Re: [asterisk-users] SwitchVox and Asterisk
Hello, sorry for not being clear, the application part of this (the voice directory) is already built, mostly working and I have no problem with that. It is based on LumenVox if anyone would like to know, with just a plain XML grammar. I do need to get SwitchVox to send a call to Asterisk/FreePBX, which will in turn call one of SW's extensions. Thanks! Luca On Mon, May 8, 2017 at 9:00 AM, Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Monday 08 May 2017 at 16:43:24, Luca Pradovera wrote: > > > Hello, > > I need to have an extension on a SwitchVox server dial out to one on an > > Asterisk (FreePBX actually) box which will host a voice directory. > > What's a voice directory? > > > The Asterisk server will then need to dial one of the SwitchVox > extensions > > if it gets a voice match. > > You mean, listen to the caller speaking and identify who they are? > > Sounds "non-trivial" to me... > > > Anyone has done that, and could let me know how? So far it looks like IAX > > peering (what SW calls "SwitchVox peering") could work? > > IAX will connect two Asterisk servers and allow them to communicate (it > stands > for Inter Asterisk eXchange) - think of it in the same way as a SIP trunk - > you can have multiple calls to/from multiple numbers going over the link. > > However, are you saying that you've already got the "voice directory" and > "voice match" parts working in Asterisk, and you just need to know how to > dial > between that and SwitchVox? > > Or is the "voice" part of the challenge also something you're looking for > help > with? > > > Antony. > > -- > Numerous psychological studies over the years have demonstrated that the > majority of people genuinely believe they are not like the majority of > people. > >Please reply to the > list; > please *don't* CC > me. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SwitchVox and Asterisk
On Monday 08 May 2017 at 16:43:24, Luca Pradovera wrote: > Hello, > I need to have an extension on a SwitchVox server dial out to one on an > Asterisk (FreePBX actually) box which will host a voice directory. What's a voice directory? > The Asterisk server will then need to dial one of the SwitchVox extensions > if it gets a voice match. You mean, listen to the caller speaking and identify who they are? Sounds "non-trivial" to me... > Anyone has done that, and could let me know how? So far it looks like IAX > peering (what SW calls "SwitchVox peering") could work? IAX will connect two Asterisk servers and allow them to communicate (it stands for Inter Asterisk eXchange) - think of it in the same way as a SIP trunk - you can have multiple calls to/from multiple numbers going over the link. However, are you saying that you've already got the "voice directory" and "voice match" parts working in Asterisk, and you just need to know how to dial between that and SwitchVox? Or is the "voice" part of the challenge also something you're looking for help with? Antony. -- Numerous psychological studies over the years have demonstrated that the majority of people genuinely believe they are not like the majority of people. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switchvox vs Asterisk codebase
I think its 1.4 or 1.2 --- On Sat, 29/5/10, Andrew Joakimsen joakim...@gmail.com wrote: From: Andrew Joakimsen joakim...@gmail.com Subject: [asterisk-users] Switchvox vs Asterisk codebase To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Saturday, 29 May, 2010, 5:56 AM Does anyone know what version of Asterisk Switchvox uses, and if it is modified in any way? FWIW, I am dealing with a provider that claims compatibility with Switchvox but not Asterisk for their SIP trunking service. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switchvox vs Asterisk codebase
On 05/29/2010 01:56 AM, Andrew Joakimsen wrote: Does anyone know what version of Asterisk Switchvox uses, and if it is modified in any way? FWIW, I am dealing with a provider that claims compatibility with Switchvox but not Asterisk for their SIP trunking service. Switchvox 4.x is based on the Asterisk Business Edition C.3 codebase, which itself is Asterisk 1.4 plus a number of backports from later versions of open source Asterisk. In addition, it contains some Switchvox-specific changes, but I don't believe there's anything that would affect SIP compatibility or interoperability in any material way. However, some providers will only certify 'known' releases of software, so it's easier for them to do that with something like Switchvox, since if the customer has Switchvox 4.5 (for example), that's a known quantity. If the customer has 'Asterisk', or even 'Asterisk 1.4', that could be one of many different versions, and could potentially have significant patches applied... which makes it more difficult for the provider to be comfortable that it will 'just work'. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
