- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 47, Issue 164
by guru singh 18 May '10

18 May '10

> ---------- Forwarded message ---------- > From: David Ponzone <david.ponzone(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Tue, 18 May 2010 16:10:02 +0200 > Subject: Re: [Freeswitch-users] Hardware Suggestions. > Like Jeremy, I would recommend Vyatta :) > I am not using that in production yet, but I am considering it for my BGP core routers. > it's possible to install it in a VM to play with it. Thanks, Vyatta looks just appropriate. Also, I think you're right about the Cisco 7940 phones not working with standard PoE appliances. Since, I'm getting refurbished models at a very moderate price, I think the trade-off in buying a Cisco PoE switch is worth it? thanks gs > David Ponzone Direction Technique > email: david.ponzone(a)ipeva.fr > tel: 01 74 03 18 97 > gsm: 06 66 98 76 34 > Service Client IPeva > tel: 0811 46 26 26

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Contact Field
by Ken Fulmer 18 May '10

18 May '10

We are connecting a FS gateway to PaeTec. They need the contact field to be the same as the calling number. Using the trunk registration 10 digit number doesn't work for them. I added the following parameter to remove "gw + external" in the contact field to the left of the @ sign: <param name="extension-in-contact" value="true"/> Is there something else I should add, so the contact field shows the 10 digit calling number? Thanks, Ken Fulmer

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dll sym error
by babak yakhchali 18 May '10

18 May '10

Hi I'm changing one of say modules, but when I try to load, it gives "dll sym error"?

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Fwd: Can you spare 5 minutes and share your thoughts on Open Source PBX systems?
by Michael Jerris 18 May '10

18 May '10

These are the surveys digium uses to quote usage numbers on open source. Feel free to fill this out if you like so they can get accurate numbers of the open source PBX marketplace and FreeSWITCH's real footprint. Mike http://app.en25.com/e/er.aspx?s=491&lid=680&elq=4b1644405c284f6380e806e2130… Begin forwarded message: > From: Digium <webannounce(a)digium.com> > Date: May 18, 2010 9:58:26 AM EDT > Subject: Can you spare 5 minutes and share your thoughts on Open Source PBX systems? > Reply-To: Digium <webannounce(a)digium.com> > > If you are having trouble reading this email, read the online version. > > > > > Dear Mike, > > Digium® is conducting its third annual survey of organizations and individuals that have installed or may install an Open Source PBX. The survey can be completed in under 5-minutes. > > Everyone completing the survey and providing an email address will receive an entry into our drawing for a NEWiPAD, and highlights of the survey results once we finish the survey. > > Please click the link and complete the survey. We thank you for your participation! > > > Take the Survey > Best Regards, > The Digium Team > > > Copyright © Digium, Inc. - The Asterisk Company > 445 Jan Davis Drive NW, Huntsville, AL 35806 > Visit our website: Digium.com | Unsubscribe | Update Subscriptions >

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headset for CELT fullband conferencing
by lands＠freenet.de 18 May '10

18 May '10

Hi, I´m looking for the "best" headset for CELT fullband conferencing. In an old thread the Logitech 350 premium headset was suggested. The Logitech seem to be "only" wideband. What do you use nowadays?

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Setup help for outbound dialing
by Bob Coleman 17 May '10

17 May '10

Hi, My setup is basically as follows: Have a sip profile that allows incoming calls(lets call it internal). Have setup dialplans to accept the inbound ddi's that I want to allow routing to an ivr application. Our server is hosted inside a sip providers network, so we dont use gateways etc, direct sip trunk via internal sofia profile. I am struggling to setup outbound calling. I can make a call through a gateway, but am not sure what to setup to allow an outbound call through our internal profile. eg originate sofia/internal/<phonenumber>@<ipaddress> <extension> Any help would be appreciated. Not sure what other info you would like but just ask Thanks Bob

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mod_ivr with condition
by budi wibowo 17 May '10

17 May '10

dear all i setup ivr successfully following xml sample, i want to add functionality to give condition based on calling number, generally i want this: if calling_id =1000 then do someaction else do some other action can i use xml for this or should use javascript ? regards budi

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Using mod_managed for directory info / DRK presentation
by Phillip Jones 17 May '10

17 May '10

In DRK presentations last conference call, he showed how to hook into the registered user / directory lookup by setting a callback from with in mod_manged code. Can anyone you saw the presentation (or otherwise) tell me how to do this. It looked so easy at the time but now I am stumped. Thanks Phil

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fetch caller profile variables
by Tihomir Culjaga 17 May '10

17 May '10

hello, i need to fetch a caller profile variable (e.g. answer_stamp) from a dialplan, can anyone help? i tried with: <action application="set" data="TEST=answer_stamp"/> <action inline="true" application="set" data="TEST=answer_stamp"/> <action inline="true" application="set" data="TEST=${answer_stamp}"/> but it doesn't help...

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resources path mapping
by budi wibowo 16 May '10

16 May '10

dear all just check /usr/local/freeswitch/conf/lang/en/demo/demo-ivr.xml and i found  <action function="play-file" data="ivr/ivr-to_repeat_these_options.wav"/> <action function="play-file" data="ivr/ivr-please.wav"/> <action function="play-file" data="voicemail/vm-press.wav"/> <action function="play-file" data="digits/9.wav"/> digits/9.wav actually read /usr/local/freeswitch/sounds/en/us/callie/digits/8000/9.wav ivr/ivr-please.wav actually read /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please.wav where's the mapping definition for 2 things above? regards budi wibowo

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