Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Strongly suggest to consider a Freeswitch/OpenSER implementation instead. Regarding purpose built and supported software.sometimes throwning billions of CMM software development to a product does not guarantee a good product... look at Micro$oft Vista. E http://Gpro.ws ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking to replicate OnSIP ........SER + Asterisk
Hello Alex, We have a customer looking to replicate OnSIP using OpenSER/Asterisk or FreeSwitch. Can you provide us a quote on the cost to completely replicate OnSIP? Thanks in advance, Ed Direct: 678.522.8511 Mail: edpimentl[at]gmail.com] Voip/IM: edpimentl [SKype | GoogleTalk ] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Network Monitoring
http://www.voip-info.org/wiki/view/Asterisk+monitoring ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A Suggestion To Asterisk Appliance Developers
Please google VoIP2.0 apps... this is old old news... even Cisco has marketed this going back to 2001. -E ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Magnetic door locks
Here are other solutions http://www.abptech.com/blog/open-doors-with-sip/ http://www.netgenium.co.uk/documents/ip_lock_controller.html http://www.premierelect.com/10.cfm?prodCode=1031&category=88 -E http://mobiquity.ws http://gpro.ws ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk + web services
Try Adhersion and or Telegraph -E http://mobiquity.ws ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Centile ipbx, anyone heard of this?
They have been around for over 8 years, and their HQ is now in France... -E ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Installation with Radius Support
Here is info on Asterisk and Radius http://www.voip-info.org/wiki/view/CW+Radius++for+Asterisk -E ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Browser based VoIP client?
Have you seen these client? http://www.mozillavoip.com/ http://tringme.com/ http://www.twoiplink.com/ http://www.openwengo.org/index.php/openwengo/public/homePage/openwengo/public/projectsFirefox (Dated, since project changed names) BTW, you can also trying to roll your own using old OpenWengo... code. that was what did. -E On Wed, Jun 4, 2008 at 2:42 PM, Hilary Miller <[EMAIL PROTECTED]> wrote: > Something that I can put on our internal company website to replace > our hardware IP phones. > > I see many web 2.0 startups offering browser based clients for their > own service, but I can't seem to find anything that I can use with my > own PBX. Do I suck at searching google or has the future not arrived > yet? > > Thanks for reading! > -- > Just Hil > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Call Center Agents and Asterisk?
Hello Steve, You are right on track and this is also what we have done with pretty good results. Of course now with Flex/Air there are a number of ways to enhance the service for the Customer/Agent Ed Mail: edpimentl[at]gmail.com Voip: edpimentl [SKype | GoogleTalk ] http://agileoss.com (Web2.0 and SOA Development ) http://mobiquity.ws (Private Label Social Network) http://youbiquity.ws (Power of One for all Social Networks) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway behind SIP ATA?
Have you looked into
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html
-E
On Jan 4, 2008 8:43 AM, Remco Barendse <[EMAIL PROTECTED]> wrote:
> >
> > You can use the D option with the Dial command.
> > Something like this should work:
> > exten => _06,1,Dial(SIP/gsm_gateway,45,D(${EXTEN})
>
>
> It worked
>
> Here is how i did it in FreePBX :
>
> 1) Setup a SIP extension for the ATA device, in my case i give it
> extension number 298. Edit the extension after creating it set DISALLOW to
> all and set ALLOW to alaw to make sure DTMF sending will work.
>
> 2) Create a custom trunk, and set as Custom Dial String :
> Local/[EMAIL PROTECTED]
>
> 3) add to extensions_custom.conf :
> [custom-gsmvoip-out]
> exten => _.,1,Dial(SIP/298,,D(ww0${EXTEN}))
>
> Note that i put a leading zero there, because for my fallback outbound
> routes i needed to strip the leading zero so i added it again here.
>
> 4) Insert the custom trunk in outbound routes
>
> That's it
>
> Hope this will save somebody else 2 days of frustration :)))
>
> Cheers!
>
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Re: [asterisk-users] Changing audio message to text message
Yes, it is call http://www.talktext.com/ -E http://mobiquity.ws http://datr.ws ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech Rec on Voicemail
Just for starter, look at CallWave, and Jott. -E ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
Why not use
1-Ruby RAGI
2-http://adhearsion.com/
or similar tools which overcome Asterisk dial plan limitations?
-E
On 8/8/07, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
>
> On Wednesday 08 August 2007 1:39:34 pm Mike wrote:
> > exten => 12345,1,AGI(agi-helloworld.agi)
>
> AGI is an application, and you've called it.
>
> > exten => 12345,1,Noop(${AGI(agi-helloworld.agi)})
>
> AGI is not a function. You cannot "nest" applications like that. The
> NoOp
> application cannot call another application.
>
> -A.
>
>
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Re: [asterisk-users] Asterisk 1.4.9.tar.gz download fails
THANKS!!! I was looking for 1.4.9 Very much appreciated. -E ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.9.tar.gz download fails
Hello Fellow Asterisk Mailing ListMembers, When I tried to download the latest version of Asterisk this is what I get: http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed Where are all the latest Asterisk 1.4.x source files? Thanks in advance, -E ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] installing * from source
Have you also consider adding adding the uBuntu steps in addition to CentOS? -E On 7/8/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: On Sun, Jul 08, 2007 at 10:05:37AM -0400, Baji Panchumarti wrote: >On 7/8/07, Dovid B wrote: > > > > > [...] > > > > I was actually thinking of creating a script that you download and it preps > > your system for an asterisk install and it does everything for you. It can > > also have an option to run as a cron job and update nightly. The issue is > > that you cant just update some ones phone system if they are using it. So > > you would need like and email or sms sent to the user telling him to run the > > update script. What do others think of this idea ? > > > Dovid, > > I am not sure about an update script due to reasons that Tzafrir > and you already pointed out. > > But I think it would be GREAT to have an initial install script that > just works, period ! > > For years installing/updating LAMP (apache, PHP & MySQL on Linux) > was a manual process, I read somewhere that Ubuntu now has a > script that does the whole thing for you, and does it correctly. Installing LAMP on Linux in the recent years has been something of the sort of: apt-get install apache php mysql (different package managers, different package names. Those are not even the actual package names in Debian). This provides you with an installation that you can easily upgrade on the next apache/php security hole. Any server distribution worth its salt has those (and let's not get into a distro fight in here) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sugar Auto-Dial with Asterisk?
Try vTiger -E On 6/14/07, Matt <[EMAIL PROTECTED]> wrote: I see that module, but it does not work with the current version of Sugar. Does anyone have a solution that works with the current version of Sugar? On 6/14/07, Nuria Fernandez <[EMAIL PROTECTED]> wrote: > > Exist a module VoiceRD to do that. > JuntaDeAndalucia_es_sf_diphone > > 2007/6/14, Matt < [EMAIL PROTECTED]>: > > > > Before I go and start coding is anyone aware of an auto-dialer > > plugin for Sugar CRM that will allow me to click a button when I'm in > > someone's account and have my phone ring and then connect me to them? > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Ed Pimentel AgileCO Founder Mail: edpimentl[at]gmail.com Mail2: edpimentl[at]ieee.org IM: edpimentl [AOL | Jabber | Yahoo | MSN ] Voip: edpimentl [SKype | GoogleTalk ] Mobile Content Marketing/Management/Digital Delivery http://mobilecentral.ws Mobile ( Context Aware, AmbientIntelligence, Location ) based Social Network http://TagR.mobi (Alpha) Mobile Payment - P2P Payment http://goowallet.ws http://agilepay.ws [S4]Secure Scalable Streaming Storage GridService http://DatR.ws Sponsor of P2PSIP open source [viasip_ng] project Based on IETF P2PSIP WG https://sourceforge.net/projects/viasip/ http://groups.google.com/group/viasip_ng ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Divitas
There will be a number of companies set to offer similar services. In 3 months we will have a 24 port SIP<->GSM<->SKYPE gateway -E On 5/27/07, Dean Collins <[EMAIL PROTECTED]> wrote: I was cleaning through some old IT magazines this long weekend when I came across a company called Divitas in the April 30th edition of Network Computing. I've never heard of them but has anyone else heard of them? Basically they have a call control appliance that can deliver centrally held up calls between not only GSM but also redirect the call to a wifi hotspot if you are in range. It seems like a neat concept that shouldn't necessarily be beyond the capabilities of Asterisk (apart from the fact that the end Win Mobile 5 / Symbian handset would need some type of client). Any thoughts? At $550 per seat looks an expensive way to transfer calls between networks but I've never seen another CPE piece of equipment that can do this. http://www.divitas.com/products Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). [image: Call Button]<http://click.mexuar.com/webuser/click/7/userurl/Cognation><http://click.mexuar.com/webuser/nojs/7/userurl/Cognation> ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ed Mail: edpimentl[at]gmail.com Mail2: edpimentl[at]ieee.org IM: edpimentl [AOL | Jabber | Yahoo | MSN ] Voip: edpimentl [SKype | GoogleTalk ] Mobile Content Marketing/Management/Digital Delivery http://mobilecentral.ws Mobile ( Context Aware, AmbientIntelligence, Location ) based Social Network http://TagR.mobi (Alpha) Mobile Payment - P2P Payment http://agilepay.ws [S4]Secure Scalable Streaming Storage GridService http://DatR.ws Sponsor of P2PSIP open source [viasip_ng] project Based on IETF P2PSIP WG https://sourceforge.net/projects/viasip/ http://groups.google.com/group/viasip_ng ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk High-Capacity Stability
Actually, OpenSER is just the you will need to scale Asterisk. We have perform a number of OpenSER to Asterisk implementation for 50k plus users -E On 5/14/07, Atlanticnynex <[EMAIL PROTECTED]> wrote: Thanks for all the input guys. This is what I had originally expected. Does anyone have any recommendations for other software configurations? I've thought about using OpenSER + rtpproxy(or media proxy), but it seems that OpenSER is not designed to do this sort of thing and would require some tricky hacking(?). I guess I'm wondering if their are any other opensource B2BUA-like softswitches that would fit what I'm looking for. What are these VoIP carriers using? Thanks, kn0x -- Thanks in advance and best regards, Ed Pimentel AgileCO Founder Web: http://AgileCO.net Mail: edpimentl[at]gmail.com Mail2: edpimentl[at]ieee.org IM: edpimentl [AOL | Jabber | Yahoo | MSN ] Voip: edpimentl [SKype | GoogleTalk ] Mobile Content Marketing/Management/Digital Delivery http://mobilecentral.ws Mobile ( Context Aware, AmbientIntelligence, Location ) based Social Network http://TagR.mobi (Alpha) Mobile Payment - P2P Payment http://agilepay.ws [S4]Secure Scalable Streaming Storage GridService http://DatR.ws Private Label Social Networks http://GooGaYa.com Sponsor of P2PSIP open source [viasip_ng] project Based on IETF P2PSIP WG https://sourceforge.net/projects/viasip/ http://groups.google.com/group/viasip_ng ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
