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<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

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---------- 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 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 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/2a8841ab666ec23ceb5688587bb1...

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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