Hi. I'm using mod_skypeopen and everything's working fine. The call comes in executes ivr and on DTMF it is forwarded into a queue. But for some reasons the phone aren't ringing. I have another trunk attached as a gateway. When I call from that number the phones rings fine. Can someone tell me why that is? Thanks.
sent from my mobile, Giovanni Maruzzelli cell: +39 347 266 56 18 On Apr 8, 2015 1:51 PM, "Ali Jibran" <ali.jibran44@gmail.com> wrote:
Hi. I'm using mod_skypeopen and everything's working fine.
The call comes in executes ivr and on DTMF it is forwarded into a queue.
But for some reasons the phone aren't ringing.
I have another trunk attached as a gateway. When I call from that number
the phones rings fine.
Can someone tell me why that is? Thanks.
Is very strange, once inside FS a skype call behaves like a sip call... Maybe you forgot to set some channel variable? Or something else, pertaining to dialplan/scripting? If you cannot find the reason, please look in the skypopen confluence (wiki) page how to report a problem, and open a jira. Is important to give us a way to re produce the problem, so remember to read the skypopen page paragraph on how to report a.problem. -giovanni
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
mod_callcenter or mod_fifo? Post your debug logs reproducing de problem and your configs On Wed, Apr 8, 2015 at 2:01 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
sent from my mobile, Giovanni Maruzzelli cell: +39 347 266 56 18
On Apr 8, 2015 1:51 PM, "Ali Jibran" <ali.jibran44@gmail.com> wrote:
Hi. I'm using mod_skypeopen and everything's working fine.
The call comes in executes ivr and on DTMF it is forwarded into a queue.
But for some reasons the phone aren't ringing.
I have another trunk attached as a gateway. When I call from that number
the phones rings fine.
Can someone tell me why that is? Thanks.
Is very strange, once inside FS a skype call behaves like a sip call...
Maybe you forgot to set some channel variable? Or something else, pertaining to dialplan/scripting?
If you cannot find the reason, please look in the skypopen confluence (wiki) page how to report a problem, and open a jira. Is important to give us a way to re produce the problem, so remember to read the skypopen page paragraph on how to report a.problem.
-giovanni
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ítalo Rossi
Mod_callcenter I'll post the logs...but as far as I can see is that it says new channel but doesn't go to ring ready when calling from skypeopen. Where as when calling normally from the trunk the logs show ring ready state. That's the only difference I can see. On Wednesday, April 8, 2015, Ítalo Rossi <italorossib@gmail.com> wrote:
mod_callcenter or mod_fifo?
Post your debug logs reproducing de problem and your configs
On Wed, Apr 8, 2015 at 2:01 PM, Giovanni Maruzzelli <gmaruzz@gmail.com <javascript:_e(%7B%7D,'cvml','gmaruzz@gmail.com');>> wrote:
sent from my mobile, Giovanni Maruzzelli cell: +39 347 266 56 18
On Apr 8, 2015 1:51 PM, "Ali Jibran" <ali.jibran44@gmail.com <javascript:_e(%7B%7D,'cvml','ali.jibran44@gmail.com');>> wrote:
Hi. I'm using mod_skypeopen and everything's working fine.
The call comes in executes ivr and on DTMF it is forwarded into a
queue. But for some reasons the phone aren't ringing.
I have another trunk attached as a gateway. When I call from that
number the phones rings fine.
Can someone tell me why that is? Thanks.
Is very strange, once inside FS a skype call behaves like a sip call...
Maybe you forgot to set some channel variable? Or something else, pertaining to dialplan/scripting?
If you cannot find the reason, please look in the skypopen confluence (wiki) page how to report a problem, and open a jira. Is important to give us a way to re produce the problem, so remember to read the skypopen page paragraph on how to report a.problem.
-giovanni
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','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 <javascript:_e(%7B%7D,'cvml','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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','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 <javascript:_e(%7B%7D,'cvml','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
-- Ítalo Rossi
Follow the instruction on skypopen wiki/co fluence page on how to report problems, please. sent from my mobile, Giovanni Maruzzelli cell: +39 347 266 56 18 On Apr 9, 2015 1:50 PM, "Ali Jibran" <ali.jibran44@gmail.com> wrote:
Mod_callcenter
I'll post the logs...but as far as I can see is that it says new channel but doesn't go to ring ready when calling from skypeopen.
Where as when calling normally from the trunk the logs show ring ready state. That's the only difference I can see.
On Wednesday, April 8, 2015, Ítalo Rossi <italorossib@gmail.com> wrote:
mod_callcenter or mod_fifo?
Post your debug logs reproducing de problem and your configs
On Wed, Apr 8, 2015 at 2:01 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
sent from my mobile, Giovanni Maruzzelli cell: +39 347 266 56 18
On Apr 8, 2015 1:51 PM, "Ali Jibran" <ali.jibran44@gmail.com> wrote:
Hi. I'm using mod_skypeopen and everything's working fine.
The call comes in executes ivr and on DTMF it is forwarded into a
queue. But for some reasons the phone aren't ringing.
I have another trunk attached as a gateway. When I call from that
number the phones rings fine.
Can someone tell me why that is? Thanks.
Is very strange, once inside FS a skype call behaves like a sip call...
Maybe you forgot to set some channel variable? Or something else, pertaining to dialplan/scripting?
If you cannot find the reason, please look in the skypopen confluence (wiki) page how to report a problem, and open a jira. Is important to give us a way to re produce the problem, so remember to read the skypopen page paragraph on how to report a.problem.
-giovanni
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ítalo Rossi
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
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Ali Jibran
-
Giovanni Maruzzelli
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Ítalo Rossi