[Asterisk-Users] Simulating the "lighted line in use" type of phone
I'd like to see if anyone out there might have some ideas on this. I have a customer who wants to move to VoIP, but who has an office full of people who are very conservative about their telephones. They would like the asterisk system that I am proposing to have something analogous to what they have right now, which are plain 4-line analog phones with a light showing any lines in use, and a button beneath that they press to choose an outbound line. Before I tell them that this is a silly retrograde behavior I thought I'd check first to see how others of you out there might handle this same desideratum. Thanks in advance. b. ___ 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] Simulating the "lighted line in use" type of phone
I would put it in a totally different light. IE; depending on who they use as an IAX/SIP carrier, they may have potentially unlimited outbound and inbound lines with the limit only imposed by the total number of indications on the phones in the office and even then, new inbound calls can still go to voicemail or a call queue. John Brian Capouch wrote: I'd like to see if anyone out there might have some ideas on this. I have a customer who wants to move to VoIP, but who has an office full of people who are very conservative about their telephones. They would like the asterisk system that I am proposing to have something analogous to what they have right now, which are plain 4-line analog phones with a light showing any lines in use, and a button beneath that they press to choose an outbound line. Before I tell them that this is a silly retrograde behavior I thought I'd check first to see how others of you out there might handle this same desideratum. Thanks in advance. b. ___ 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
RE: [Asterisk-Users] Simulating the "lighted line in use" type of phone
I have a similar situation, I have an office with 4 lines that are answered in different names and wanted to give users the option to see the status of each line etc. I have some cisco vip30 with the select buttons and appropriate leds, I was going to set these up on a * box and see how developed the skinny driver is as far as making these buttons/leds let the users select particular lines and indicate status. The lines are mainly analogue on x100 cards Maybe somebody has already worked on this and can give some guidance. The hardware looks like it will do the job, the only question is if we can hack the software to do what we want. craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: Wednesday, 25 February 2004 08:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Simulating the "lighted line in use" type of phone I'd like to see if anyone out there might have some ideas on this. I have a customer who wants to move to VoIP, but who has an office full of people who are very conservative about their telephones. They would like the asterisk system that I am proposing to have something analogous to what they have right now, which are plain 4-line analog phones with a light showing any lines in use, and a button beneath that they press to choose an outbound line. Before I tell them that this is a silly retrograde behavior I thought I'd check first to see how others of you out there might handle this same desideratum. Thanks in advance. b. ___ 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
Re: [Asterisk-Users] Simulating the "lighted line in use" type of phone
That's fine for outbound lines, but what if I want to call the guy in the next office ? I have to call him and get redirected to his busy vm just to know that he's on the phone. This is a huge issue with the recepetionist with the 'master console'. How does he/she know whether a user is busy or not ? 20 year old phone systems offer this capability. I see this as a serious shortfall of asterisk / currently available ip phones right now. - Chris Clifton - Original Message - From: "John Fraizer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, February 24, 2004 3:42 PM Subject: Re: [Asterisk-Users] Simulating the "lighted line in use" type of phone > > I would put it in a totally different light. IE; depending on who they use > as an IAX/SIP carrier, they may have potentially unlimited outbound and > inbound lines with the limit only imposed by the total number of indications > on the phones in the office and even then, new inbound calls can still go to > voicemail or a call queue. > > John > > Brian Capouch wrote: > > I'd like to see if anyone out there might have some ideas on this. > > > > I have a customer who wants to move to VoIP, but who has an office full > > of people who are very conservative about their telephones. > > > > They would like the asterisk system that I am proposing to have > > something analogous to what they have right now, which are plain 4-line > > analog phones with a light showing any lines in use, and a button > > beneath that they press to choose an outbound line. > > > > Before I tell them that this is a silly retrograde behavior I thought > > I'd check first to see how others of you out there might handle this > > same desideratum. > > > > Thanks in advance. > > > > b. > > ___ > > 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
Re: [Asterisk-Users] Simulating the "lighted line in use" type of phone
Chris Clifton wrote: That's fine for outbound lines, but what if I want to call the guy in the next office ? I have to call him and get redirected to his busy vm just to know that he's on the phone. This is a huge issue with the recepetionist with the 'master console'. How does he/she know whether a user is busy or not ? 20 year old phone systems offer this capability. I see this as a serious shortfall of asterisk / currently available ip phones right now. Gastman will take care of that, but I've still been faced with occasional segfaults with it. There's also the possiblility of using something like expect, or the manager interface, or even training a person to "show channels" in the CLI. Of course, this means that the "receptionist" is going to have to have a terminal with a window up dedicated to this function. That's where things are going, so I don't mind doing it much, other than the fact that I'm breaking their desire to simply map the old functions over to asterisk. B. ___ 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] Simulating the "lighted line in use" type of phone
On Tue, Feb 24, 2004 at 03:49:36PM -0500, Chris Clifton wrote: > That's fine for outbound lines, but what if I want to call the guy in the > next office ? I have to call him and get redirected to his busy vm just to > know that he's on the phone. > > This is a huge issue with the recepetionist with the 'master console'. How > does he/she know whether a user is busy or not ? > > 20 year old phone systems offer this capability. > > I see this as a serious shortfall of asterisk / currently available ip > phones right now. I'm scratching out a way of doing this using the manager port. There is a TON of info that is passed to the manager port once you are authenticated. I'd love the interface to be Java (portablilty), but it could be web based with a short refresh (say 5 seconds) from a database. Tim -- >< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << >< ___ 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] Simulating the "lighted line in use" type of phone
Take a look at my GUI app: http://sourceforge.net/projects/astguiclient/ It'll run on Linux and Windows, it's written in perl and it'll list every channel(Zap/SIP/Local) that is active on your system updated every second. You can also do a lot of other things with it too. We've been using it here at my company for months on over 60 computers and it scales rather well. MATT--- -Original Message- From: Tim Sailer [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 24, 2004 4:24 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Simulating the "lighted line in use" type of phone On Tue, Feb 24, 2004 at 03:49:36PM -0500, Chris Clifton wrote: > That's fine for outbound lines, but what if I want to call the guy in the > next office ? I have to call him and get redirected to his busy vm just to > know that he's on the phone. > > This is a huge issue with the recepetionist with the 'master console'. How > does he/she know whether a user is busy or not ? > > 20 year old phone systems offer this capability. > > I see this as a serious shortfall of asterisk / currently available ip > phones right now. I'm scratching out a way of doing this using the manager port. There is a TON of info that is passed to the manager port once you are authenticated. I'd love the interface to be Java (portablilty), but it could be web based with a short refresh (say 5 seconds) from a database. Tim -- >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< ___ 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
RE: [Asterisk-Users] Simulating the "lighted line in use" type of phone
Just a thought on this topic, if you're using Cisco 7940/7960 IP handsets it should be possible to write a Perl script on the Asterisk box that updates the XML directory to show the current status of extensions. Then when the receptionist is on the phone, they could hit the services button, scroll through the list of extensions; see what the persons status is, and even transfer the call right through by pressing the Dial button underneath the extension if they hit Transfer before the Services button. I don't have enough experience with coding the XML directory extension on the Cisco beyond a simple phone-directory, but Cisco do have a programming guide available and there are some advanced features you could use. In fact even with a simple XML phone-directory, all you need do is have the Perl script interrogate Asterisk for the extension status, and if the extension is busy, simply append BUSY in capitals onto the end of the persons name. Then when the extension frees, remove the BUSY message... this should work, and since the phone does a HTTP request every time the user hits the Services button, if it's a script that's executed on the server then they should have an up to date view of who's busy/free. Granted if someone who was busy hangs up before the receptionist can scroll to their name, it will still show as busy, but apart from that it should do the trick and could be the next best thing to a live LED busy lamp field. This would remove the need for a PC running status software; although at the end of the day a software solution could probably be a lot more user friendly and flexible than what you can do through XML. The Cisco IP phones may be a little pricier than some others on the market, but as far as I'm concerned they currently are the bee's knees as far as IP phones go. Awesome call quality and best speakerphone I've ever had, and coupled with the XML/HTTP capabilities of the phone you can't go too wrong. Before you jump in the deep end with deploying IP telephony in an office environment, get yourself set up with two (or just 1 if money is tight) Cisco 7940/7960's w/SIP firmware talking with Asterisk and see what can be done. Besides, the Cisco 7940/7960's look far sexier than 20 year old dinosaur key stations and you'll be the envy of everyone in the office :-) Cheers, Chris Lee > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Chris Clifton > Sent: Wednesday, 25 February 2004 6:50 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Simulating the "lighted line in use" type of > phone > > That's fine for outbound lines, but what if I want to call the guy in the > next office ? I have to call him and get redirected to his busy vm just to > know that he's on the phone. > > This is a huge issue with the recepetionist with the 'master console'. How > does he/she know whether a user is busy or not ? > > 20 year old phone systems offer this capability. > > I see this as a serious shortfall of asterisk / currently available ip > phones right now. > > - Chris Clifton ___ 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] Simulating the "lighted line in use" type of phone
> Then when the receptionist is on the phone, they could hit the services > button, scroll through the list of extensions; see what the persons > status is, and even transfer the call right through by pressing the Dial > button underneath the extension if they hit Transfer before the Services > button. Um, no. Reception needs an 'at a glance' view of all extensions -- I don't think you've ever worked a switchboard or seen some of these people handle the call volumes they are used to. Anything involving button presses or scrolling would be useless. +2 points for trying, though. :-) I am positive I saw add-on modules for Cisco phones to show line/extension status, though. Regards, Andrew ___ 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] Simulating the "lighted line in use" type of phone
Hi! > I am positive I saw add-on modules for Cisco phones to show > line/extension status, though. It appears that for the SNOM 220 there is a 20 key extension pad, and you can use up to 3 of those per phone --> maximum of 60 lines to monitor? Cheers, Philipp ___ 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] Simulating the "lighted line in use" type of phone
> I am positive I saw add-on modules for Cisco phones to show line/extension > status, though. Your did http://cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008 008883d.html alas no one has done any integration work with * yet ___ 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] Simulating the "lighted line in use" type of phone
> > Then when the receptionist is on the phone, they could hit the services > > button, scroll through the list of extensions; see what the persons > > status is, and even transfer the call right through by pressing the Dial > > button underneath the extension if they hit Transfer before the Services > > button. > > Um, no. Reception needs an 'at a glance' view of all extensions -- I > don't > think you've ever worked a switchboard or seen some of these people handle > the call volumes they are used to. Anything involving button presses or > scrolling would be useless. True, probably too much assumption there on my part. I figured the old switchboards of yesteryear were pretty much redundant, or had loads reduced these days with direct in dial through BRI/PRI interfaces. Anyway, in a high call volume environment you might as well go the software route and get setup with a computer and a nice colourful busy lamp display on screen. Cheers, Chris Lee ___ 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] Simulating the "lighted line in use" type of phone
On Tue, Feb 24, 2004 at 04:33:41PM -0500, mattf wrote: > Take a look at my GUI app: > > http://sourceforge.net/projects/astguiclient/ > > It'll run on Linux and Windows, it's written in perl and it'll list every > channel(Zap/SIP/Local) that is active on your system updated every second. > > You can also do a lot of other things with it too. > > We've been using it here at my company for months on over 60 computers and > it scales rather well. I looked at it, and it won't work for me the way you have written it without me hacking at it. You insist on having MySQL run in network mode. I really don't want another port/service running on a box that is internet-facing... call me silly. :) For things like this, I'll only have the service running locally. Tim -- >< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << >< ___ 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] Simulating the "lighted line in use" type of phone
Hi, > -Original Message- > That's fine for outbound lines, but what if I want to call > the guy in the next office ? I have to call him and get > redirected to his busy vm just to know that he's on the phone. > > This is a huge issue with the recepetionist with the 'master > console'. How does he/she know whether a user is busy or not ? > > 20 year old phone systems offer this capability. > > I see this as a serious shortfall of asterisk / currently > available ip phones right now. Asterisk has no idea of what's 'inside' and 'outside' of the PBX. That's part of the flexibility you get. The behaviour you want can be delivered by using more intelligent IP-phones (Cisco 7960 can probably do this if you program the XML dialogs yourself), but it is handywork to do, because everybody's setup is different. BTW, using ADSI might allow you to do the same... BTW, if you care to use MGCP, RFC3149 might be for you - it's an extension to MGCP based on XML signalling. How difficult is it to combine this with Asterisk systems? Florian ___ 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] Simulating the "lighted line in use" type of phone
That's great, but the Polycom phones and the Cisco phones are already capable of this. We just need asterisk to comply: >From the Polycom admin manual at: http://www.polycom.com/common/pw_item_show_doc/0,1276,2545,00.pdf 5.2.6 Shared Call Appearance Signaling A shared line is an address of record managed by a back to back user agent (B2BUA) server. The B2B server allows multiple endpoints to register locations against the address of record. SoundPoint® IP supports shared call appearances (SCA) using the SUBSCRIBENOTIFY method in the "SIP Specific Event Notification" framework (RFC 3265). The events used are: . "call-info" for call appearance state notification . "line-seize for the telephone to ask to seize the line 5.2.6.1.1 Call-info header The "call-info" header is used in INVITE, SUBSCRIBE, and NOTIFY to give appearance index information. "call-info" header format: Call-Info = "Call-Info" HCOLON info *(COMMA info) info = LAQUOT absoluteURI RAQUOT *( SEMI info-param) info-param = ( "purpose" EQUAL ( "icon" / "info" / "card" / token)) / generic-param Two "generic-param" fields are defined: "appearance-index" EQUAL ( (1*DIGIT) / STAR ) ) "appearance-state" EQUAL ( "idle" / "seized" / "progressing" / "alerting" / "active" / "held" ) Example of call-info header in INVITE request: INVITE sip:[EMAIL PROTECTED] ... From: ... To: ... Call-Info: ;appearance-index=3 ... Example of call-info header in response: SIP/2.0 200 OK ... From: ... To: ... Call-Info: ;appearance-index=3 ... Example of call-info header in NOTIFY request: NOTIFY sip:[EMAIL PROTECTED] ... From: ... To: ... Call-Info: ;appearance-index=2;appearance-state=held, ;appearance-index=3;appearance-state=idle, ;appearance-index=4;appearance-state=idle Event: call-info ... 5.2.6.1.2 Call-info event package SoundPoint® IP sends a SUBSCRIBE message with "event" set to "call-info" to subscribe to call information for a specific address of record. For example: SUBSCRIBE sip:[EMAIL PROTECTED] ... From: ... To: ... event: call-info ... The server sends NOTIFY messages to all telephones which share that line to indicate the line state. For example: NOTIFY sip:[EMAIL PROTECTED] ... From: ... To: ... Call-Info: ;appearance-index=1;appearance-state=active, ;appearance-index=2;appearance-state=held, ;appearance-index=3;appearance-state=idle, ;appearance-index=4;appearance-state=idle Event: call-info ... And there's more. Now, if someone could point me in the right direction. Chan_Sip2, maybe? Olle E. Johansson, where are you? John - Original Message - From: "Tim Sailer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, February 24, 2004 9:37 PM Subject: Re: [Asterisk-Users] Simulating the "lighted line in use" type of phone > On Tue, Feb 24, 2004 at 04:33:41PM -0500, mattf wrote: > > Take a look at my GUI app: > > > > http://sourceforge.net/projects/astguiclient/ > > > > It'll run on Linux and Windows, it's written in perl and it'll list every > > channel(Zap/SIP/Local) that is active on your system updated every second. > > > > You can also do a lot of other things with it too. > > > > We've been using it here at my company for months on over 60 computers and > > it scales rather well. > > I looked at it, and it won't work for me the way you have written it > without me hacking at it. You insist on having MySQL run in network > mode. I really don't want another port/service running on a box that > is internet-facing... call me silly. :) For things like this, I'll only > have the service running locally. > > Tim > > -- > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< > >> Tim Sailer >< Coastal Internet, Inc. << > >> Network and Systems Operations >< PO Box 726 << > >> http://www.buoy.com >< Moriches, NY 11955 << > >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< > ___ > 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
RE: [Asterisk-Users] Simulating the "lighted line in use" type of phone
> I looked at it, and it won't work for me the way you have written it > without me hacking at it. You insist on having MySQL run in network > mode. I really don't want another port/service running on a box that > is internet-facing... call me silly. :) For things like this, I'll > only have the service running locally. > > Tim Just because a port is active doesn't mean that the world has to be able to see it. I've not used this script(astguiclient), is it intended to be ran from the client, or from the server? - Andrew Thompson http://aktzero.com/ ___ 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] Simulating the "lighted line in use" type of phone
It actually has components that run on the Asterisk server as well as a GUI app that runs on the desktop(Win32 or Linux). astguiclient was really designed for a larger environment than most of the Asterisk client apps that are out there, and it is not as easy to set up. It was also initially designed for the specific needs of my company. It needs a MySQL server to be located somewhere that is accessible to the Asterisk server and the client machines.(I'm thinking I should add a $MySQL_port variable to the config files so that you can easily change the MySQL port the server or client apps connect to for those with "creative" firewalls) The reasons that MySQL is used are simple: Scalability and Stability I figured that out when I tried to have 60 concurrent connections to the manager port getting updated info on channel statuses and the Asterisk server quickly slowed to a crawl. I wrote an application that runs on the Asterisk server to grab channel info and dump it to the MySQL database so that potentially hundreds of clients can be running at the same time without impacting the Asterisk server at all. Also, if you use the new optional Central Queue System that is bundled with the latest release, the client app never actually connects to the Asterisk server, all call initiations, transfers, and other commands are put into a central MySQL queue table and are initiated locally on the Asterisk server. But you don't have to use the CQS to have full functionality of the GUI app. If anyone has any questions about astguiclient just send me an email. MATT--- -Original Message- From: Andrew Thompson [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 25, 2004 5:47 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Simulating the "lighted line in use" type of phone > I looked at it, and it won't work for me the way you have written it > without me hacking at it. You insist on having MySQL run in network > mode. I really don't want another port/service running on a box that > is internet-facing... call me silly. :) For things like this, I'll > only have the service running locally. > > Tim Just because a port is active doesn't mean that the world has to be able to see it. I've not used this script(astguiclient), is it intended to be ran from the client, or from the server? - Andrew Thompson http://aktzero.com/ ___ 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
RE: [Asterisk-Users] Simulating the "lighted line in use" type of phone
On Wed, 25 Feb 2004, mattf wrote: > It actually has components that run on the Asterisk server as well as a GUI > app that runs on the desktop(Win32 or Linux). > > astguiclient was really designed for a larger environment than most of the > Asterisk client apps that are out there, and it is not as easy to set up. It > was also initially designed for the specific needs of my company. > > It needs a MySQL server to be located somewhere that is accessible to the > Asterisk server and the client machines.(I'm thinking I should add a > $MySQL_port variable to the config files so that you can easily change the > MySQL port the server or client apps connect to for those with "creative" > firewalls) How about allowing people to specify 127.0.0.1? ;) That solves the problem of running it on a different box. Just bind the local instance to that IP only. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Simulating the "lighted line in use" type of phone
On Wed, Feb 25, 2004 at 05:46:55PM -0500, Andrew Thompson wrote: > > I looked at it, and it won't work for me the way you have written it > > without me hacking at it. You insist on having MySQL run in network > > mode. I really don't want another port/service running on a box that > > is internet-facing... call me silly. :) For things like this, I'll > > only have the service running locally. > > > > Tim > > Just because a port is active doesn't mean that the world has to be able > to see it. True, but my * server is on the public Internet, not behind a firewall. > I've not used this script(astguiclient), is it intended to be ran from > the client, or from the server? Both, it seems. Tim -- >< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << >< ___ 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] Simulating the "lighted line in use" type of phone
On Tue, Feb 24, 2004 at 10:37:21PM -0500, Tim Sailer wrote: > On Tue, Feb 24, 2004 at 04:33:41PM -0500, mattf wrote: > > Take a look at my GUI app: > > > > http://sourceforge.net/projects/astguiclient/ > > > > It'll run on Linux and Windows, it's written in perl and it'll list every > > channel(Zap/SIP/Local) that is active on your system updated every second. > > > > You can also do a lot of other things with it too. > > > > We've been using it here at my company for months on over 60 computers and > > it scales rather well. > > I looked at it, and it won't work for me the way you have written it > without me hacking at it. You insist on having MySQL run in network > mode. I really don't want another port/service running on a box that > is internet-facing... call me silly. :) For things like this, I'll only > have the service running locally. OK, following up on my own e-mail, I have the gui working with the unix socket by doing the following: in libs/AST_VICI_conf.pl: $DB_server = '';# MySQL server IP #added unix socket to support local use - tps 02252004 $DB_unixsocket = '/var/run/mysqld/mysqld.sock'; # path to local socket in AST_WINphoneAPP_0.9.pl and AST_update.pl, changed every instance of: my $dbh = Net::MySQL->new(hostname => "$DB_server", database => "$DB_database", user => "$DB_user", password => "$DB_pass") or die "Couldn't connect to database: \n"; to my $dbh = Net::MySQL->new(unixsocket => "$DB_unixsocket", hostname => "$DB_server", database => "$DB_database", user => "$DB_user", password => "$DB_pass") or die "Couldn't connect to database: \n"; Now it starts up and runs, and the Zap channels are seen, but not the SIP ones yet... Tim -- >< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << >< ___ 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