[Asterisk-Users] Need someone to write a console application for us.
WARNING: This has been cross-posted in the Asterisk-BIZ mailing list. Hello, we are looking for a unix programmer who can write the following type of application that works with asterisk. 1. Operator Console. When a call comes in on a did, the did should be looked up in a database(against a list of did's assigned to companies) and the name of the "called company" should be looked up. If a flag on the DID entry in the database is to forward to the operator, then the name of the called company is displayed on the operator console, ie "FooCompany is being called on line xxx-xxx-". If not, the call should be forwarded to the IVR/Context for the company.(the context will be looked up in a database table as well). The operator console should be able to answer, park, transfer and hang up calls it recieves. Additional calls(up to 3 at a time) should be able to be displayed and answered. 2. Asterisk will need to be configured to work with the above application. We don't know the best way to do this. Probably some sort of forwarding of the call to the "operator extension" and agi. We have a small initial budget($800.00) which I know is very low for something like this, but we would like to work with someone to do some sort of payment or licensing plan for the application with eventual buyout. The program can be in any language, ncurses/c, perl-ncurses, php-ncurses anything you like just as long as it works and is somewhat fast. I would write this myself, but I don't have the time since we are nearing deployment of all of the actual infrastructure for our showplace client. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] follow me configuration web page??
Does anybody have an example follow-me configuration web page code written in either php or perl that can write out the follow-me config into the asterisk files? I'd like to setup something on our office voip server that I can change as needed via a web page rather than writing the script by hand. If no one has one handy, I may write one. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and CompactPCI boards??
Are there any compactPCI boards that work with Asterisk? Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] XORCOM RAPID Asterisk - Suggestions?
We like it alot. It makes rapid deployment of asterisk boxes a breeze. Brent On 8/17/05, Sharadindu Mohanty <[EMAIL PROTECTED]> wrote: > Hey Guys, > Wanted a Suggestion..Howz this Xorcom Asterisk?I am using it and till now > its fine as currently it is in testing stage with 3-4 users. > > Any Ideas??? > > Thanks > > Sharadindu Mohanty > > > How much free photo storage do you get? Store your holiday snaps for FREE > with Yahoo! Photos. Get Yahoo! Photos > How much free photo storage do you get? Store your holiday snaps for FREE > with Yahoo! Photos. Get Yahoo! Photos > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk supported compact pci boards
Are there any compact pci boards which asterisk supports? Thanks, Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how do we block registration based on ip/subnet?
How does one block registration of sip phones using asterisk if that sip phone is on a subnet other than the one allowed? Thanks, Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need unique switchboard/op-panel written
Hello, we are looking for someone to write a simple switchboard application for asterisk which can park, place on hold, and transfer to extensions. Standard switchboard/operator console features but we also need the following ability. We need to have a screen/panel which monitor the operators/main extension and displays a DID's owner. For instance, let's say we forward all of our DID's to a main extension. What we need to happen is, when someone calls a DID, the DID is looked up in a database(either odbc or astdb) and the called person's name is diplayed on the operators screen. IE a call comes in on did 555-555-, that did is looked up, and on the operator's screen, it should say "Call Arriving for Joe Company", the secretary should then be able to do all normal operator functions such as answer the call, forwarding, placing on hold, etc. The switchboard application can either be unix console based(perl-ncurses?) or Windows based. Anybody have a line on a good asterisk programmer that could do something like this? Brent Clements IO Networks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switchboards
On the actual switchboard side of this, how would asterisk handle this...going to a central person/extension as well as notifying that person who is calling and what line they are calling on. I have a customer right now who is hesitant to move to asterisk because of the switchbaord situation. Right now they have an executone which as a few t1's coming in. DID's are assigned to customers. In the executone switchboard application when a call comes in on a DID, the called party's(one of the businesses in the building) name shows up on a console and the receptionist can answer and forward the call as normal. I guess it's as easy as all of the DID's point to a single extension, then in that extension's context we could have an agi script which looks up the name associated with the did and display it some how. Something to think about...anyone has suggestions? On 8/5/05, Tom Hayden <[EMAIL PROTECTED]> wrote: > Well Don, it depends on how you get things setup from your telco. You > could get a T1/PRI (or E1) and just trunk all your calls. Then you > could have DIDs for each employee along with a primary number, which > could receive multiple calls at a time (it's just a DID). If you go > with the POTS solution, then you'll need to get a channel bank and do > things that way. Frankly, it would be a hell-of-a-lot-easier and > cheaper if you got trunked calls. Why buy 23 POTS channels when you > can probably get an equivalent number of channels + DIDs for much less > money? > > -- > Tom Hayden > Astoria Telecom, LLC > www.astoriatelecom.net > > On 8/5/05, Don Brearley <[EMAIL PROTECTED]> wrote: > > > > Hello, > > > > I am still researching my dive into Asterisk at my workplace, and I was > > wondering about how switchboard > > activities are handled.. Right now, a call comes into our switchboard, > > and the operator forwards them > > to the appropriate line, thus freeing up the primary number and allowing > > more calls in. Everyone on > > campus has a direct-dial line as it is right now. I want to eliminate most > > of those lines, and switch everyone > > to extensions instead. > > > > If I understand correctly, with Asterisk, i'll need to figure out how many > > lines are in use at any time > > (i'll say 20% to be safe) -- so I'll need to have roughly 25-30 POTS lines > > "on standby" for inbound calls? > > > > The way I see it in my head is, a call comes in on the primary number, and > > the operator will forward > > them to the correct extension, and Asterisk will route the call to another > > circuit, freeing up the > > primary line. Is this correct? > > > > (Sorry for my lack of correct terminology.. still getting familiar!) > > > > - Don > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > Tom > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and the IAD2431 via MGCP
Can you explain your thought pattern further? On 8/4/05, Mark Phillips <[EMAIL PROTECTED]> wrote: > Surely This is an ethernet issue rather than a * one? > > You are still left with the problem of the protocol on the trunk too. > > > > brent clements wrote: > > I have the following upcoming install and I'm trying to do it without > > having to resort to Digium t1 cards. > > > > > > I have a Cisco IAD2431 being installed by our Carrier. That Carrier > > will be providing 2 IP Trunks via ethernet handoff into the Cisco > > IAD2431. The CIsco IAD2431 has Two T1 ports installed and we would > > have to install a digium card to support those two t1's. > > > > What I'd like to do going completely ethernet with this bad boy. > > > > I'd like to have the following scenario: > > > > > > Carrier > > || > > || 2 IP trunks via eth handoff > > || > > || > > IAD2431 (this guy has 2 eth's on it available for networking goodness) > > \ > > \ Ethernet > > \ > > Powerful Asterisk Server > > \ > > \ > >--\--- > > SIPPHONESIPPHONE > > > > > > > > I'm assuming this all goes through using MGCP > > Anybody think this is doable? > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > Mark, G7LTT/KC2ENI > Randolph, NJ > http://www.g7ltt.com > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Voice Mail Server and older Executone PBX..can it be done?
Yeah we have been planning this deployment for 4 months, we are up to 300 phones as the trial. 1k phones is a bit overwhelming. B On 8/4/05, Mark Phillips <[EMAIL PROTECTED]> wrote: > Wow!! 1000 nodes. Good luck!!!! > > brent clements wrote: > > Well this is a temporary fix. We are gearing up a load balanced > > asterisk cluster with SER right now to replace their aging phone > > system. We have about 300 initial phones going to 1k VOIP phones > > we'll be installing. > > > > Brent > > > > On 8/4/05, Mark Phillips <[EMAIL PROTECTED]> wrote: > > > >>Not done it with Executone but have with others. The biggest problem is > >>knowing that you have a VM waiting for you. There's know way to light > >>the light on the PBX system. > >> > >>How many nodes on the old system? If its not to many why not replace it > >>with * entirely? Bring them out of the dark ages. > >> > >>Mark > >> > >>brent clements wrote: > >> > >>>Does anyone have experience with melding Asterisk with an older Executone > >>>PBX? > >>> > >>>I have a client whose existing voicemail server(repartee) has become > >>>bonkers and we need to stick a VM system in there asap. I thought > >>>asterisk would be a good thing to use. > >>> > >>>Does anyone have experience with the older Executone PBX's and asterisk? > >>> > >>>Any caveats, any tips, any things I should be aware of? > >>> > >>>Thanks, > >>>Brent > >>>___ > >>>Asterisk-Users mailing list > >>>Asterisk-Users@lists.digium.com > >>>http://lists.digium.com/mailman/listinfo/asterisk-users > >>>To UNSUBSCRIBE or update options visit: > >>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>> > >> > >>-- > >> > >>Mark, G7LTT/KC2ENI > >>Randolph, NJ > >>http://www.g7ltt.com > >>___ > >>Asterisk-Users mailing list > >>Asterisk-Users@lists.digium.com > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >>To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > Mark, G7LTT/KC2ENI > Randolph, NJ > http://www.g7ltt.com > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Voice Mail Server and older Executone PBX..can it be done?
Well this is a temporary fix. We are gearing up a load balanced asterisk cluster with SER right now to replace their aging phone system. We have about 300 initial phones going to 1k VOIP phones we'll be installing. Brent On 8/4/05, Mark Phillips <[EMAIL PROTECTED]> wrote: > Not done it with Executone but have with others. The biggest problem is > knowing that you have a VM waiting for you. There's know way to light > the light on the PBX system. > > How many nodes on the old system? If its not to many why not replace it > with * entirely? Bring them out of the dark ages. > > Mark > > brent clements wrote: > > Does anyone have experience with melding Asterisk with an older Executone > > PBX? > > > > I have a client whose existing voicemail server(repartee) has become > > bonkers and we need to stick a VM system in there asap. I thought > > asterisk would be a good thing to use. > > > > Does anyone have experience with the older Executone PBX's and asterisk? > > > > Any caveats, any tips, any things I should be aware of? > > > > Thanks, > > Brent > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > Mark, G7LTT/KC2ENI > Randolph, NJ > http://www.g7ltt.com > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and the IAD2431 via MGCP
I have the following upcoming install and I'm trying to do it without having to resort to Digium t1 cards. I have a Cisco IAD2431 being installed by our Carrier. That Carrier will be providing 2 IP Trunks via ethernet handoff into the Cisco IAD2431. The CIsco IAD2431 has Two T1 ports installed and we would have to install a digium card to support those two t1's. What I'd like to do going completely ethernet with this bad boy. I'd like to have the following scenario: Carrier || || 2 IP trunks via eth handoff || || IAD2431 (this guy has 2 eth's on it available for networking goodness) \ \ Ethernet \ Powerful Asterisk Server \ \ --\--- SIPPHONESIPPHONE I'm assuming this all goes through using MGCP Anybody think this is doable? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Features you'd like to see in a GUI?
Please stop hijacking this thread. Your little spat is not the point of this thread. Thanks On 8/4/05, Senad J <[EMAIL PROTECTED]> wrote: > [EMAIL PROTECTED] wrote: > > Senad, > > > > I don't want to take this conversation much further and send a laundry > > list of issues we faced with Switchware to this forum, and the myriads > > of bugs we have wrestled with in the past 18 months, as that list is > > too > > long and this forum is not for that purpose. > > We never said that our solutions are without bugs... However, we > sort bugs like no other company that I know off. In addition, software > without new features (and consequently producing bugs) is usually > "dead" software. > > > >> In addition, our clients we deal on regular basis do > >> not "disappear". Each and every client is looked after by us. > >> They do not, "install" the software try it and then come back > >> after several weeks/months and start the install process all > >> over again... That is not a workable model for anyone. > > > > The answer to your above comments is already known to you and it is > > already mentioned in Para-1. As you already knew, the two install > > efforts by me, one in September-October 2004 and One in > > November-December 2004 have failed to install a working software. On > > several occasions when I have reverted back to you on giving me a > > working version, your answer ( as well as Stephen Wingfield's ) was > > that > > Bicom don't have one ready. > > > Failed to install.. incorrect, system was installed and was working. The > fact that > you tried to install ASTCC and god know what else with it is was not and > will not be supported > by us. > > After you destroyed working copy, we offered to re-install it (for a charge > of $250), but > you have chosen the option to wait for SWITCHware CD. So, far we have not > produced > SWITCHware CD because SWITCHware product is too valuable to be distributed > by CD. > > > Infact, as I remember this correctly, Switchware was taken out of your > > product list due to these issues. > > As for us taking down SWITCHware off the product list... yes, you are right. > We had too many people wanting to buy it with additional features added to > it so we took it > off the product list. > > > Moreover is it ethical or legitimate > > to sell a software that has to be installed only by your technical > > team > > and works only on a specific hardware for a given network card and > > for a > > given IP. You don't even have a tarball to download and install till > > today, let alone a CD image?. This is not the way products are sold, > > where everything is closed, even the features of Asterisk that are > > available otherwise. > > It is our business model that you questioning... > What is next.. do you want to tell me what to wear? > > >> Our "installed" clients certainly do not try to look > >> into our source code and then say "it is not working". > > > > I don't understand what you mean by looking into your source code, but > > if you mean that "once Switchware is installed, live with what Bicom > > allows you to do with Asterisk, and don't ask questions like 'Why > > Cannot > > I use 4 Digit extensions' or 'Why cant I use ASTCC with Switchware' > > Because ACTCC it is NOT our supported product... > > > or 'Why cannot I create more than X number of channels'" > > Because every hardware box has limitation on how many calls it can process. > > > > or millions of > > other such questions, my answer is that this is totally out of line > > with > > the purpose of Asterisk as an Open Source software and if Digium knew > > what is going on, probably you are going to have legal issues as far > > as > > the GPL of the code is concerned, in the same way Sysmaster did with > > their SM7000 products. > > Are you a lawyer, or Digium representative to be able to make this comments? > > > Your Channel Locking and per channel pricing policy I am sure will > > put > > Oracle and Larry Ellsion to shame. > > > > I feel that Bicom should have spent more time making Switchware or > > PBXware work cleanly, rather than spending most of the time in > > copy protecting and closed sourcing Asterisk so that if the customer > > has > > to move the installation to another server, they have to call you for > > a > > new license code and pay you for the installation, rather than doing > > it > > himself. > > Again.. this is OUR business model... NOT yours. > > In addition to all of the above, how many times did we fail to respond to > your > support query or a just a simple "chat"? > In another words, you bringing up your issues here to this mailing list has > done no > favour to any one and we will NOT watch our reputation being talked about > > I do not think that other list users wish to read about issues which are not > directly related > to asterisk. You will not receive further reply from me on the matter. > > > > Senad Jordanovic > > > _
[Asterisk-Users] Asterisk Voice Mail Server and older Executone PBX..can it be done?
Does anyone have experience with melding Asterisk with an older Executone PBX? I have a client whose existing voicemail server(repartee) has become bonkers and we need to stick a VM system in there asap. I thought asterisk would be a good thing to use. Does anyone have experience with the older Executone PBX's and asterisk? Any caveats, any tips, any things I should be aware of? Thanks, Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Features you'd like to see in a GUI?
Please don't hijack this thread. On 8/4/05, Kanuri, Seshu (Company IT) <[EMAIL PROTECTED]> wrote: > Senad, > > I don't want to take this conversation much further and send a laundry > list of issues we faced with Switchware to this forum, and the myriads > of bugs we have wrestled with in the past 18 months, as that list is too > long and this forum is not for that purpose. > > >In addition, our clients we deal on regular basis do > > not "disappear". Each and every client is looked after by us. > > They do not, "install" the software try it and then come back > > after several weeks/months and start the install process all > > over again... That is not a workable model for anyone. > > The answer to your above comments is already known to you and it is > already mentioned in Para-1. As you already knew, the two install > efforts by me, one in September-October 2004 and One in > November-December 2004 have failed to install a working software. On > several occasions when I have reverted back to you on giving me a > working version, your answer ( as well as Stephen Wingfield's ) was that > Bicom don't have one ready. > > Infact, as I remember this correctly, Switchware was taken out of your > product list due to these issues. Moreover is it ethical or legitimate > to sell a software that has to be installed only by your technical team > and works only on a specific hardware for a given network card and for a > given IP. You don't even have a tarball to download and install till > today, let alone a CD image?. This is not the way products are sold, > where everything is closed, even the features of Asterisk that are > available otherwise. > > > Our "installed" clients certainly do not try to look > > into our source code and then say "it is not working". > > I don't understand what you mean by looking into your source code, but > if you mean that "once Switchware is installed, live with what Bicom > allows you to do with Asterisk, and dont ask questions like 'Why Cannot > I use 4 Digit extensions' or 'Why cant I use ASTCC with Switchware' or > 'Why cannot I create more than X number of channels'" or millions of > other such questions, my answer is that this is totally out of line with > the purpose of Asterisk as an Open Source software and if Digium knew > what is going on, probably you are going to have legal issues as far as > the GPL of the code is concerned, in the same way Sysmaster did with > their SM7000 products. > > Your Channel Locking and per channel pricing poilicy I am sure will put > Oracle and Larry Ellsion to shame. > > I feel that Bicom should have spent more time making Switchware or > PBXware work cleanly, rather than spending most of the time in > copyprotecting and closed sourcing Asterisk so that if the customer has > to move the installation to another server, they have to call you for a > new license code and pay you for the installation, rather than doing it > himself. > > This is against the open source philosophy we are all trying to benefit > from. > > Seshu Kanuri > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Senad J > Sent: Thursday, August 04, 2005 12:47 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Features you'd like to see in a GUI? > > > If your intention is just to build a GUI for Asterisk, read no > > further. > > If your desire is to build something more purposeful, your best bet > > would be to see the existing commercial GUI/HostedPBX offerings like > > Pbxware and Switchware from bicomsystems.com ( > > http://www.bicomsystems.com) and Thirdlane Technologies > > (http://www.thirdlane.com/opensource.htm) > > and > > the Open Source software like AMP and try to emulate (or preferably > > improve upon) them. > > > > My suggestion is to create a "VOIP Business in a Box System" that has > > inter-alia following list of modules: > > > > 1) GUI To configure Administer Asterisk Extensions across many servers > > 2) Postpaid and Prepaid Billing modules with realtime call progress > > detection and call cut-off 3)CRM Module for customers to register and > > provide their information for recurring billing. > > 4)Web based conference room management module 5)Web based click to > > dial and callback module > > > > Many of these modules are already available on Open Source like > > SugarCRM, AsreskiCC etc., and [EMAIL PROTECTED] CD contains AMP and > > SugarCRM at this time, besides other Open Source utilities like > > PhpMyAdmin. > > > > Here is the bottomline: > > -- > > The real need is for a commercially deployable solution that can > > create a business, without too many additions to it. > > > > Bicom Systems has promised for too long that their Pbxware and > > Switchware can fullfill this need to create a business, but they never > > > deliverd their promise. PBXware and Switchware have been a total and > > expensive disappointment to me an
Re: [Asterisk-Users] Features you'd like to see in a GUI?
There are so many asterisk management guis out there,some good, some not. I would suggest doing something different. I think there is a big need for a opensource virtual hosted pbx interface. I think it would help out alot of the smaller ITSP's who are trying to get into the virtual hosted pbx market but don't have the money or resources to develop or purchase a commercial product. We paid somebody to do ours though, if there was an opensource version when we were getting into this, we'd probably use it and also contribute. That's just my two cents. Brent Clements IO Networks On 8/3/05, Sherwood McGowan <[EMAIL PROTECTED]> wrote: > Hello all, > I'm currently working on an Open Source Asterisk Real Time GUI, called ARTCP > (I know it's not the most imaginative name), based in PHP and Javascript. > > What I'd like to know, is what features you'd like to see? I'm making two > different versions, PBX and Provider, so the features for each would be > different. > > Let me know, as I'd like to try to get as much as possible right the first > time. > > Cheers, > Sherwood McGowan > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Generic Question: Why should I use Asterisk over SIPxchange?
For those of you who have been working with asterisk for a while and who have experience with SIPxchange, why have you chosen Asterisk over the latter? What are some significant differences between the two that those of you familiar with both have discovered? Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] This should work right??? Any caveats that you guys know about?
Hello, long time lurker, first time writer We have the following set up ITSP | | Internet | | Cisco 2600 | | SwitchAsterisk Server running 1.0.9(has public ip) | | Cisco 515e Pix Firewall running Pix OS 5.3(run's a class c 1-to-1 nat and pat) | | Grandstream GXP-2000(run latest fw from grandstream site 1.0.1.9) The grandstream registers with the public asterisk server fine. I even see one of the dynamic nat addresses being assigned. The Pix Firewall has sip fixed up and all VOIP related ports are wide open. This is the issue: We can make outgoing calls, but we can't receive calls when the grandstream is behind the firewall If we move the grandstream in front of the pix and give it a public ip, everything works fine. What is even wierder is the fact that one of our network users who is behind the pix firewall can use ATT's VOIP service just fine. Are there any things I should be looking for? In general is the setup above pretty common? I've looked through the Wiki and searched google many times but nothing that can give me any pointers. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] System hardware requirements for *
What would be a the minimum hardware requirements for a small asterisk pbx that would only have 2 pots lines coming in(2 fxo ports) but with 4 extensions(4 fxs ports) And enough space to hold up to a month of vmail for those 4 extensions/users? Heck what are the typical hardware requirements of * anyways? I can't seem to find this on the website. I know it all depends on the situation,but what's typical? Thanks, Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multi-Line sip phone?
Hi, I'm looking for a multi-line sip/ip phone that can answer multiple incoming paths. IE a secretary sits at a front desk and can answer multiple incoming lines/DID's. Is there something like this? Thanks, Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What exactly does IAX and SIP termination mean???
So how does Asterisk fit in to the scheme of things when connecting to a IAX/Sip termination provider? Does a IAX/Sip termination provider just provide incoming call termination or does it do inbound and outbound? -Brent - Original Message - From: "Sean Cook" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Wednesday, December 01, 2004 10:26 PM Subject: Re: [Asterisk-Users] What exactly does IAX and SIP termination mean??? > Simplest explanation: > > IAX and SIP are protocols that allow devices to talk VOIP. When you > connect to a proxy device via either protocol, the proxy is said to > terminate those protocols. > > Brent Clements wrote: > > > I think I have idea what IAX and SIP termination means, but can > > someone explain this to me? > > > > For instance, how and why would I use someone like iax.cc? > > > > Thanks, > > Brent > > > > > > > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What exactly does IAX and SIP termination mean???
I think I have idea what IAX and SIP termination means, but can someone explain this to me? For instance, how and why would I use someone like iax.cc? Thanks, Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] After setting up my FXO card, what should I now order from my telco?
Well...the office doesn't currently have broadband, so vonage would be out. We don't plan on getting broadband at that office either so...PSTN is my only access option. I'm was just curious if there was a specific type of cheap line I should order. Please define the following though: 1. Vonage Line 2. Soft Line -Brent - Original Message - From: Luke Catranis To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, December 01, 2004 12:02 AM Subject: RE: [Asterisk-Users] After setting up my FXO card,what should I now order from my telco? Get a vonage line unlimited for $24.99 per month and get a softline and have vonage set it up as a rollover line. Grand total $34.98 per month. Route outbound to the ZAP/VONAGE first, then to the SIP/VONAGE NEXT, if need be setup an IAX termination account with voicepulse or iax.cc -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent ClementsSent: Wednesday, December 01, 2004 1:00 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] After setting up my FXO card,what should I now order from my telco? Ok, so I'm setting up my small office. I have my asterisk machine setup and I have 3 sip phones connected as my stations and a 4 port FXO card ready to go(planning on only using 2 lines currently). What should I now order from my telco(sbc in this case) Everytime I call, they want to sell me this expensive $50 package that bundles everything and that's for a single line. Is there a specific type of line I should request? What is everyone else doing? How much is everyone else paying for single line PSTN access for their small/medium setup's? Thanks, Brent ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] After setting up my FXO card, what should I now order from my telco?
Ok, so I'm setting up my small office. I have my asterisk machine setup and I have 3 sip phones connected as my stations and a 4 port FXO card ready to go(planning on only using 2 lines currently). What should I now order from my telco(sbc in this case) Everytime I call, they want to sell me this expensive $50 package that bundles everything and that's for a single line. Is there a specific type of line I should request? What is everyone else doing? How much is everyone else paying for single line PSTN access for their small/medium setup's? Thanks, Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
