Re: [asterisk-biz] Remote SIP monitor
On Wed, 2010-01-06 at 09:45 +0100, Olle E. Johansson wrote: I'm adding manager events and storing data in a realtime database - one record per call leg. What I'm wondering is how we should handle call transfers and hold situations. A call that's transferred has multiple streams and RTCP is only valid for one stream. Would it make more sense to expose the RTCP information as part of the CEL (Call Event Logging) infrastructure? That way, you could tell what events in the call may have triggered the additional streams. In a perfect world, we might even have something like: Event: Incoming call from Alice Event: Outgoing call to Bob Event: Asterisk bridges Alice to Bob Event: RTCP report Event: RTCP report Event: RTCP report Event: Alice places Bob on hold Event: RTCP report (new stream, Bob hears hold music from Asterisk) Event: Unhold Event: RTPC report (Alice and Bob, again) Event: Bob transfers Alice to Charlie Event: RTCP report (new stream, Alice and Charlie) Event: Hangup Obviously that's an oversimplified example, but I really think it makes more sense to put the RTCP reports in the CEL logs, rather than having another arbitrary log for the data. Maybe we should move this discussion to the -dev list to discuss the more technical details? Anyhoo, just wanted to add my own two cents (US cents, before interest, taxes, depreciation, and amortization) -- Jared Smith Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] how can i play IVR without Answer the call.....
On Wed, 2009-11-18 at 22:00 +0300, Mian Asif wrote: i am facing one issue, i am running one IVR with playback option and one IVR with Background option.. but in user site call has been connect and user provider charge him, i want to play one IVR file but call should not connect customer site, its a simple IVR message that user is not registered. You should probably be asking this on the -users list, but here goes anyway. You should pass the noanswer option to the Playback application or the n option to the Background application, to tell them not to answer the call. exten = 123,1,Progress() exten = 123,n,Playback(user-is-not-registered,noanswer) exten = 123,n,Playback(goodbye,noanswer) exten = 123,n,Hangup() -- Jared Smith Training Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Brisbane AU Asterisk help
On Tue, 2009-11-03 at 08:12 -0500, David Shauger wrote: Anyone in Brisbane AU that can help train some folks on using Asterisk? Digium has an authorized training partner in Melbourne. I realize that's not exactly close to Brisbane, but we might be able to arrange for them to host a class in Brisbane. Please contact me offline for more details. -- Jared Smith Training Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] How do i enter two different SMTP in Asterisk VoiceMail
On Tue, 2009-06-09 at 11:35 -0600, Brandon B. wrote: I am 100% confident sendmail, postfix and exim properly handle SMTP delivery issues, so how can you justify calling for functionality within Asterisk to do this same task? Or, if configuring sendmail/postfix/exim/qmail are beyond your abilities, there's also ssmtp, which is a super-simply mail submission agent. -- Jared Smith Training Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Dear Termination Providers,
On Fri, 2009-04-24 at 22:20 -0700, John Todd wrote: Why is this? My belief (which is based on painful experience) is that they will always overcharge by 3%-15% because of data slop, and they will make it as difficult as possible for you to argue the cost delta. It's a way of bleeding the clueless. If you're smart, you'll know how to catch them at this game. If you're a sloppy carrier who doesn't care or thinks that it doesn't matter, they'll double their profit on you. I couldn't agree more. I once had the opportunity (in a previous job) to turn up accounts with several different long-distance providers, run several hundred thousand calls through each one, and then compare their advertised rates to the bill that came at the end of the month. We had seven providers that passed our initial screen process and went on to real-world testing. Of those, only *one* actually billed what they said they would. (And, even more surprising... they were the most communicative and by far the best to work with.) Three of the seven providers over-billed us by more than 25%! The most common issue I found with overbilling was carriers were claiming to do six-second or one-second increment billing, but actually charging for full minutes. Other common issues included charging for uncompleted calls, claiming a flat rate but charging a blended rate, or charging completely different rates than their rate table specified. I hate to admit it, but I'm now a firm believer that most carriers in the business of selling minutes are purposefully deceptive, or at a minimum almost completely incompetent. I'm also a firm believer that they feel justified in doing so because their customers rarely challenge them on it. (It's human nature to want to avoid auditing the phone bill... When is the last time you did this?) In short -- you've got the tools to be able to audit your carrier and see how well they're living up to their promised rates. Do your homework, and you might save more money than you realize. -- Jared Smith Training Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Asterisk 1.4.22.1 and zombies
On Wed, 2009-01-21 at 15:54 +0100, Sabine Jordan wrote:
The defunct processes are from the application but these proceses are
children of the asterisk process.
[snip]
Any more ideas would be appreciated. Thank you in advance.
One of the things that can cause problems with AGI scripts is if your
AGI script doesn't properly listen for a SIGHUP signal from the
operating system, and react accordingly. When the caller hangs up,
Asterisk ends up sending the SIGHUP signal to your application, and if
your application doesn't listen for that signal and shut down, it can
obviously become defunct.
Unfortunately, with PHP, this is a little more difficult than it ought
to be.
I found this little nugget of knowledge a long time ago on one of the
mailing lists or forums (but unfortunately didn't keep track of where I
found it), and it's been useful for me. Your mileage may obviously
vary.
You must have the pcntl module for PHP installed for this to work
properly.
?php
declare(ticks=1);
// don't buffer output... flush it immediately
ob_implicit_flush(true);
function sig_handler($signo){
// the code in this function will execute when the caller hangs up
// and Asterisk sends the SIGHUP signal.
//
// for example, you could have your program write a database log
// and/or close any existing connections to the outside world, and
// then exit
}
// this tells PHP which function to run when it receives SIGHUP
// from Asterisk
pcntl_signal(SIGHUP, sig_handler);
// MAIN PROGRAM GOES HERE
?
Like I said earlier, this may or may not solve your problem, but it's
worked well for me with PHP AGI scripts.
--
Jared Smith
Digium, Inc. | Training Manager
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Re: [asterisk-biz] Lost job in this economic risk
On Thu, 2008-11-06 at 02:52 +0100, Trixter aka Bret McDanel wrote: in that case the international trade stuff seemed appropriate to this list given the quantity of goods that are made in china and exported. I do believe even the digim cards are done this way. This is not true. Digium's hardware cards are designed and manufactured in the United States. -- Jared Smith Training Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Returning Error Codes
On Sun, 2008-11-02 at 12:24 -0500, Igor Hernandez wrote: Recently we've had a client come to us requesting a specific error code be returned from our switch because they need it for their LCR. Is there a way in Asterisk to return a specific error code instead of just passing down the error code returned from the carrier's switch? Recent versions of Asterisk allow you to pass a cause code as the first (and only) argument to the Hangup() application. -- Jared Smith Training Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] General development funding: discussion and survey
On Fri, 2008-10-31 at 10:20 -0400, Steve Totaro wrote: A profit margin for Asterisk or Digium? Where is the line drawn here. The line was moved quite a bit with Adwords debacle. John isn't suggesting in any way that this would be for Digium's financial benefit -- he's simply trying to see if there's interest from the community in having an easier way for community members to help fund community development. -- Jared Smith Training Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] noises from Digium tdm410p
On Fri, 2008-10-10 at 13:14 -0400, Leo Soares wrote: Any of you guys ever had a issue with Digium tdm410p card making noises on the line Don't forget that the your card comes with technical support from Digium! If the other suggestions on this thread don't help you solve the problem, I'm sure one of Digium's tech support staff would be happy to help you figure out the cause of the problem. -- Jared Smith Training Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Asterisk Billing System
On Tue, 2008-06-17 at 17:06 +1000, MBIT Technologies wrote: I am looking for an asterisk billing system. If you know of one that has the following specific requirements please let me know. You may want to check out Freeside (at www.freeside.biz). It's a web-based billing platform originally written for ISPs, which over the past couple of year or so been extended to do billing for VoIP as well. It does all the fancy billing features you'd expect, such a pro-rating of services, many types of billing options (monthly rate, per-minute, bundles of minutes, etc.), extensive reporting, and so on. And best of all, it's an open-source project! (And FYI, I'm not related to the project in any way, other than I know one of the guys who helps write it and I've used it a couple of times and have been very very impressed. In other words, I'm an actual customer and not a paid endorser.) -- Jared Smith Training Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Asterisk Billing System
On Tue, 2008-06-17 at 08:09 -0400, Jared Smith wrote: You may want to check out Freeside (at www.freeside.biz). Just to be 100% clear here... this is my own personal recommendation, and not a Digium-endorsed or Digium-sponsored recommendation. -Jared ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Please Remove this email from list : yupee phone
On Tue, 2008-05-13 at 10:47 -0700, Naveed Nazir wrote: Please remove yupeephone from the list We are getting an out of office message for every email on the list... I've unsubscribed him from the mailing list, which should take care of the problem. Let me know if the problem persists. -- Jared Smith Training Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Pulver media in trouble ?
On Sun, 2008-04-06 at 23:02 +0300, Moshe Maeir wrote: In any case I would assume that if Pulvermedia does go under (which I hope not) Digium will find another company to partner with, though that may not save this years show. If Pulvermedia does go under, Digium will continue to have AstriCon -- we'll run it ourselves. As one of the people that came from the Sokol and Associates team, I have firsthand knowledge of what it takes to run the show, and I'm sure we can make it a successful event by ourselves in the event that Pulvermedia goes out of business. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Pulver media in trouble ?
On Sun, 2008-04-06 at 13:57 -0400, Steve Totaro wrote: No, it is not enough. A simple statement from one of the Adtran Guys, Jared Smith, or even Mark Spencer would quickly dispell this misinformation or FUD. Sorry for the slow response, guys... I spent the entire weekend away from email. (Lucky me!) Yes, AstriCon will continue to be held, no matter what happens with Pulvermedia. Just so that everyone is on the same page, let me clarify a few things. This will (would have been?) the first year that Pulvermedia was handling AstriCon. Before then, it was done by Sokol and Associates. (Well, last year we were formally part of Digium by the time the show rolled around, but the same Sokol crew did the vast majority of the work... You get the idea.) I still haven't heard one way or the other what's happening with Pulvermedia, but assuming for a minute that they do fold up shop, I have no doubt whatsoever in Digium's ability to run with the show and make it very successful. I think our track record speaks quite well for itself. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Testing. Hello List!
On Mon, 2008-04-07 at 13:58 -0400, SIP wrote: Same here. Posted at 10:30. Nothing. Emailed the list admin. Nothing. I'm thinking about smoke signals next. Sorry, the list admin (myself!) has been on an airplane for a good part of the afternoon, and hasn't had access to email. As soon as I can get the Internet working in the hotel I'm staying at (in Toronto, of all places), I'll dive in and take a look. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Test (was: Re: asterisk-biz Digest, Vol 44, Issue 13)
On Thu, 2008-03-06 at 12:23 -0500, Matthew Rubenstein wrote: I have had lots of problems sending messages to this list, including requests for offers of products I'm trying to buy. I have talked directly with Jared Smith, Digium's list operator. I still have the same problems. Nothing has changed. This list has some serious basic defects as a platform to do business with. The problem is our spam filter that fronts the list. I've been working to tune it over the past several months, but it still rejects a certain portion of legitimate posts, despite my best efforts. I'll go in and try to tweak it a little more, but I can't guarantee anything at this point. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Digium's Exceptional Satisfaction Program
As many of you may well know, Digium has been investing a great deal of time and effort to build the very best telephony products in the industry. We're committed to producing the highest quality hardware and software solutions, along with things like training and support to make your Asterisk deployment a successful one. As part of this effort, Digium is launching its Exceptional Satisfaction Program. I won't bore you with all of the details here (see links below for more detailed info), but in a nutshell we've extended the warranties on almost our entire line of hardware and commercial software products, and have thrown in a money-back guarantee as well. The blog post announcing the program can be found at http://blogs.digium.com/2008/02/11/digium-puts-its-money-where-its-mouth-is/. The details of the program can be found at http://www.digium.com/ESP. We've also created a FAQ page at http://www.digium.com/en/company/riskfree-facts.php. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Digium's Exceptional Satisfaction Program
On Wed, 2008-02-13 at 15:39 +0100, Trixter aka Bret McDanel wrote: Why not also offer a money back guarantee on the GPL stuff, obviously disqualifying anyone who worked as a middleman? The GPL versions of Asterisk are available for free, so I'm not sure I understand what you're asking for here. Most FOSS projects have money back guarantees. They do? I'm not aware of any that do. Most have a clause that says something to the effect of This program is distributed in the hope that it will be useful, but without any warranty; without even the implied warranty of merchantability or fitness for a particular purpose. In fact, the GPL itself has the following clause: 11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHEN OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES PROVIDE THE PROGRAM AS IS WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU. SHOULD THE PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING, REPAIR OR CORRECTION. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Open source Asterisk billing solution
On Mon, 2008-02-04 at 08:39 -0500, Tom Moore wrote: I've been asked to setup an Asterisk server with accounting functions for a client and his customers. To keep it short and sweet he wants to provide dids to his customers and charge a monthly service fee for the customer having the line and when the time comes offer packages of minutes and have charges be added to the account when the package minutes run out. I know several people doing this with the Freeside billing package (see http://freeside.biz/). Originally written as an ISP billing platform, it handles just about any billing situation you can think of, including month charges, pro-rated months, per minute or block-of-minutes type billing, etc. as well as some advanced features like account provisioning, etc. (A friend of mine is one of the core developers, so my opinion is probably somewhat biased... your mileage may vary.) -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Asterisk Business Edition
On Mon, 2008-02-04 at 11:00 +0700, Dome Charoenyost wrote: I found Asterisk Business Edition supports up to 40 simultaneous calls with upgrades to 240 calls available. in digium web site. What's mean ? Asterisk Business Edition not include source code ? No, Asterisk Business Edition does not include the source code. While it's built from the same source as the open source version of Asterisk, it's licensed under a more traditional commercial binary-only software license. If you chose to go the Asterisk Business Edition route, you get technical support, warrant, and some patent indemnification. If on the other hand you go the open-source route, you get support from the community (mailing lists such as this one, forums, blogs, and so on). The nice thing is, you get to choose which is best for your particular situation. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] How to make TE220B work well?
On Mon, 2008-01-28 at 18:59 +0800, [EMAIL PROTECTED] wrote: I used trixbox 1.2.3 with a E1 card(not digium).The call often disconnected,and the echo is great. Recently, I bought a E1 card from digium --TE220B. It can not work on trixbox 1.2.3. The te200series-user-manual says the card need asterisk 1.2.20 or newer and zaptel 1.2.20 or newer, so I update trixbox1.2.3(asterisk 1.2.12 and zaptel 1.2.9) to trixbox 2.2 (asterisk 1.2.24 and zaptel 1.2.21).But the disconnected rate increased much,the sip_phone can not be used. I don't know if I chose a wrong release ,or I don't use the card correctly. I want somebody can give me some clews. Thank you in advance! Do you have callprogress=yes and/or busydetect=yes in your zapata.conf file? If so, turn those settings off for your E1 channels. They're the most likely cause for disconnects during a call. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Billing solutions?
On Tue, 2008-01-22 at 13:10 -0800, Nitzan Kon wrote: I've seen a lot of open source billing systems out there, some of them look really nice, but they're mostly geared towards calling card usage and it would probably take a lot of effort to convert them to my needs. If I were you, I'd check out the open-source Freeside project. They seem to be able to handle just about anything related to billing, and I know that the authors have been putting a lot of time into VoIP billing features. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Digium Relaxes Google Adword Policy
On Fri, 2008-01-18 at 18:03 -0600, Danny Windham wrote: Over the past week Digium has received a number of charged responses regarding the recent change in policy regarding use of Digium trademarked terms in Google AdWords. Some of the responses supported our attempts to better control the use of Digium trademarked terms. Some of the responses disagreed with the policy, but respected Digium’s right to have changed the policy. Others were from individuals who clearly were unhappy with the change and the process by which it was implemented. We have listened carefully to the feedback, and as a result are relaxing our Google AdWord policy. For your information, we've put together a list of frequently asked questions regarding Digium's policy towards Goodle AdWords and our trademarks. The FAQ is currently available on our corporate blog at http://blogs.digium.com/ for those that are interested. If you have any further questions or concerns, don't hesitate to email me. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Digium enforcing its trademarks
On Mon, 2008-01-14 at 06:33 -0800, Flavio Goncalves wrote: I have received today a notice of disaproval of my ad in Google Adwords, because it was using the word Asterisk (A Digium´s trademark). In response to an increase in the occurrence of unauthorized use of Digium trademarked terms, Digium has recently expanded our efforts to control the use of Digium’s trademarks. One the measures taken has been to work within Google’s defined programs to control the purchase of Digium trademarked terms as Google Adwords. In doing so we proactively provided Google with a list of companies that have signed agreements with Digium permitting the use of Digium trademarks. Organizations on that list will continued to be allowed by Google to purchase Digium trademarked terms as Google Adwords. However, if your organization has been recently been contacted by Google and required to stop using the Digium marks, you may contact Digium directly to request that your company be added to the list of authorized organizations. For those organizations that are resellers and partners, that are not formally Digium authorized, why not sign up and become a reseller or technology partner? To do so please visit http://www.digium.com/en/ecosystem/ and click through to your selected type of partnership. If the formal Digium programs don’t fit your situation, you may send a request to the [EMAIL PROTECTED] address with some history of your past use and the specifics of your request for the future. If you are unsure of the Digium trademark policy, please visit http://www.digium.com/en/company/view-policy/5. We will attempt to turn around all requests within 24-48 business hours. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Whats New at Digium the Asterisk Company
On Tue, 2007-11-13 at 10:33 -0800, shadowym wrote: Didn't Mark Spencer recently criticize Fonality's Hosted business model? I think there's some confusion here on exactly what the new Switchvox product is, so I'd recommend that people read the press release at http://www.switchvox.com/sv?page=press_room/hosted_svx for more detailed information. In short, Switchvox (now part of Digium) has designed a package that ITSPs can purchase and use to deploy a hosted-PBX solution in their own data centers. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] SIP Fax OK?
On Tue, 2007-08-07 at 13:38 -0400, Andrew Joakimsen wrote: Yes, if all the SIP endpoints support ITU-T Recommendation T.38. The problem is there is no such thing right now in Asterisk and no plans by Digium to add it. This is not quite true... just because there is no T.38 endpoint support in Asterisk doesn't mean Digium isn't planning to add it. I have a pretty good feeling that if you're patient, you'll see Asterisk support T.38 (and not just T.38 passthrough) sometime in the future. Unfortunately, I don't know any of the details of when that will be. If you have a SIP provider supporting T.38 then you can have calls from your T.38 capable devices forwarded to that provider, but you must use Asterisk 1.4 and pray it works. The T.38 passthrough support in Asterisk 1.4 has worked for me the few times I've tried it with endpoints that support T.38. (And it certainly worked better for me than trying to fax over SIP/RTP using the G.711 codec and hoping that jitter didn't cause your fax to fail.) -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Checking Carrier Reliability?
On Fri, 2007-08-03 at 10:22 -0700, Douglas Garstang wrote: What about dead air? How do you check for dead air? What about quality? If we're talking SIP traffic, one suggestion would be to look at the RTCP traffic coming back to you from the carrier, and see what type of jitter/packet loss/out of order packets you're seeing. Obviously some (or maybe even most) of that is beyond their control if the packets are going across the internet, but it can be a useful tool to use. Obviously IAX doesn't use RTCP to report that information, but it still comes across in the IAX packets nonetheless. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] AstriCon -- Last chance to speak!
We're getting close to wrapping up the final list of speakers for AstriCon, but wanted to give the Asterisk community one more chance to speak up and be heard. If you're interested in presenting at AstriCon, please go to http://www.astricon.net/ and click on the Speak at AstriCon link on the right-hand side of the page. If you know of someone who we should invite to speak, we're open to suggestions as well. Feel free to email me (off the list, please!) with your recommendations. I'm looking forward to a great conference, and hope to see you all there! -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] IP Phone/Hard Phone
On Thu, 2007-07-19 at 12:17 -0600, Seysan wrote: One more question: Are the Polycoms or Linksys or SNOM phones are AutoCodec? I mean let say there is three HARD Phones by Polycom for example, and 1 Softphone that only supports G711 or GSM codecs, But the Default codec in Polycom is set to G729, Without Transcoding or any Codec Conversion by the Asterisk, can polycom automatically switch to G711 to establish the call ? Yes, they will automatically negotiate a common codec as part of the call setup process, if possible. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] So Digium bought Sokol/Astricon
On Tue, 2007-07-17 at 20:33 -0400, Andrew Joakimsen wrote: They were probably mad of the Astri name... You know how Digium can be when it comes to the use of their names such as Asterisk. Being one of the former employees of Sokol Associates, I may have a little more insight than most into this. (I don't pretend to know everything involved, as I wasn't involved in the discussions about Digium acquiring us.) As far as I know, Digium never had any problem with the AstriCon name for the conference.[1] I think they simply saw this as an opportunity to augment both their marketing and training departments through both the AstriCon conference and the Asterisk training classes, as well as hire some pretty big names in the Asterisk community. (I'm not saying that to toot my own horn in any degree.) I think above all it was just good fit. In short, I think you're safe to take off your tinfoil hats :-) Now, as part of my new role at Digium, it's my job to help keep Digium focused on the community and making sure that the needs of the community are being met, as well as representing Digium back to the community. If you have questions or concerns about Digium as a company or with Digium products or services, please let me know and I'll pass those along, and try to make sure you get a response. -- Jared Smith Community Relations Manager Digium, Inc. [1] I'm no lawyer (and I don't play one on TV), but my limited understanding of trademark law is that you're required to defend your trademark or you lose it, so I can completely understand why Digium defends the Asterisk trademark. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Maximum Load
On Wed, 2007-07-18 at 16:40 -0400, Matthew Rubenstein wrote: If the Asterisk community (eg. the asterisk-[users,biz] lists, user groups, resellers, etc) wanted to register our interest in these benchmarks, and even offer help producing them (in an organized way), who at Digium would we contact? Is there anyone inside Digium who is interested in seeing them produced, who we could help in the campaign? That person would be me. I'll go ahead and pass this along to the appropriate people inside Digium and see if we can't get some traction for some publishable benchmarks. Feel free to contact me with the types of benchmarks you'd like to see done, and I'll see what we can do. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
