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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