Re: [asterisk-users] Passing variables over IAX2 -- IAXVAR patch?
Isn't this channel specific? Why is this being added? It does not work with SIP. It does not appear to be architecturally generic. This gets added, but yet a channel specific enhancement for SIP that would be beneficial for endusers does not get added. Again, Asterisk is good at transferring calls around, but when it comes to end users, the developers just keep closing the tickets on the much needed features. On Mon, Mar 24, 2008 at 4:36 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Monday 24 March 2008 16:08, Al Baker wrote: at what Rev of Asterisk has, or will, the patch exist in the Standard Release ? 1.6.0. -- Tilghman ___ -- 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] Passing variables over IAX2 -- IAXVAR patch?
On Mon, Mar 24, 2008 at 5:21 PM, Watkins, Bradley [EMAIL PROTECTED] wrote: Being able to pass variables around between systems is by *definition* channel-specific, since the channel driver is the module responsible for speaking a given protocol. Besdies, SIP already has (and has had for a long time) a method for doing this (SIP headers). So does ISDN, for that matter (IEs). I stand corrected. This is, in effect, I guess, completing a feature set. It is just catching IAX up with the other channels. I have no problem with that at all. Honestly, I never thought about using SIP headers to pass variables. I see it as a good feature, I was just ticked that a patch that had been worked on for so long was killed only because it was channel specific, and then this patch gets added. I understand now that feature wise, this wasn't really adding anything, it was more or less completing the feature set, a feature that I was not aware of to begin with until now. ___ -- 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] BLF and Snom phones
On Sat, Mar 22, 2008 at 7:17 AM, Philipp Kempgen [EMAIL PROTECTED] wrote: http://bugs.digium.com/view.php?id=5014 The response on that issue from Russell is the kind of response that really ticks me off. No, no, no, we don't want any real features that users want, we want basic, boring features. Asterisk is a call center system, not for regular, everyday business users. It could be so much more, though... Works great as an advanced IVR as a front end to a real phone system, though. ___ -- 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] Which IP Phone is really the best?
Wow! That is a good question. I can't believe no one has ever asked that before. Seriously, before letting someone join this mailing list, it should ask them a simple question: When you don't know the answer to something, do you: a) blindly ask hundreds of people with no regard to whether it was recently discussed b) search your favorite search engine for an answer first. We all know what this poster did. I realize my messages may seem rude and obnoxious, but let's face it, I'm just saying what the rest of you are thinking. I learned by reading, reading, and reading. The answer to almost every question is out there, you just have to look. It irritates me when other people don't even try to do research first and just want someone to spoon feed them the information. I wonder how many of those people are being paid for their knowledge, when in fact, they know nothing and are counting on the community to do their work for them. Anyway, that's enough for my rant for today. I'll leave with one thing: www.google.com - it's amazing what you can find On Aug 31, 2007 12:11 PM, William Herrera [EMAIL PROTECTED] wrote: I need to quote a client for a job and I was just wondering. Out of all the IP Phones out there, which one is the best and why? Thank you all, all opinions will be accepted. *William Herrera*** *LAN/WAN Technical Consultant* ___ --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] Trixbox and mail2fax
On Jan 2, 2008 12:23 AM, Daniel [EMAIL PROTECTED] wrote: Hi there, is there any howto how do i configure a asterisk/trixbox for mail2fax? The fax must be send over sipgate or other SIP peers. (i dont have any normal telephones connected). Do people even read the mail list anymore, or do they just land on this planet, subscribe to the list, and ask the same questions that's been asked over and over and over and over and over and over Read the archives, then ask questions! Or, at the minimum, take a look at a conversation that's been going on over the past two days or so. ___ --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] Softswitch digim
On Dec 2, 2007 3:42 PM, Carlos Rojas [EMAIL PROTECTED] wrote: I'm looking the softswitch in digium website, anyone test the softswitch? Nope. No one has tested it or used it. Try the one at cisco.com. ___ --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] Cisco 7970G line buttons
On 6/30/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I just upgraded my 7970G to the SIP firmware. What I'd like to do is have the 8 line buttons be able to make outbound calls using the same account (for practical purposes, same caller-ID). Since the phone is going to have a single public DID, when a call comes in, it should ring on the first available line. So, if I'm on line 1 and a call comes in, it should ring on line 2. How can this be done? Can I configure all lines to the same SIP user and will that do what I want? Do I have to configure each line to a separate user and then setup some roll-over configuration in Asterisk? Welcome to our Planet. Here on Planet Earth we have a search engine named Google. Specifically, www.google.com, please reference this amazing resource sometime while you are visiting our planet. We may seem to be simpletons to your advanced civilization, but most of us reference this resource before asking questions that have been answered thousands of times in the past. Thank you, and please enjoy your stay! Remember, www.google.com, it's amazing! ___ --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] Hot GXP-2000
Why? Because of their excellent customer support in taking care of a problem? At the price of the Grandstreams compared to others, I can deal with a couple of bad apples. I can buy two Grandstream's for the price of a phone with similar features. I can deal with a lot of bad apples at that ratio. Their sidecar is so cheap compared to others it's not even funny. Plus, I can't even get some of the functionality the GXP's give me from other phones. On 6/9/07, Dovid B [EMAIL PROTECTED] wrote: One of the reasons why I stand clear of Grandstream - Original Message - From: Carlos Chavez [EMAIL PROTECTED] To: Asterisk asterisk-users@lists.digium.com Sent: Friday, June 08, 2007 6:47 PM Subject: [asterisk-users] Hot GXP-2000 ___ --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
Re: [asterisk-users] Best Hardphone (Subjective?)
On 4/2/07, Corporate IT Solutions - Michael Dunne [EMAIL PROTECTED] wrote: So subjectively what would be the best Hardphone for a small/medium business with multiple line support, BLF, etc. Does _anyone_ read the archives anymore? This is like a weekly question or something. ___ --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] Cisco 30VIP Phone
On 3/21/07, Chris Nighswonger [EMAIL PROTECTED] wrote: I have just successfully configured a Cisco 30VIP to work with my Asterisk server. I have seven of these phones new and would like to deploy them. I am wondering if anyone has this phone deployed with Asterisk and can suggest configuration of the various buttons, etc. (Bare with me as I am new to Asterisk.) So which is it? You either have it configured or you don't. ___ --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] Cisco 7970 with skinny on * 1.4.1
On 3/21/07, Richard Klingler [EMAIL PROTECTED] wrote: As chan_sccp is pretty much dead, doesn't compile on FBSD anyway and isn't supported on * 1.4.x I tried going with chan_skinny... chan_sccp is far from dead and it works with 1.4. more fud being spread... ___ --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 1.4.2 chan_zap
On 3/21/07, Jeremiah Millay [EMAIL PROTECTED] wrote: chan_zap won't compile in asterisk 1.4.2 when used with zaptel 1.4.0. The changelog has this entry: And you missed all the other discussions about it not working? Or, are you just special and wanted your own thread? ___ --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] A request for your input.
On 3/22/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello P.S The program that I am using is open source, of course (www.phpsurveyor.org)! What part of the survey is running Asterisk? ___ --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] CN=Diarmaid O'Loughlin/O=QAD1 is out of the office.
On 3/22/07, Diarmaid O'Loughlin [EMAIL PROTECTED] wrote: I will be out of the office starting 22-03-2007 and will not return until 16-04-2007. Congratulations! I will be on holidays until the 16th of April. I will have no access to e-mail during this time. If this is a matter relating to IT support, please contact the IT Helpdesk by either opening a ticket via http://helpdesk.qad.com/request. Or calling the EMEA helpdesk on +31 20 654 7139. I'll be sure and do that. First ticket will be How many support personnel does it take to make sure the out of office replies don't get sent to thousands of users who just don't care? If this is an urgent matter not relating to IT support. Please contact either Jim Josey (jzj-at-qad.com), Paul Callan (plc-at-qad.com) or you may contact me again after Apr 16th. I'll be sure and contact them next time I have a non IT-related problem. ___ --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] voip-info.org status update
On 3/15/07, OCOSA List Acct. [EMAIL PROTECTED] wrote: Hi All, and if you all depend on James' site so much then you need to donate some time or contact him about getting a mirror. The so called new site Google didn't go down, and if you had bothered searching the archives of this list you would have known this has all been discussed before. Whoever the powers that be that run the wiki did not want help before. People had begged to be able to mirror the site. This could have all been avoided. Maybe this time they will be a little more interested in getting some mirrors. And as for whoever said this reflects on Asterisk, if you have to depend on the wiki to help your clients maybe you need to step back and see how this reflects on you. If you're charging customers by the hour for something, you need to know the stuff and not have to spend most of that time searching for answers. Yeah, I know, most consultants these days have no clue about what they sell to clients. We see that every day on this list alone. Step up and be different. ___ --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: Fwd: [asterisk-users] Some queries on g729 license.
On 1/9/07, Al Bochter [EMAIL PROTECTED] wrote: So do you think Digum and Sipro is now one in the same code with G729 in mind? If saying this will make this go away, then yes. They both use the same code. The patented code is the same. ___ --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] Some queries on g729 license.
On 1/8/07, Al Bochter [EMAIL PROTECTED] wrote: Mike, So tell me what this FREE open source G729 is I am told that you can use these Codecs with your Asterisk ! http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ You can do it Freely !! Al, I don't know if you're stupid, or you just like stirring things up. Once again, READ! Read the entire article before posting it. To quote: To use G.729 or G.723.1 you may need to pay a royalty fee. Please see http://www.sipro.com for details. Please note that this code is available for you to download for education purposes only and if a patent exists in your country for G.729 or G.723.1 then you should contact the owner of that patent and request their permission before executing the code. Now, Al, what does that say? I don't know what country you live in (and don't care), but if you live in a country (or possibly do business with a country) that honors patents, then you will have to pay to license this codec. Just because I _can_ break the law, does not mean that I should, or that I have the right to, or that it's ok to do so. ___ --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] Some queries on g729 license.
On 1/8/07, Al Bochter [EMAIL PROTECTED] wrote: I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 I only know what I have been told... I have been told... I have never used... All common phrases with this person. I have never seen somebody spread as much third party information as this person spreads. He knows nothing, yet informs all. It's real simple to give it a shot and see what happens. If you can't afford $10 for testing, maybe you are in the wrong business. You claim to have clients using Asterisk. I'd hate to be one of your clients. I would hope that my consultant has at least tried the things he/she is suggesting to me. FWIW, I have used Digium's g729 and it works great, and is about as simple to install as you can get. ___ --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: RE : [asterisk-users] Happy 2007!!!
On 12/31/06, Adam Jacob Muller [EMAIL PROTECTED] wrote: It's still 2006 here -Adam Well, Adam, I guess it is all about you. What does the rest of the world look like as it revolves around you? ___ --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] The Good, Bad and Scam VoIP Providers
On 12/24/06, Al Bochter [EMAIL PROTECTED] wrote: So you would deal with a criminal ? Bret McDanel was *Convicted Of Cybercrimes* And did you finish reading wherever you got that information? No, because it didn't suit your allegations. Try finishing it: Prosecution voluntarily moved to vacate the conviction Have you ever gotten a speeding ticket, or anything similar? If so, then you are a criminal as well. Maybe your end users need to know this information. I've had enough of this dirt bag troll. And PS, don't send me anymore direct emails. I didn't request them, and they look like spam to me. ___ --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] The Good, Bad and Scam VoIP Providers
Geez Al, let it go. We've heard your rants for what seems like years now (even though it's only been weeks). No one cares anymore. ___ --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
