Re: [Freeswitch-users] Ringing don't exist
The only way you will get 180/183 is that there is an endpoint that take time/ or ring on the other side, but what I see it is like an Internal IVR calling with immediate response. my question why do you need to see 180/183, if so you can defined another soft phone and make it rings, what is you business case. From: amani mansour <amani.mansour2@gmail.com <mailto:amani.mansour2@gmail.com>> Subject: Re: [Freeswitch-users] Ringing don't exist Date: March 30, 2016 at 10:59:50 AM GMT+3 To: Ahmed habiba <ahabiba@gmail.com <mailto:ahabiba@gmail.com>>, FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Reply-To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Can anyone assist me please ? Thank you. 2016-03-30 8:02 GMT+01:00 amani mansour <amani.mansour2@gmail.com <mailto:amani.mansour2@gmail.com>>: Good morning Mr Ahmed , Thank you sir , i modifie the ring back but no change in the tcpdumps but I think (probably sure) that the problem is in the bridge : i don't know what i must set in data = user/ ... or sofia/gateway/... remarque that the extension 600 is registred when i added an external gateway like this : sip-sofia/external <include> <gateway name="gateway_to_600.xml"> <param name="proxy" value="192.168.3.5"/> <param name="username" value="600"/> <param name="password" value="$${default_password}"/> <param name="register" value="true"/> </gateway> </include> 2016-03-30 6:44 GMT+01:00 Ahmed habiba <ahabiba@gmail.com <mailto:ahabiba@gmail.com>>: Hi, the configuration you did will not generate any 180/183, as per your configuration you answer immediately once you get invite, if you are trying something like ring-back-tone you can try the below, this will allow some music to play while the other phone is ringing <action application="set" data="ringback=[data]"/> Thanks, Ahmed Habiba. From: amani mansour <amani.mansour2@gmail.com <mailto:amani.mansour2@gmail.com>> Subject: Re: [Freeswitch-users] Ringing don't exist Date: March 30, 2016 at 12:12:43 AM GMT+3 To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Reply-To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Hi Mr Bote the problem is not in the number look when i do in fs_cli : sofia status internal reg ==> i have 2 numbers registred : 1000 and 1011 but only 1000 is configured by softphone 1011 : i did it using external gateway when i dial 1011 from the softphene where registred the number 1000 ,, so the number 1000 is registred in the FS NOT IN THE SOFT PHONE and now i want to call the number 1000 only when it ring (180) or the method progress sesiion arrive i listen a short music example titris music now i receive the all methodes done well also i listen the music but i dont receive the method nor 180 nor 183 can you help me please with regards amani 2016-03-29 21:56 GMT+01:00 Bote Man <bote_radio@botecomm.com <mailto:bote_radio@botecomm.com>>: Any device or soft phone that you want to register to FS should have its own entry in conf/directory/default/*.xml This contains its authentication ID and password to register it to FreeSWITCH. Then you need an entry in conf/dialplan/default.xml or other dialplan that tells FS how to route that number. The Vanilla configuration files available for FS come with directory files for users 1000.xml through 1019.xml and a default dialplan entry that calls 1000-1019. So it will not know how to route 600 nor 423 until you add routes for those numbers in the dialplan. Or you could change 423 to 1001 and enter its credentials into 1001.xml and it should call both directions. --- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence <http://freeswitch.org/confluence> From: amani mansour Sent: Tuesday, 29 March, 2016 16:04 Subject: Re: [Freeswitch-users] Ringing don't exist Mr i have : 1 FS A(virtual machine 1)--------------------------------------------->in my conputer i installed a soft phone ( number 1000 is regestred from the soft phone ) i did a gateway which permet to register another number 423 in freeswitch A so now i have 2 numbers registred in FS but only one is regestred by the soft phone i want to dial 423 from the soft phone 1000 normally the call will be done normally ) but i don't have the methode ring i wish that i have well describe the problem t thank you sir ,i wish you can help me best regards amani 2016-03-29 17:58 GMT+01:00 Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>>: What device is not sending the 180? why is it not sending it? On Mar 29, 2016, at 12:19 PM, amani mansour <amani.mansour2@gmail.com <mailto:amani.mansour2@gmail.com>> wrote: yes i know mr Brian ,in fact i asked why ? 2016-03-29 17:14 GMT+01:00 Brian West <brian@freeswitch.org <mailto:brian@freeswitch.org>>: There is no 180/183 in there, just a 200OK /b On Tue, Mar 29, 2016 at 10:59 AM, amani mansour <amani.mansour2@gmail.com <mailto:amani.mansour2@gmail.com>> wrote: Hi, I need to make a call from my soft phone to another freeswitch ,when the call is established (i meant (INVITE ,OK , RINGING or Progress.dession .) and after the ring is done to the freeswitch i will listen for example a song ,for that i make an external gateway ,but the step "ringing or progress session " is not estableshed i need to call the extension 600 <extension 600.PNG> <tcpdump.PNG> thanks with regards amani _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>
in my application i need to test a call for example in a distant FS so from my FS server through gateway that i configure i will configure and register a test number in the other FS what i need to see that the call is estableshed well ( so i would like to see the progress session) after 2 or 3 second i pass to listen a music [image: Images intégrées 2] I don't know why extension 600 to 600 is canceled [image: Images intégrées 1] 2016-03-30 9:40 GMT+01:00 Ahmed habiba <ahabiba@gmail.com>:
The only way you will get 180/183 is that there is an endpoint that take time/ or ring on the other side, but what I see it is like an Internal IVR calling with immediate response.
my question why do you need to see 180/183, if so you can defined another soft phone and make it rings, what is you business case.
*From: *amani mansour <amani.mansour2@gmail.com> *Subject: **Re: [Freeswitch-users] Ringing don't exist* *Date: *March 30, 2016 at 10:59:50 AM GMT+3 *To: *Ahmed habiba <ahabiba@gmail.com>, FreeSWITCH Users Help < freeswitch-users@lists.freeswitch.org> *Reply-To: *FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Can anyone assist me please ? Thank you.
2016-03-30 8:02 GMT+01:00 amani mansour <amani.mansour2@gmail.com>:
Good morning Mr Ahmed , Thank you sir , i modifie the ring back but no change in the tcpdumps
but I think (probably sure) that the problem is in the bridge : i don't know what i must set in data = user/ ... or sofia/gateway/... remarque that the extension 600 is registred when i added an external gateway like this : sip-sofia/external <include> <gateway name="gateway_to_600.xml"> <param name="proxy" value="192.168.3.5"/> <param name="username" value="600"/> <param name="password" value="$${default_password}"/> <param name="register" value="true"/> </gateway> </include>
2016-03-30 6:44 GMT+01:00 Ahmed habiba <ahabiba@gmail.com>:
Hi,
the configuration you did will not generate any 180/183, as per your configuration you answer immediately once you get invite, if you are trying something like ring-back-tone you can try the below, this will allow some music to play while the other phone is ringing
<action application="set" data="ringback=[data]"/>
Thanks,
Ahmed Habiba.
*From: *amani mansour <amani.mansour2@gmail.com> *Subject: **Re: [Freeswitch-users] Ringing don't exist* *Date: *March 30, 2016 at 12:12:43 AM GMT+3 *To: *FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Reply-To: *FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org
Hi Mr Bote the problem is not in the number look when i do in fs_cli : sofia status internal reg ==> i have 2 numbers registred : 1000 and 1011 but only 1000 is configured by softphone 1011 : i did it using external gateway when i dial 1011 from the softphene where registred the number 1000 ,, so the number 1000 is registred in the FS NOT IN THE SOFT PHONE and now i want to call the number 1000 only when it ring (180) or the method progress sesiion arrive i listen a short music example titris music now i receive the all methodes done well also i listen the music but i dont receive the method nor 180 nor 183
can you help me please
with regards amani
2016-03-29 21:56 GMT+01:00 Bote Man <bote_radio@botecomm.com>:
Any device or soft phone that you want to register to FS should have its own entry in conf/directory/default/*.xml This contains its authentication ID and password to register it to FreeSWITCH.
Then you need an entry in conf/dialplan/default.xml or other dialplan that tells FS how to route that number.
The Vanilla configuration files available for FS come with directory files for users 1000.xml through 1019.xml and a default dialplan entry that calls 1000-1019. So it will not know how to route 600 nor 423 until you add routes for those numbers in the dialplan.
Or you could change 423 to 1001 and enter its credentials into 1001.xml and it should call both directions.
--- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence
*From:* amani mansour *Sent:* Tuesday, 29 March, 2016 16:04 *Subject:* Re: [Freeswitch-users] Ringing don't exist
Mr i have : 1 FS A(virtual machine 1)--------------------------------------------->in my conputer i installed a soft phone ( number 1000 is regestred from the soft phone )
i did a gateway which permet to register another number 423 in freeswitch A
so now i have 2 numbers registred in FS but only one is regestred by the soft phone
i want to dial 423 from the soft phone 1000 normally the call will be done normally ) but i don't have the methode ring
i wish that i have well describe the problem t
thank you sir ,i wish you can help me
best regards amani
2016-03-29 17:58 GMT+01:00 Michael Jerris <mike@jerris.com>: What device is not sending the 180? why is it not sending it?
On Mar 29, 2016, at 12:19 PM, amani mansour <amani.mansour2@gmail.com> wrote:
yes i know mr Brian ,in fact i asked why ?
2016-03-29 17:14 GMT+01:00 Brian West <brian@freeswitch.org>: There is no 180/183 in there, just a 200OK
/b
On Tue, Mar 29, 2016 at 10:59 AM, amani mansour < amani.mansour2@gmail.com> wrote:
Hi, I need to make a call from my soft phone to another freeswitch ,when the call is established (i meant (INVITE ,OK , RINGING or Progress.dession .) and after the ring is done to the freeswitch i will listen for example a song ,for that i make an external gateway ,but the step "ringing or progress session " is not estableshed
i need to call the extension 600 <extension 600.PNG> <tcpdump.PNG>
thanks with regards amani
_________________________________________________________________________ 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
_______________________________________________ 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
Why don't u think of below option [softphone1]<--->[freeswitch1]<---IMT--->[freeswitch2]<---->[softphone2] IMT is the way to connect two freeswitch boxes together https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/396570... The above scenario will allow you to test rel call scenario between two phones on different FS instances On Wednesday, March 30, 2016, amani mansour <amani.mansour2@gmail.com> wrote:
in my application i need to test a call for example in a distant FS so from my FS server through gateway that i configure i will configure and register a test number in the other FS what i need to see that the call is estableshed well ( so i would like to see the progress session) after 2 or 3 second i pass to listen a music
[image: Images intégrées 2]
I don't know why extension 600 to 600 is canceled [image: Images intégrées 1]
2016-03-30 9:40 GMT+01:00 Ahmed habiba <ahabiba@gmail.com <javascript:_e(%7B%7D,'cvml','ahabiba@gmail.com');>>:
The only way you will get 180/183 is that there is an endpoint that take time/ or ring on the other side, but what I see it is like an Internal IVR calling with immediate response.
my question why do you need to see 180/183, if so you can defined another soft phone and make it rings, what is you business case.
*From: *amani mansour <amani.mansour2@gmail.com <javascript:_e(%7B%7D,'cvml','amani.mansour2@gmail.com');>> *Subject: **Re: [Freeswitch-users] Ringing don't exist* *Date: *March 30, 2016 at 10:59:50 AM GMT+3 *To: *Ahmed habiba <ahabiba@gmail.com <javascript:_e(%7B%7D,'cvml','ahabiba@gmail.com');>>, FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','freeswitch-users@lists.freeswitch.org');>> *Reply-To: *FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','freeswitch-users@lists.freeswitch.org');>>
Can anyone assist me please ? Thank you.
2016-03-30 8:02 GMT+01:00 amani mansour <amani.mansour2@gmail.com <javascript:_e(%7B%7D,'cvml','amani.mansour2@gmail.com');>>:
Good morning Mr Ahmed , Thank you sir , i modifie the ring back but no change in the tcpdumps
but I think (probably sure) that the problem is in the bridge : i don't know what i must set in data = user/ ... or sofia/gateway/... remarque that the extension 600 is registred when i added an external gateway like this : sip-sofia/external <include> <gateway name="gateway_to_600.xml"> <param name="proxy" value="192.168.3.5"/> <param name="username" value="600"/> <param name="password" value="$${default_password}"/> <param name="register" value="true"/> </gateway> </include>
2016-03-30 6:44 GMT+01:00 Ahmed habiba <ahabiba@gmail.com <javascript:_e(%7B%7D,'cvml','ahabiba@gmail.com');>>:
Hi,
the configuration you did will not generate any 180/183, as per your configuration you answer immediately once you get invite, if you are trying something like ring-back-tone you can try the below, this will allow some music to play while the other phone is ringing
<action application="set" data="ringback=[data]"/>
Thanks,
Ahmed Habiba.
*From: *amani mansour <amani.mansour2@gmail.com <javascript:_e(%7B%7D,'cvml','amani.mansour2@gmail.com');>> *Subject: **Re: [Freeswitch-users] Ringing don't exist* *Date: *March 30, 2016 at 12:12:43 AM GMT+3 *To: *FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','freeswitch-users@lists.freeswitch.org');>
*Reply-To: *FreeSWITCH Users Help < freeswitch-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','freeswitch-users@lists.freeswitch.org');>
Hi Mr Bote the problem is not in the number look when i do in fs_cli : sofia status internal reg ==> i have 2 numbers registred : 1000 and 1011 but only 1000 is configured by softphone 1011 : i did it using external gateway when i dial 1011 from the softphene where registred the number 1000 ,, so the number 1000 is registred in the FS NOT IN THE SOFT PHONE and now i want to call the number 1000 only when it ring (180) or the method progress sesiion arrive i listen a short music example titris music now i receive the all methodes done well also i listen the music but i dont receive the method nor 180 nor 183
can you help me please
with regards amani
2016-03-29 21:56 GMT+01:00 Bote Man <bote_radio@botecomm.com <javascript:_e(%7B%7D,'cvml','bote_radio@botecomm.com');>>:
Any device or soft phone that you want to register to FS should have its own entry in conf/directory/default/*.xml This contains its authentication ID and password to register it to FreeSWITCH.
Then you need an entry in conf/dialplan/default.xml or other dialplan that tells FS how to route that number.
The Vanilla configuration files available for FS come with directory files for users 1000.xml through 1019.xml and a default dialplan entry that calls 1000-1019. So it will not know how to route 600 nor 423 until you add routes for those numbers in the dialplan.
Or you could change 423 to 1001 and enter its credentials into 1001.xml and it should call both directions.
--- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence
*From:* amani mansour *Sent:* Tuesday, 29 March, 2016 16:04 *Subject:* Re: [Freeswitch-users] Ringing don't exist
Mr i have : 1 FS A(virtual machine 1)--------------------------------------------->in my conputer i installed a soft phone ( number 1000 is regestred from the soft phone )
i did a gateway which permet to register another number 423 in freeswitch A
so now i have 2 numbers registred in FS but only one is regestred by the soft phone
i want to dial 423 from the soft phone 1000 normally the call will be done normally ) but i don't have the methode ring
i wish that i have well describe the problem t
thank you sir ,i wish you can help me
best regards amani
2016-03-29 17:58 GMT+01:00 Michael Jerris <mike@jerris.com <javascript:_e(%7B%7D,'cvml','mike@jerris.com');>>: What device is not sending the 180? why is it not sending it?
On Mar 29, 2016, at 12:19 PM, amani mansour <amani.mansour2@gmail.com <javascript:_e(%7B%7D,'cvml','amani.mansour2@gmail.com');>> wrote:
yes i know mr Brian ,in fact i asked why ?
2016-03-29 17:14 GMT+01:00 Brian West <brian@freeswitch.org <javascript:_e(%7B%7D,'cvml','brian@freeswitch.org');>>: There is no 180/183 in there, just a 200OK
/b
On Tue, Mar 29, 2016 at 10:59 AM, amani mansour < amani.mansour2@gmail.com <javascript:_e(%7B%7D,'cvml','amani.mansour2@gmail.com');>> wrote:
Hi, I need to make a call from my soft phone to another freeswitch ,when the call is established (i meant (INVITE ,OK , RINGING or Progress.dession .) and after the ring is done to the freeswitch i will listen for example a song ,for that i make an external gateway ,but the step "ringing or progress session " is not estableshed
i need to call the extension 600 <extension 600.PNG> <tcpdump.PNG>
thanks with regards amani
_________________________________________________________________________ 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
_______________________________________________ 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
-- Thanks and Best Regards, Ahmed Habiba Mob: +20 10 37 82 970 *Success: believe (Vision) plus commitment (Action)*
thank you sir ,i do it :) with regards amani 2016-03-30 10:46 GMT+01:00 Ahmed habiba <ahabiba@gmail.com>:
Why don't u think of below option
[softphone1]<--->[freeswitch1]<---IMT--->[freeswitch2]<---->[softphone2]
IMT is the way to connect two freeswitch boxes together
https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/396570...
The above scenario will allow you to test rel call scenario between two phones on different FS instances
On Wednesday, March 30, 2016, amani mansour <amani.mansour2@gmail.com> wrote:
in my application i need to test a call for example in a distant FS so from my FS server through gateway that i configure i will configure and register a test number in the other FS what i need to see that the call is estableshed well ( so i would like to see the progress session) after 2 or 3 second i pass to listen a music
[image: Images intégrées 2]
I don't know why extension 600 to 600 is canceled [image: Images intégrées 1]
2016-03-30 9:40 GMT+01:00 Ahmed habiba <ahabiba@gmail.com>:
The only way you will get 180/183 is that there is an endpoint that take time/ or ring on the other side, but what I see it is like an Internal IVR calling with immediate response.
my question why do you need to see 180/183, if so you can defined another soft phone and make it rings, what is you business case.
*From: *amani mansour <amani.mansour2@gmail.com> *Subject: **Re: [Freeswitch-users] Ringing don't exist* *Date: *March 30, 2016 at 10:59:50 AM GMT+3 *To: *Ahmed habiba <ahabiba@gmail.com>, FreeSWITCH Users Help < freeswitch-users@lists.freeswitch.org> *Reply-To: *FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org
Can anyone assist me please ? Thank you.
2016-03-30 8:02 GMT+01:00 amani mansour <amani.mansour2@gmail.com>:
Good morning Mr Ahmed , Thank you sir , i modifie the ring back but no change in the tcpdumps
but I think (probably sure) that the problem is in the bridge : i don't know what i must set in data = user/ ... or sofia/gateway/... remarque that the extension 600 is registred when i added an external gateway like this : sip-sofia/external <include> <gateway name="gateway_to_600.xml"> <param name="proxy" value="192.168.3.5"/> <param name="username" value="600"/> <param name="password" value="$${default_password}"/> <param name="register" value="true"/> </gateway> </include>
2016-03-30 6:44 GMT+01:00 Ahmed habiba <ahabiba@gmail.com>:
Hi,
the configuration you did will not generate any 180/183, as per your configuration you answer immediately once you get invite, if you are trying something like ring-back-tone you can try the below, this will allow some music to play while the other phone is ringing
<action application="set" data="ringback=[data]"/>
Thanks,
Ahmed Habiba.
*From: *amani mansour <amani.mansour2@gmail.com> *Subject: **Re: [Freeswitch-users] Ringing don't exist* *Date: *March 30, 2016 at 12:12:43 AM GMT+3 *To: *FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Reply-To: *FreeSWITCH Users Help < freeswitch-users@lists.freeswitch.org>
Hi Mr Bote the problem is not in the number look when i do in fs_cli : sofia status internal reg ==> i have 2 numbers registred : 1000 and 1011 but only 1000 is configured by softphone 1011 : i did it using external gateway when i dial 1011 from the softphene where registred the number 1000 ,, so the number 1000 is registred in the FS NOT IN THE SOFT PHONE and now i want to call the number 1000 only when it ring (180) or the method progress sesiion arrive i listen a short music example titris music now i receive the all methodes done well also i listen the music but i dont receive the method nor 180 nor 183
can you help me please
with regards amani
2016-03-29 21:56 GMT+01:00 Bote Man <bote_radio@botecomm.com>:
Any device or soft phone that you want to register to FS should have its own entry in conf/directory/default/*.xml This contains its authentication ID and password to register it to FreeSWITCH.
Then you need an entry in conf/dialplan/default.xml or other dialplan that tells FS how to route that number.
The Vanilla configuration files available for FS come with directory files for users 1000.xml through 1019.xml and a default dialplan entry that calls 1000-1019. So it will not know how to route 600 nor 423 until you add routes for those numbers in the dialplan.
Or you could change 423 to 1001 and enter its credentials into 1001.xml and it should call both directions.
--- Bote FreeSWITCH Docs Janitor http://freeswitch.org/confluence
*From:* amani mansour *Sent:* Tuesday, 29 March, 2016 16:04 *Subject:* Re: [Freeswitch-users] Ringing don't exist
Mr i have : 1 FS A(virtual machine 1)--------------------------------------------->in my conputer i installed a soft phone ( number 1000 is regestred from the soft phone )
i did a gateway which permet to register another number 423 in freeswitch A
so now i have 2 numbers registred in FS but only one is regestred by the soft phone
i want to dial 423 from the soft phone 1000 normally the call will be done normally ) but i don't have the methode ring
i wish that i have well describe the problem t
thank you sir ,i wish you can help me
best regards amani
2016-03-29 17:58 GMT+01:00 Michael Jerris <mike@jerris.com>: What device is not sending the 180? why is it not sending it?
On Mar 29, 2016, at 12:19 PM, amani mansour <amani.mansour2@gmail.com> wrote:
yes i know mr Brian ,in fact i asked why ?
2016-03-29 17:14 GMT+01:00 Brian West <brian@freeswitch.org>: There is no 180/183 in there, just a 200OK
/b
On Tue, Mar 29, 2016 at 10:59 AM, amani mansour < amani.mansour2@gmail.com> wrote:
Hi, I need to make a call from my soft phone to another freeswitch ,when the call is established (i meant (INVITE ,OK , RINGING or Progress.dession .) and after the ring is done to the freeswitch i will listen for example a song ,for that i make an external gateway ,but the step "ringing or progress session " is not estableshed
i need to call the extension 600 <extension 600.PNG> <tcpdump.PNG>
thanks with regards amani
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*Success: believe (Vision) plus commitment (Action)*
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participants (2)
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Ahmed habiba
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amani mansour