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[Freeswitch-users] Freeswitch IVR: call drop on transfer to busy extensions


 
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francesco.piccinin at ...
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PostPosted: Sun Mar 13, 2016 9:57 am    Post subject: [Freeswitch-users] Freeswitch IVR: call drop on transfer to Reply with quote

Hi all,

I'm experiencing an issue with IVR on production Freeswitch cluster.

I'm going to explain the problem as clear as possible just to understand if anyone already get it and hopefully fix it.


Phone A makes a call that match an IVR services: call is transfered to context services in xml services.
Call is then answered, audio menĂ¹ is prompted: phone A makes a selection (ex dtmf 2) and the call is trasfered to context default xml default in order to ring phone B.

If phone B is in busy state the call is dropped.

On cli logs you can observe this behaviour:
Mod Sofia first hangup the channel with a USER BUSY cause and then it sends a BYE to Phone A (or to a gateway in case of external calls) including Q850 cause=17 user busy.

Phone A or Gateway drop the call without busy tone.


In this case I'm expecting a SIP 486 (user busy message) instead of a bye, is that correct?


I'm using the app "limit" in order to count active calls on extensions and store them in system db (shared between freeswitch servers).


Hope you can help me solving the problem.


Thanks

Regards


Francesco
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gmaruzz at gmail.com
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PostPosted: Sun Mar 13, 2016 11:39 am    Post subject: [Freeswitch-users] Freeswitch IVR: call drop on transfer to Reply with quote

Please pastebin all related dialplan and debug info, then puiblish here the pastebin link


Btw, if you are transferring the call to an internal extension, you must program yourself what you want as a result when user is busy. EG: in default "local extensions" (1000-1019) in default dialplan, call goes to voice mail.

You can have a look at "local extensions" in default dialplan for more hints.


-giovanni

On Sun, Mar 13, 2016 at 2:40 PM, Francesco Piccinin <francesco.piccinin@gmail.com (francesco.piccinin@gmail.com)> wrote:
Quote:
Hi all,

I'm experiencing an issue with IVR on production Freeswitch cluster.

I'm going to explain the problem as clear as possible just to understand if anyone already get it and hopefully fix it.


Phone A makes a call that match an IVR services: call is transfered to context services in xml services.
Call is then answered, audio menĂ¹ is prompted: phone A makes a selection (ex dtmf 2) and the call is trasfered to context default xml default in order to ring phone B.

If phone B is in busy state the call is dropped.

On cli logs you can observe this behaviour:
Mod Sofia first hangup the channel with a USER BUSY cause and then it sends a BYE to Phone A (or to a gateway in case of external calls) including Q850 cause=17 user busy.

Phone A or Gateway drop the call without busy tone.


In this case I'm expecting a SIP 486 (user busy message) instead of a bye, is that correct?


I'm using the app "limit" in order to count active calls on extensions and store them in system db (shared between freeswitch servers).


Hope you can help me solving the problem.


Thanks

Regards


Francesco












_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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



--
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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francesco.piccinin at ...
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PostPosted: Mon Mar 14, 2016 6:28 am    Post subject: [Freeswitch-users] Freeswitch IVR: call drop on transfer to Reply with quote

Hello Giovanni,

thanks for you quick answer.


Please find below patebin link you requested:

XML Dialplan file's extract:
http://pastebin.com/dmHAKWxT


Internal call fscli logs:
http://pastebin.com/KfnaYN11


External call fscli logs
http://pastebin.com/2Khp1MaX



Regarding dialplan files, I'm going to explain here call flow:

Default.xml
1.call hits extension call_local, lua script is run and channel vasr cType=SRV, service_type=IVR

2. extension apps in xml services is execute
Services.xml
lua script is run to retrieve ivr name

3. on selection (2 in the logs) call is transfer to 8910 XML calls

Calls.xml

4. extension: call_inside is execute and call is hangup after
<action application="limit" data="db ${domain_name} MAX_${cCalled} ${maxcallsB} !USER_BUSY"/>


As you can see from the logs mod sofia hangup with user busy and then send a bye to external or internal channel:

2016-03-14 11:37:40.266607 [DEBUG] mod_sofia.c:506 Channel sofia/external/00403737752@10.200.24.10 (00403737752@10.200.24.10) hanging up, cause: USER_BUSY
2016-03-14 11:37:40.266607 [DEBUG] mod_sofia.c:558 Sending BYE to sofia/external/00403737752@10.200.24.10 (00403737752@10.200.24.10)


2016-03-14 11:38:54.166606 [DEBUG] mod_sofia.c:506 Channel sofia/internal/5532@insiel.fvgvoipcoll.it (5532@insiel.fvgvoipcoll.it) hanging up, cause: USER_BUSY
2016-03-14 11:38:54.166606 [DEBUG] mod_sofia.c:558 Sending BYE to sofia/internal/5532@insiel.fvgvoipcoll.it (5532@insiel.fvgvoipcoll.it)



I tried to removed the limit app and the calls then hint limit_exceeded extension in calls.xml but behaviour is the same.

It seems to me that continue_on_fail where passing over IVR is not considered.



If I call directly (without IVR) extension 8910 (while it is busy) behaviour for both internal or external call is correct and busy tone is received.


Looking forward for a clue Smile


Thanks

Regards

Francesco
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gmaruzz at gmail.com
Guest





PostPosted: Tue Mar 15, 2016 3:25 pm    Post subject: [Freeswitch-users] Freeswitch IVR: call drop on transfer to Reply with quote

Seems the pastebins are private, not accessible.
Anyway, you are probably on the right path.
Try to recode avoiding relying on it


sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT Il 14/Mar/2016 12:27, "Francesco Piccinin" <francesco.piccinin@gmail.com (francesco.piccinin@gmail.com)> ha scritto:
Quote:
Hello Giovanni,

thanks for you quick answer.


Please find below patebin link you requested:

XML Dialplan file's extract:
http://pastebin.com/dmHAKWxT


Internal call fscli logs:
http://pastebin.com/KfnaYN11


External call fscli logs
http://pastebin.com/2Khp1MaX



Regarding dialplan files, I'm going to explain here call flow:

Default.xml
1.call hits extension call_local, lua script is run and channel vasr cType=SRV, service_type=IVR

2. extension apps in xml services is execute
Services.xml
lua script is run to retrieve ivr name

3. on selection (2 in the logs) call is transfer to 8910 XML calls

Calls.xml

4. extension: call_inside is execute and call is hangup after
<action application="limit" data="db ${domain_name} MAX_${cCalled} ${maxcallsB} !USER_BUSY"/>


As you can see from the logs mod sofia hangup with user busy and then send a bye to external or internal channel:

2016-03-14 11:37:40.266607 [DEBUG] mod_sofia.c:506 Channel sofia/external/00403737752@10.200.24.10 (00403737752@10.200.24.10) hanging up, cause: USER_BUSY
2016-03-14 11:37:40.266607 [DEBUG] mod_sofia.c:558 Sending BYE to sofia/external/00403737752@10.200.24.10 (00403737752@10.200.24.10)


2016-03-14 11:38:54.166606 [DEBUG] mod_sofia.c:506 Channel sofia/internal/5532@insiel.fvgvoipcoll.it (5532@insiel.fvgvoipcoll.it) hanging up, cause: USER_BUSY
2016-03-14 11:38:54.166606 [DEBUG] mod_sofia.c:558 Sending BYE to sofia/internal/5532@insiel.fvgvoipcoll.it (5532@insiel.fvgvoipcoll.it)



I tried to removed the limit app and the calls then hint limit_exceeded extension in calls.xml but behaviour is the same.

It seems to me that continue_on_fail where passing over IVR is not considered.



If I call directly (without IVR) extension 8910 (while it is busy) behaviour for both internal or external call is correct and busy tone is received.


Looking forward for a clue Smile


Thanks

Regards

Francesco












_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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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francesco.piccinin at ...
Guest





PostPosted: Wed Mar 16, 2016 3:43 am    Post subject: [Freeswitch-users] Freeswitch IVR: call drop on transfer to Reply with quote

Hello Giovanni and Bote,

here you are updated pastebin on suggested url:


Freeswitch dialplan files
https://pastebin.freeswitch.org/24597


FSCLI internal call log:
https://pastebin.freeswitch.org/24598


FSCLI external call log:
https://pastebin.freeswitch.org/24599



Looking forward to your feedback.


thanks

Regards


Francesco
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