MC-

Here is the pastebin: http://pastebin.freeswitch.org/18064

I created small log and recreated my problem.

1. at main ivr, pressed 7, then at conf menu went to static 509 (ivr option 9), and then pressed # to get transfered to 759 which is my dynamic extension

Rodney

From: "freeswitch-users-request@lists.freeswitch.org" <freeswitch-users-request@lists.freeswitch.org>
To: freeswitch-users@lists.freeswitch.org
Sent: Friday, December 23, 2011 2:20 AM
Subject: FreeSWITCH-users Digest, Vol 66, Issue 148

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Today's Topics:

1. Re: transfer to dynamic con from static (Michael Collins)
2. Radio stream binding ? (Sherif Omran)
3. faxing and file permissions (Tim Titov)
4. int/ext dial tones not synchronous (georg@riseup.net)
5. Re: Radio stream binding ? (Sherif Omran)
6. Re: Feedback (jay binks)
7. Re: Ringback on PRI line (Ryan V)

Pastebin the actual console output, including the stuff leading up to when you press #.

-MC

On Thu, Dec 22, 2011 at 3:38 PM, Rodney <notlikeme75@yahoo.com> wrote:

everytime i use transfer from caller controls i get "destination out of order" on console. so i use the execute extension and get problems. is this correct?

<control action="execute_application" digits="#" data="transfer 759 xml default"/>

`rodney

hello guys,

can any body help me to use

bind_meta_app

to connect a channel to a radio stream using gnuradio.org project?

thanks in advance

regards,
Sherif

When faxing with freeswitch the directory and containing file need to be readable by "other". Otherwise, mod_spandsp complains about the file being inaccessible. You can also see on mod_spandsp wiki people doing chmod o+r $TMPFAX. Why can't this be run under the freeswitch user only, so that you can do basic access control?

Hello all,

I made the first experiences with the dialplan, and after some while, I
really got calls from the outside routed to one of my sip phones.

However, my phone is ringing something like 1-3 seconds earlier than the
dial tone the caller is hearing.

I googled around, but found no solution.

As a start, I just used this extension:
<extension name="Local_Extension">
<condition field="destination_number" expression="^(4930XXXXXXXX)$">
<action application="bridge" data="user/12@192.168.1.93"/>
</condition>
</extension>

Thanks,
Georg

Hello all,

does mod_scout play asf streams?

thanks

--- On Fri, 12/23/11, Sherif Omran <sherifomran2000@yahoo.com> wrote:

From: Sherif Omran <sherifomran2000@yahoo.com>
Subject: [Freeswitch-users] Radio stream binding ?
To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org>
Date: Friday, December 23, 2011, 3:00 AM
hello guys,

can any body help me to use
bind_meta_app

to connect a channel to a radio stream using gnuradio.org project?

thanks in
 advance

regards,
Sherif
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Agreed...

put what you have currently on Git Hub so we can check it out...

you sounded like you were already fairly advanced in your progress,

so throwing something that mostly works on Git Hub should only take an hour.

can wait to check it out...

Jay

On 22 December 2011 11:34, Jeff Lenk <jeff@jefflenk.com> wrote:

Chad,

Any update on this? Looking forward to seeing some of your contributions and
perhaps being able to merge them into the mainline code.

Sincerely,
Jeff

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_________________________________________________________________________
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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Sincerely

Jay

Adding following entries into inbound dial plan solved the problem.

        <action application="ring_ready" />
        <action application="set" data="ringback=data=ringback=${in-ring}"/>
        <action application="set" data="instant_ringback=true"/>

Thanks,

Venkatesh K

On Fri, Dec 23, 2011 at 4:50 AM, Michael Collins <msc@freeswitch.org> wrote:

This is probably a PRI config issue. Doublecheck the PRI protocol settings with your provider. Also, get a d-channel trace and drop it on pastebin. (see the freetdm wiki page for d-chan capture instructions.)

I'm sure Moy or one of the guys here will be happy to take a look.

-MC

On Thu, Dec 22, 2011 at 12:06 AM, Ryan V <ryan@kaevee.com> wrote:

Hi,

We are running freeswitch with Sangoma PRI and Analog cards. There is no ring back for calls coming in on PRI line. Calls coming in on FXO lines are fine. Calling party gets ring back.

I have added following to file conf/dialplan/public/00_inbound_did.xml

<action application="set" data="instant_ringback=true"/>
<action application="set" data="ringback=%(2000,4000,440.0,480.0)"/>

Any suggestions to resolve this problem?

Thanks,

Venkatesh K

_________________________________________________________________________
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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http://www.freeswitch.org

_________________________________________________________________________
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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