[Freeswitch-users] Ringing don't exist

Wed Mar 30 12:51:31 MSD 2016

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 at 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 at 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 at gmail.com>, FreeSWITCH Users Help <
> freeswitch-users at lists.freeswitch.org>
> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
> Can anyone assist me please ? Thank you.
>
> 2016-03-30 8:02 GMT+01:00 amani mansour <amani.mansour2 at 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 at 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 at 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 at lists.freeswitch.org>
>>> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at 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 at 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 at 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 at gmail.com>
>>>> wrote:
>>>>
>>>> yes i know mr Brian ,in fact i asked why ?
>>>>
>>>> 2016-03-29 17:14 GMT+01:00 Brian West <brian at 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 at 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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>>>> http://www.freeswitchsolutions.com
>>>>
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>>>
>>>
>>
>
>
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