thanks guys

Avi,  your suggestion makes perfect sense, that's what I'll do

jeff

On Sat, Dec 8, 2012 at 6:19 PM, <freeswitch-users-request@lists.freeswitch.org> wrote:
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Today's Topics:

   1. Problem freeswitch and asterisk mediagateway bridge no direct
      connection (Thomas Hoellriegel)
   2. Re: Problem freeswitch and asterisk mediagateway  bridge no
      direct connection (covici@ccs.covici.com)
   3. xml_curl result not found (Jeff Geddes)
   4. Re: xml_curl result not found (Avi Marcus)
   5. Re: xml_curl result not found (Cal Leeming [Simplicity Media Ltd])
   6. Re: IVR menu delay - 1 second between events
      (Cal Leeming [Simplicity Media Ltd])
   7. recovery_profile_name - any explanation?
      (Cal Leeming [Simplicity Media Ltd])


---------- Forwarded message ----------
From: Thomas Hoellriegel <admin@blindi.net>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Sat, 8 Dec 2012 16:25:31 +0100 (CET)
Subject: [Freeswitch-users] Problem freeswitch and asterisk mediagateway bridge no direct connection
Hi,
i have setup a voicechat (social network  for handycap users).
Fs and asterisk works on the same host:
eth0 mainip fs
eth0:0 asteriskip
I connecting freeswitch (siprouter line managemnt), asterisk (mediagateway),
then i  connect from a sip provider
Fs answer the call, and forward to bridge to asterisk  fine. then i make a direct connection to fs: (no sipprovider)
via sipuri: "dorf@dorf.blindi.net"
fs answer the calls, but the bridge don.t works to asterisk.
the bridge hangup before asterisk answer the call from fs

the problem is only to connect fs from the world. Why works only fine from a sip provider?
must i edit the internal profile to bridge calls to another instance?

Can you help please? thanks.



i connection

---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste für blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc


---------- Forwarded message ----------
From: covici@ccs.covici.com
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Sat, 08 Dec 2012 12:02:11 -0500
Subject: Re: [Freeswitch-users] Problem freeswitch and asterisk mediagateway bridge no direct connection
Why use asterisk at all?  What is it doing?

Thomas Hoellriegel <admin@blindi.net> wrote:

> Hi,
> i have setup a voicechat (social network  for handycap users).
> Fs and asterisk works on the same host:
> eth0 mainip fs
> eth0:0 asteriskip
> I connecting freeswitch (siprouter line managemnt), asterisk
> (mediagateway),
> then i  connect from a sip provider
> Fs answer the call, and forward to bridge to asterisk  fine. then i
> make a direct connection to fs: (no sipprovider)
> via sipuri: "dorf@dorf.blindi.net"
> fs answer the calls, but the bridge don.t works to asterisk.
> the bridge hangup before asterisk answer the call from fs
>
> the problem is only to connect fs from the world. Why works only fine
> from a sip provider?
> must i edit the internal profile to bridge calls to another instance?
>
> Can you help please? thanks.
>
>
>
> i connection
>
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste für blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting@freeswitch.org
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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--
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici@ccs.covici.com




---------- Forwarded message ----------
From: Jeff Geddes <geddes.jeffrey@gmail.com>
To: freeswitch-users@lists.freeswitch.org
Cc: 
Date: Sat, 8 Dec 2012 13:32:45 -0400
Subject: [Freeswitch-users] xml_curl result not found
I'm using xml_curl to generate a dialplan and send back result status="not found" if something throws an error.  I don't want it to fail back to the in-memory dialplan if there is an error.  Is there a way to do this?

thanks
jeff


---------- Forwarded message ----------
From: Avi Marcus <avi@avimarcus.net>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Sat, 8 Dec 2012 20:10:29 +0200
Subject: Re: [Freeswitch-users] xml_curl result not found
Why send a not found? Send a hangup with a failure reason, and that will be the end of processing.

-Avi


On Sat, Dec 8, 2012 at 7:32 PM, Jeff Geddes <geddes.jeffrey@gmail.com> wrote:
I'm using xml_curl to generate a dialplan and send back result status="not found" if something throws an error.  I don't want it to fail back to the in-memory dialplan if there is an error.  Is there a way to do this?

thanks
jeff

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

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

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

FreeSWITCH-users mailing list
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---------- Forwarded message ----------
From: "Cal Leeming [Simplicity Media Ltd]" <cal.leeming@simplicitymedialtd.co.uk>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Sat, 8 Dec 2012 20:48:33 +0000
Subject: Re: [Freeswitch-users] xml_curl result not found
Exclude any dialplan sections from your config, and thus it won't have anything to fall back onto.

However, if any other modules are configured to look for configs (??) then you may need to use the approach Avi mentioned, i.e. sending back a hangup with failure reason.

Cal

On Sat, Dec 8, 2012 at 5:32 PM, Jeff Geddes <geddes.jeffrey@gmail.com> wrote:
I'm using xml_curl to generate a dialplan and send back result status="not found" if something throws an error.  I don't want it to fail back to the in-memory dialplan if there is an error.  Is there a way to do this?

thanks
jeff

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

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.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




---------- Forwarded message ----------
From: "Cal Leeming [Simplicity Media Ltd]" <cal.leeming@simplicitymedialtd.co.uk>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Sat, 8 Dec 2012 21:09:54 +0000
Subject: Re: [Freeswitch-users] IVR menu delay - 1 second between events
I've raised a JIRA ticket for this;
http://jira.freeswitch.org/browse/FS-4924

Cal

On Fri, Dec 7, 2012 at 8:08 PM, Cal Leeming [Simplicity Media Ltd] <cal.leeming@simplicitymedialtd.co.uk> wrote:
Just trawled through the source and found this;

src/switch_ivr_menu.c:
if (status == SWITCH_STATUS_SUCCESS) {
status = switch_ivr_sleep(session, 1000, SWITCH_FALSE, NULL);
}

So, I changed 1000 to 10, the delay has now disappeared and it seems to work perfectly.



---------- Forwarded message ----------
From: "Cal Leeming [Simplicity Media Ltd]" <cal.leeming@simplicitymedialtd.co.uk>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Sat, 8 Dec 2012 22:18:37 +0000
Subject: [Freeswitch-users] recovery_profile_name - any explanation?
Hello,

I've just been trawling Google for an explanation of what exactly the variable 'recovery_profile_name' means, but no luck.

It seems to relate to the following change from 4 months ago;
https://github.com/freeSwitch/freeswitch/commit/2a8841ab666ec23ceb5688587bb14d16bf193b77

The commit message states 'change mod_sofia to use new core based recovery engine'.

But, it is unclear when this variable should be relied upon, and under what circumstances it will be used under.

Any clarifications would be good, I will of course put in time to update the wiki too.

Thanks

Cal

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