeout doesn't work when SIP 183 is received.
switch_ivr_originate.c checks MEDIA READY condition, but not for ANSWER condition, "media ready" includes early media also :-(
On Sunday 13 June 2010, Madovsky wrote:
Sergey,
I did this
<action application="set" data="continue_on_fail=true"/> <action application="ring_ready"/>repk <action application="set" data="call_timeout=20"/>
before bridge and works for me
Franck
----- Original Message ----- From: "Sergey Okhapkin" <sos@sokhapkin.dyndns.org> To: <freeswitch-users@lists.freeswitch.org> Sent: Sunday, June 13, 2010 2:05 PM Subject: Re: [Freeswitch-users] How to limit call attemp time?
Correct, I need to limit ringing time and continue dialplan execution if
no
answer within N seconds.
It's trivial "find me" service, serial DID forwarding to multiple PSTN numbers.
On Sunday 13 June 2010, David Ponzone wrote:
What you want is a way to limit the ringing time isnt it ? I think it's not possible. I needed that some time ago, and I never found out the solution.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 13/06/2010 à 18:25, Sergey Okhapkin a écrit :
progress_timeout exits the bridge if NO 180 or 183 received, I'm receiving 183.
On Sunday 13 June 2010, João Mesquita wrote:
Look at leg_progress_timeout or just progress_timeout.
On Sunday, June 13, 2010, Sergey Okhapkin
<sos@sokhapkin.dyndns.org> wrote: > Yes, I tried ignore_early_media. call_timeout doesn't work if > early media > received. > > On Sunday 13 June 2010, Code Ghar wrote: >> Hi Sergey >> >> Have you tried ignore_early_media? It may help. You can find more >> information from http://wiki.freeswitch.org/wiki/Channel_Variables >> >> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin >> >> <sos@sokhapkin.dyndns.org>wrote: >>> Which variable should be set to exit bridge application if the >>> call >>> has not been answered within the specified time? call_timeout >>> variable >>> works if called >>> SIP end point responds with "180 Ringing", but doesn't work if >>> the endpoint responds with early media. >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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Thanks! bypass_media=true in combination with ignore_early_media=true did the trick. On Sunday 13 June 2010, Code Ghar wrote:
I tested it the following way:
<extension name="outside-call"> <condition field="destination_number" expression="^2125550000$"> <action application="set" data="continue_on_fail=true" /> <action application="set" data="ignore_early_media=true" /> <action application="set" data="call_timeout=10" /> <action application="bridge" data="sofia/gateway/gateway1/$1"/> <action application="sleep" data="5000" /> <action application="set" data="call_timeout=15" /> <action application="bridge" data="sofia/gateway/gateway2/$1" /> <action application="set" data="call_timeout=30" /> <action application="bridge" data="sofia/gateway/gateway3/$1" /> </condition> </extension>
After 10 seconds, FS sends "cancel" to gateway1 and sends "invite" to gateway2 after sleeping for a while (optional). Once it sends invite to gateway2, after 15 seconds FS sends "cancel" to gateway2 and sends "invite" to gateway3. If after 30 seconds call is not answered it sends "cancel" to gateway3 and then does not call any other gateway. From all three gateways I received "183 Session Progress".
If at any point call is answered, then subsequent gateways are not tried. For example, if call is answered through gateway1, then after hangup it doesn't attempt gateway2 and gateway3.
Now comes your situation, where you need to try one gateway only. In this situation the following may be of help.
<extension name="outside-call"> <condition field="destination_number" expression="^2125550000$"> <action application="set" data="continue_on_fail=true" /> <action application="set" data="ignore_early_media=true" /> <action application="set" data="call_timeout=10" /> <action application="bridge" data="sofia/gateway/gateway1/$1"/> </condition> </extension>
As you can see, we have eliminated all gateways except gateway1. I have tested this myself and after 10 seconds FS sends "cancel" to gateway1 and call is disconnected.
Hope this helps. I used the examples at following wiki links to create and test the above configurations.
Misc. Dialplan Tools bridge http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge
Dialplan FollowMe http://wiki.freeswitch.org/wiki/Dialplan_FollowMe
On Sun, Jun 13, 2010 at 1:56 PM, Sergey Okhapkin
<sos@sokhapkin.dyndns.org>wrote:
Just tried, leg_timeout doesn't work too.
On Sunday 13 June 2010, Peder wrote:
How about leg_timeout on that specific leg of the call. I had an issue where a general call_timeout didn't work if there was no response but leg_timeout did work. Not sure exactly how it works if there is a
response
though. Worth a shot.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of
Sergey
Okhapkin Sent: Sunday, June 13, 2010 1:25 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] How to limit call attemp time?
Are you getting SIP 180 or SIP 183 from the B leg? Everything works fine when SIP 180 is received, but call_timeout doesn't work when SIP 183 is received.
switch_ivr_originate.c checks MEDIA READY condition, but not for ANSWER condition, "media ready" includes early media also :-(
On Sunday 13 June 2010, Madovsky wrote:
Sergey,
I did this
<action application="set" data="continue_on_fail=true"/> <action application="ring_ready"/>repk <action application="set" data="call_timeout=20"/>
before bridge and works for me
Franck
----- Original Message ----- From: "Sergey Okhapkin" <sos@sokhapkin.dyndns.org> To: <freeswitch-users@lists.freeswitch.org> Sent: Sunday, June 13, 2010 2:05 PM Subject: Re: [Freeswitch-users] How to limit call attemp time?
Correct, I need to limit ringing time and continue dialplan execution
if
no
answer within N seconds.
It's trivial "find me" service, serial DID forwarding to multiple PSTN numbers.
On Sunday 13 June 2010, David Ponzone wrote:
What you want is a way to limit the ringing time isnt it ? I think it's not possible. I needed that some time ago, and I never found out the solution.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 13/06/2010 à 18:25, Sergey Okhapkin a écrit :
progress_timeout exits the bridge if NO 180 or 183 received, I'm receiving 183.
On Sunday 13 June 2010, João Mesquita wrote: > Look at leg_progress_timeout or just progress_timeout. > > On Sunday, June 13, 2010, Sergey Okhapkin > > <sos@sokhapkin.dyndns.org> wrote: >> Yes, I tried ignore_early_media. call_timeout doesn't work if >> early media >> received. >> >> On Sunday 13 June 2010, Code Ghar wrote: >>> Hi Sergey >>> >>> Have you tried ignore_early_media? It may help. You can find
more
>>> information from
http://wiki.freeswitch.org/wiki/Channel_Variables
>>> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin >>> >>> <sos@sokhapkin.dyndns.org>wrote: >>>> Which variable should be set to exit bridge application if >>>> the call >>>> has not been answered within the specified time? call_timeout >>>> variable >>>> works if called >>>> SIP end point responds with "180 Ringing", but doesn't work >>>> if the endpoint responds with early media. >>>> >>>> _______________________________________________ >>>> 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-
>>>>us e rs 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-us
>>er s http://www.freeswitch.org
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However I think it's FS bug. call_timeout should wait for answer, regardless of early media. On Sunday 13 June 2010, Sergey Okhapkin wrote:
Thanks! bypass_media=true in combination with ignore_early_media=true did the trick.
On Sunday 13 June 2010, Code Ghar wrote:
I tested it the following way:
<extension name="outside-call"> <condition field="destination_number" expression="^2125550000$"> <action application="set" data="continue_on_fail=true" /> <action application="set" data="ignore_early_media=true" /> <action application="set" data="call_timeout=10" /> <action application="bridge" data="sofia/gateway/gateway1/$1"/> <action application="sleep" data="5000" /> <action application="set" data="call_timeout=15" /> <action application="bridge" data="sofia/gateway/gateway2/$1" /> <action application="set" data="call_timeout=30" /> <action application="bridge" data="sofia/gateway/gateway3/$1" /> </condition> </extension>
After 10 seconds, FS sends "cancel" to gateway1 and sends "invite" to gateway2 after sleeping for a while (optional). Once it sends invite to gateway2, after 15 seconds FS sends "cancel" to gateway2 and sends "invite" to gateway3. If after 30 seconds call is not answered it sends "cancel" to gateway3 and then does not call any other gateway. From all three gateways I received "183 Session Progress".
If at any point call is answered, then subsequent gateways are not tried. For example, if call is answered through gateway1, then after hangup it doesn't attempt gateway2 and gateway3.
Now comes your situation, where you need to try one gateway only. In this situation the following may be of help.
<extension name="outside-call"> <condition field="destination_number" expression="^2125550000$"> <action application="set" data="continue_on_fail=true" /> <action application="set" data="ignore_early_media=true" /> <action application="set" data="call_timeout=10" /> <action application="bridge" data="sofia/gateway/gateway1/$1"/> </condition> </extension>
As you can see, we have eliminated all gateways except gateway1. I have tested this myself and after 10 seconds FS sends "cancel" to gateway1 and call is disconnected.
Hope this helps. I used the examples at following wiki links to create and test the above configurations.
Misc. Dialplan Tools bridge http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge
Dialplan FollowMe http://wiki.freeswitch.org/wiki/Dialplan_FollowMe
On Sun, Jun 13, 2010 at 1:56 PM, Sergey Okhapkin
<sos@sokhapkin.dyndns.org>wrote:
Just tried, leg_timeout doesn't work too.
On Sunday 13 June 2010, Peder wrote:
How about leg_timeout on that specific leg of the call. I had an issue where a general call_timeout didn't work if there was no response but leg_timeout did work. Not sure exactly how it works if there is a
response
though. Worth a shot.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of
Sergey
Okhapkin Sent: Sunday, June 13, 2010 1:25 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] How to limit call attemp time?
Are you getting SIP 180 or SIP 183 from the B leg? Everything works fine when SIP 180 is received, but call_timeout doesn't work when SIP 183 is received.
switch_ivr_originate.c checks MEDIA READY condition, but not for ANSWER condition, "media ready" includes early media also :-(
On Sunday 13 June 2010, Madovsky wrote:
Sergey,
I did this
<action application="set" data="continue_on_fail=true"/> <action application="ring_ready"/>repk <action application="set" data="call_timeout=20"/>
before bridge and works for me
Franck
----- Original Message ----- From: "Sergey Okhapkin" <sos@sokhapkin.dyndns.org> To: <freeswitch-users@lists.freeswitch.org> Sent: Sunday, June 13, 2010 2:05 PM Subject: Re: [Freeswitch-users] How to limit call attemp time?
Correct, I need to limit ringing time and continue dialplan execution
if
no
answer within N seconds.
It's trivial "find me" service, serial DID forwarding to multiple PSTN numbers.
On Sunday 13 June 2010, David Ponzone wrote:
What you want is a way to limit the ringing time isnt it ? I think it's not possible. I needed that some time ago, and I never found out the solution.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 13/06/2010 à 18:25, Sergey Okhapkin a écrit : > progress_timeout exits the bridge if NO 180 or 183 received, > I'm receiving > 183. > > On Sunday 13 June 2010, João Mesquita wrote: >> Look at leg_progress_timeout or just progress_timeout. >> >> On Sunday, June 13, 2010, Sergey Okhapkin >> >> <sos@sokhapkin.dyndns.org> wrote: >>> Yes, I tried ignore_early_media. call_timeout doesn't work if >>> early media >>> received. >>> >>> On Sunday 13 June 2010, Code Ghar wrote: >>>> Hi Sergey >>>> >>>> Have you tried ignore_early_media? It may help. You can find
more
>>>> information from
http://wiki.freeswitch.org/wiki/Channel_Variables
>>>> On Sun, Jun 13, 2010 at 10:42 AM, Sergey Okhapkin >>>> >>>> <sos@sokhapkin.dyndns.org>wrote: >>>>> Which variable should be set to exit bridge application if >>>>> the call >>>>> has not been answered within the specified time? >>>>> call_timeout variable >>>>> works if called >>>>> SIP end point responds with "180 Ringing", but doesn't work >>>>> if the endpoint responds with early media. >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-use >>>>>rs
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>us e rs 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-us
>>>er s http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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On Sun, Jun 13, 2010 at 3:02 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
However I think it's FS bug. call_timeout should wait for answer, regardless of early media.
I'm not sure that statement is correct. The purpose of the call_timeout variable is documented here: http://wiki.freeswitch.org/wiki/Channel_Variables#call_timeout Perhaps you needed this var: http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_answer_timeout According to the description, it sounds exactly like what you were asking about. Also, you can attack this problem from a totally different angle but using execute_on_answer: http://wiki.freeswitch.org/wiki/Variable_execute_on_answer And then you can do whatever you want for 20 seconds (or however long) and if the far end never answers then you can move on and handle the no answer condition. -MC
participants (7)
-
Code Ghar
-
David Ponzone
-
João Mesquita
-
Madovsky
-
Michael Collins
-
Peder
-
Sergey Okhapkin