Re: [asterisk-users] Good comparisons on cheaper VOIP phones
Rob Hillis wrote: Michael wrote: My experience with Grandstream is that are one of the better 'cheap' ones, but cheap non the less. I am yet to run into a worse IP phone than the Grandstreams - although having said that, I should say that I've always steered clear of most of the Chinese no-name brand phones. They're unstable, temperamental and upgrading the firmware is a crapshoot half the time since you never know what new bugs will be introduced and quite often you can't downgrade the firmware if you don't like the newer firmware. My suggestion would be to look at the Snom 300 (although they are very simplistic phones), the Polycom IP330 (I have a feeling the 320s don't support PoE) or the Linksys phones. I noted an earlier post saying that these phones were overpriced and designed to lock you in to Linksys gear - my experience has been completely different. The SPA-942 is quite cheap and integrates nicely with Asterisk. The SPA-962 is considerably more expensive - but considering the size of the colour LCD screen, they're not that badly priced. (as an aside, the button banks for the SPA-962 are one of the /cheapest/ available!) ___ -- 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 having deployed a fair amount of phones I have the following observation (and these observations are worth what you paid for them :-) ) 1. Linksys 942, my preferred mainstream desk phone, a bit more expensive than the Polycom IP330. Be careful as there are two SKUs with and without power supply (which is true of the ip330). The 942 has a nice large backlit screen, nice big MWI light, takes a 3.5mm headphone. With latest firmware, now supports BLF and LDAP. 2. Polycom IP330 (320 is same, except it doesn't have the 2port ethernet switch built in PC port). Biggest dis advantage is no voicemail key, you have to assign a speed dial to Line2 (there is a plastic VM key you can use, but you have to swap the plastic key overlay). 3.5mm headphone jack. 3. Grandstream phones... they work with quirks for call xfer and conference as an example. The phone buttons don't have a business feel to them my 2 cents... jim (www.sigma-networks.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] AsteriskNOW Web GUI
I am a long time user and reseller of Thirdlane PBX Manager. From my standpoint the implementation tools are outstanding and the fact that the files are easy to follow means it allows a consultant to comstomize the behavior yet allow the end user to maintain going forward. good luck! Steve Totaro wrote: The README is here: svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow /Configuration = You may install sample configuration files by doing make samples. Also you will need to edit your Asterisk configuration files to enable the GUI properly, specifically: 1) In http.conf: [general] enabled = yes enablestatic = yes/ I am looking at Thirdlane's solution now. Very impressive and modest cost. Thanks, Steve bkruse wrote: As Tzafrir stated, it will NOT work with 1.2.x. Where is this html.conf, which README? I will update it. I will write a brief page on setting up the *GUI for all who want to know.. There are SOME GUI's that work with 1.2, however, I almost guarantee none of them are client side, such as this one. -bk Steve Totaro wrote: Will this work on 1.2.x? I just installed it and did make samples. The README references a file called html.conf which does not exist and also abruptly ends with the word to on a blank line. Besides that, what would the URL be for AsteriskNow? Is that customizable in the elusive html.conf file? Any GUIs that are easily installed on existing systems and work with 1.2.x? Thanks, Steve bkruse wrote: svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow thegui; cd thegui; sh configure; make sudo make install ; clear ; echo 'completed' -bk Yann JOUANIN wrote: You can do it from svn server , I think there is a page in the wiki Best, yann *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Jeremy Mann *Envoyé :* vendredi 24 août 2007 17:30 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* [asterisk-users] AsteriskNOW Web GUI Is the web GUI for AsteriskNOW able to be loaded on an existing server(that was installed from ubuntu-server and asterisk loaded from source)? ___ --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] multiple pbxes, multiple domains, same user ids?
You may want to consider the multi-tenant version of Thirdlane's PBX Manager (www.thirdlane.com). I've been using for a long time and very happy with both single and multi-tenant versions. Benjamin Jacob wrote: Anthony Francis wrote: Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the "domain" field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk maintains the registration of the latest registree!! thats really sad for me . Any work around for this one(multiple pbx)? I would be zapped and amazed if multiple pbx isn't possible in Asterisk. Help anyone? cheers - Ben. you have to do different sip-ids, I am guessing you are probably using the extension #, you dont need to do that. What do you mean by multiple-pbx's anyway? I hope you don't mean multiple instances of *.What I am sure you mean is multiple dial plans, and yes, * is multi-tenant friendly. What we do for uniqueness is use the last 8 digits of the device mac addr or other unique number followed by a dash "-" followed by the extension number. Anthony Thanks Anthony. I definitely don't mean multiple instances of asterisk. Multiple dial plans, hmm.. yes.. in a way. Multiple pbx ... in short, provide pbxes for two entirely different organizations, say, Microsoft and IBM (can i use these names in here? ;-) ). Each would have many extensions, but each office can have identical extensions, e.g. you can have extensions 4001 in both. But one would be [EMAIL PROTECTED] and the other would be [EMAIL PROTECTED] . [EMAIL PROTECTED] should be able to call any user within Microsoft. To step outside the organization, you would put in some logic(dialplans). So, i want to have pbx for microsoft and another pbx for IBM. Is it possible to have two or more pbxes within one Asterisk instance. Hope you got my point. cheerz - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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
[Asterisk-Users] help with compiling addons for cdr
Im running Asterisk 1.0.7. Ive checked out using cvscheckoutasterisk-addons. When I make install I get the following errors: app_addon_sql_mysql.c:162:36: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given Im using the default FC3 mysql: mysql-server-3.23.58-16.FC3.1 perl-DBD-MySQL-2.9003-5 mysql-3.23.58-16.FC3.1 mysql-devel-3.23.58-16.FC3.1 php-mysql-4.3.11-2.4 libdbi-dbd-mysql-0.6.5-9 MySQL-python-0.9.2-4 Ive search wiki, etal and have found a couple of references with a proposed patch file, but the patch file fails too. I would appreciate any assistance. ___ 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] Grandstream Quality Survey.... :P
Brian, Can you compare Ford and Mercedes or BMW? Both are cars and drives.. but you have different feeling and price in/for each car ..same here Grandstream is low-cost solution for end-users/small business , Cisco IP Phones are couple times more expensive ,but they have more features, less bugs and more fancy. Also, don't forget that Grandstream is muuuch smaller company compare to Cisco and they are new company, they have much less customers/phones sold out then Cisco, so it takes time to find all bugs and fix them, also to release new firmware. We had conversation with Grandstream how to improve there phones, so they are working on it. I am sure in 2004 GS will go high and we will have less probs with them. We are looking now to improve GS products and start collecting all bugs/probs and send them to GS. Idea is that we are opening Online forums and special Grandstream products mailing list. Some support people from Grandstream will be participating in Forums and Mailing lists, so we will have direct communication between GS and Online community, hopefully it will help us to solve more probs. Grandstream is very interested to make nice product and sell more, so they will be fixing bugs for sure, otherwise they will be out of business. Grandstream forums URL : http://forum.xvoip.com/viewforum.php?f=7 Grandstream products support mailing list: [EMAIL PROTECTED] Regards, Alexander -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, December 24, 2003 10:32 AM To: Asterisk List Subject: Re: [Asterisk-Users] Grandstream Quality Survey :P Yes when you upgrade to beta code you may have to reboot 3 times for the phone to function properly. Then cross your fingers that the phone will accually register with * once you do that. bkw On Wed, 24 Dec 2003, Dave Cotton wrote: On Wed, 2003-12-24 at 15:50, WipeOut wrote: I just loaded the b13p4.30.zip firmware and now I am not able to log into the GS admin interface.. anyone else having this problem? Yep been there. Panicked, rebooted the phone and it responded as normal. I just tried it again because of your question and had to reboot again to get in. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Quality Survey.... :P
Robert, We are going to deploy GS phones in our free voice network, therefore we require somekind of web-presence, which will reflect GS support,etc. Unfortunately not all of our users are subscribed to Asterisk mailing list. Acting as GS distributors, we are making separate forum for this, which doesn't belong to Asterisk. My postage about new forum was just as information only about new resource. And I assume asterisk mailing list if primarily designed to Asterisk support and not Grandstream phones... Also Grandstream phones are being used in different platforms too, not necessarily only with Asterisk, this is why they are looking for separate resource. In all cases, I will be posting here copy's of interesting messages/infos from Grandstream, so we all know what's going on. And I agree with your comment on BETA firmware. I assume people have to understand what is Beta release and what is stable official release of software. Of course by using Beta software, which is not approved officially and launched bugs will appear. Regards, Alexander rnc Info Lists Sent: Wednesday, December 24, 2003 12:05 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P ... Alexander, I agree with your email but setting up MORE forums and mailing lists is not productive. GS phones have problems interacting with the VoIP services and Asterisk. The BEST places for the GS folks to get feedback AND to interact with the people who are using their phones are on these already existing mailing lists. I don't know why you insist on creating even more websites/email lists for VoIP support. Why not encourage GS to get visible on these lists and interact with their customers here, where they can get the most concentrated feedback (good and bad). Also, a comment for the general list. To me BETA code means that it is NOT yet RELEASED as PRODUCTION code. For anyone to think that Beta code comes without problems is being a bit shortsighted. If you get beta code that works without problems then that is great, otherwise give the developer feedback so that he can fix the bugs and don't complain about the problems it caused you. Otherwise wait on the official production releases. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Merry Christmas and Happy New Year from XVOIP
Dear All, On behalf of XVOIP, LLC/Stealth Telecommunications, WISH YOU A MERRY CHRISTMAS AND A HAPPY PROSPEROUS NEW YEAR. Thanks to everyone for such great place as Asterisk community, for all your answers, suggestions, time, examples, help. We plan to support Asterisk project and in 2004 we will be launching new projects, which will include: 1700/777 access number from PSTN to IAXTEL in US/Canada (free access), Personal LCR (least cost routing engine), IAX/XVOIP Exchange. US DIDs in 40 US States and Ontario Province, Canada. All announcements will be posted before New Year to [EMAIL PROTECTED] list and asterisk-users mailing lists. Asterisk mailing-list members will be receiving special bonuses and will have priority on our network. Asterisk is an International community and let me try to congratulate you in your own language (See below ;-)). MERRY CHRISTMAS AND HAPPY NEW YEAR! Alexander Kandelaki Afrikaans - Geseknde Kersfees en 'n gelukkige nuwe jaar Argentine - Feliz Navidad y Feliz Año Nuevo Bohemian - Vesele Vanoce Brazilian - Boas Festas e Feliz Ano Novo Bulgarian - Vesela Koleda i chestita nova godina! Catalan - Bon Nadal i un Bon Any Nou! Chinese - Sing Dan Fae Lok. Gung Hai Fat Choi (Cantonese) Chinese - Shen Dan Kuai Le Xin Nian Yu Kuai (Mandarin) Chinese - Shen tan jie kuai le. Hsin Nien Kuaile Croatian - Sretan Bozic Czech - Stastne a vesele vanoce a stastny novy rok! Danish - Glaedelig Jul og godt nyter Dutch - Vrolijk Kerstfeest en een Gelukkig Nieuw Jaar Dutch - Prettige kerstdagen en een gelukkig nieuw jaar English - Merry Christmas and a Happy New Year Eskimo - (inupik) Jutdlime pivdluarit ukiortame pivdluaritlo! Esperanto - Felican Kristnaskon kaj Bonan Novjaron! Estonian - Rõõmusaid jõulupühi ja head uut aastat! Faeroese - Gledhilig jol og eydnurikt nyggjar! Filipinos - Maligayang Pasko Finnish - Hyvää joulua ja onnellista uutta vuotta! Flemish - Zalig Kerstfeest en Gelukkig nieuw jaar French - Joyeux Noel et Bonne Année! Scots Gaelic - Nollaig chridheil agus Bliadhna mhath yr! Galician - Bo Nadal German - Frohe Weihnachten und ein gl|ckliches Neues Jahr! Greek - Hronia polla kai eytyhismenos o kainourios hronos Greek - Hronia polla ke eftihismenos o kenourios hronos Hausa - Barka da Kirsimatikuma Barka da Sabuwar Shekara! Hawaian - Mele Kalikimaka ame Hauoli Makahiki Hou! Hungarian - Kellemes karacsonyi uennepeket es boldog ujevet! Icelandic - Gledhileg jsl og farsflt komandi ar! Indonesian - Selamat Hari Natal dan Selamat Tahun Baru! Iraqi - Idah Saidan Wa Sanah Jadidah Irish Gaelic - Nollaig Shona duit Irish Gaelic - Nollaig Shona Irish Gaelic - Nollaig faoi shean agus faoi shonas duit agus bliain nua faoi mhaise dhuit! Italian - Buon Natale e Felice Anno Nuovo! Japanese - Meri Kurisumasu soshite Akemashite Omedeto! Latin - Natale hilare et Annum Faustum! Latvian - Priecigus Ziemsvetkus un Laimigu Jaungadu! Lithuanian - Linksmu Kaledu Maltese - Nixtieklek Milied tajjeb u is-sena t-tabja! Modern Greek - Kala Christougenna kai evtichismenos o kainourios chronos! Norwegian - God Jul Og Godt Nytt Aar Pennsylvania German - En frehlicher Grischtdaag un en hallich Nei Yaahr! Polish - Vesowe Boze Narodzenie Polish - Wesolych Swiat i Szczesliwego Nowego Roku Portuguese - Boas Festas Portuguese - Feliz Natal e um Prospero Ano Novo Romanian - Craciun fericit si un an nou fericit Russian - S nastupaiushchim Novym godom i s Rozhdestvom Khristovym! Romanche - (sursilvan dialect): Legreivlas fiastas da Nadal e bien niev onn! Serbian - Hristos se rodi Slovakian - Sretan Bozic or Vesele vianoce Slovak - Vesele Vianoce i na zdravie v novom roku! Slovenian - Vesele bozicne praznike in srecno novo leto Spanish - Feliz Navidad y Próspero Año Nuevo Swedish - God Jul Och Ett Gott Nytt Ar Thai - Suk san wan Christmas Thai - Suk san wan pee mai - Happy New Year Trukeese - (Micronesian) Neekiriisimas annim oo iyer seefe feyiyeech! Turkish - Noeliniz kutlu olsun ve yeni yilinis kutlu olsun! Turkish - Noeliniz Ve Yeni Yiliniz Kutlu Olsun Ukrainian - Srozhdestvom Kristovym Ukrainan - Z novym rokom i s rizdvom Hrystovym! Ukrainan - Khrystos Rodevsia Vietnamese - Chuc mung nam moi va Giang Sinh vui ve Welsh - Nadolig Llawen a Blwyddyn Newydd Da! Yoruba - E ku odun, e ku iye'dun! image001.gif
RE: [Asterisk-Users] Port density: DS3 cards?
Lucent TNT box price is attractive, but based on real experience it is not very VOIP friendly. You have to consider it. It is hard to interconnect with Cisco for example. I have no idea about Max TNT-Asterisk interconnection. We are using Nextone softswitch and able to serve clients and interconnect via Cisco's to Max TNT only via NExtone, but direct interconnect Cisco-MaxTNT almost impossible. However, if you are using TNT's on both terminating/originating ends, then it is extremely great solution. Regards, Alexander -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Sent: Thursday, December 04, 2003 5:51 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Port density: DS3 cards? At 02:34 PM 12/4/2003, you wrote: However, considering the traffic volumes that you are talking about, is it really true to say that the traditional telco cards are astronomically priced, given the amount of revenue that can be generated per month on a DS3? Eight quad-span T-1 cards from Digium: $8,970 Three reasonable-quality asterisk servers: $1,000 One T-1/DS-3 MUX: $5000 Total system cost: $14,970 That actually sounds quite reasonable to me. However, if I were doing this myself I would look hard at getting a MAX TNT with VoIP capability off eBay. The price would be equivalent or less, the interface would be more complicated, but all the DSP would be done by the MAX. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] multiple simultanous calls with iconnecthere/delta three
Todd, We can pass multiple calls simultaneously, no problem. I can't contact you off-list because your email address is masqueraded by Mailing list. Please contact me at [EMAIL PROTECTED] Regards, Alexander -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd Wallace Sent: Thursday, November 20, 2003 6:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] multiple simultanous calls with iconnecthere/delta three has anyone configured asterisk to talk to iconnecthere (or any carrier) and have more than one line in/out. I want to be able to allow for multiple simultaneous calls. If so what product did you buy if not, is there any other carrier that can do this?? Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mailing list email masquerading.
Why mailing list info was changed and no information about Senders email address in available anymore? Is this fight with spammers? It means from this moment, if we want to reply someone off-topic/off-list we cant do it, because we dont see Senders email address, only name of person and we only possible way to do it , via mailing list. Thanks, Alexander E-Mail: [EMAIL PROTECTED] * Unofficial Asterisk Forums : http://asterisk.xvoip.com *
RE: [Asterisk-Users] Dialup Internet through Asterisk server?
Matt, Of course it is possible, PPP server on Linux for remote access is simple and so difficult to implement, and it has nothing to do with Asterisk. You have to deal with Linux config for PPP in order to setup dialup server. If you need some assistance let me know, I can point you to some resources on Net. Regards, Alexander -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Thursday, November 20, 2003 7:08 PM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Dialup Internet through Asterisk server? Hello, I was wondering if it was easy to setup a dialup internet account system in Asterisk somehow(like PPP) laptop with modem - PSTN line - Asterisk server - Server's broadband internet connection Has anyone done this? Thanks, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [Asterisk-Users] Mailing list email masquerading.
Please disregard previous posting. My mail client is nuts .. Thanks, Alexander -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk online forums Sent: Thursday, November 20, 2003 7:48 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Mailing list email masquerading. Why mailing list info was changed and no information about Senders email address in available anymore? Is this fight with spammers? It means from this moment, if we want to reply someone off-topic/off-list we cant do it, because we dont see Senders email address, only name of person and we only possible way to do it , via mailing list. Thanks, Alexander E-Mail: [EMAIL PROTECTED] * Unofficial Asterisk Forums : http://asterisk.xvoip.com *
[Asterisk-Users] Asterisk Business discussion again
Hello all, Last couple weeks we had a lot of business discussions on mailing list, however some people don't like it, some people don't needed it, etc. I had couple discussions with Asterisk community members, who is interested to have business discussions about Asterisk, including but not limited to : business implementations, reselling , Asterisk commercial packages, IP phones, Asterisk One Stop Shop solutions, Wholesale termination, providers list, etc. Idea to which we come up with some members of our Asterisk community is to create Asterisk Elite Business mailing list. Which will be unofficial Asterisk list. This list will be moderated by couple moderators and will be really Elite List, we will not allow to access it spammers ,etc. During next couple days, we will publish some draft about our vision for Business implementation for Asterisk and related projects. We welcome anyone who is interested in business discussions about Asterisk and solutions based on Asterisk to join this Elite Business Asterisk List. We are also looking for people, individuals, independent consultants who has deep knowledge of Asterisk from technical side. If youfeel serious about Asterisk, about great IP PBX software, please join us. Please be informed, it is not one more technical mailing list, we are not going to discuss all technical issues, main topics will be related to business, marketing, sales. It is just unofficial Business discussion list. Please send inquiries to [EMAIL PROTECTED] to get registered. Also we will be publishing archive and information from businesses-list in our Private Forum , on http://asterisk.xvoip.com under Asterisk Elite section, which will be hidden from people who is not registered. Please feel free to join us : [EMAIL PROTECTED] Regards, Alexander Unofficial Asterisk ForumsURL : http://asterisk.xvoip.comRegistration is : http://asterisk.xvoip.com/profile.php?mode=registerNew XVOIP network , get your +1 777 number today. [EMAIL PROTECTED]
Re: [Asterisk-Users] Asterisk Business discussion again
Adam, We had discussions about business list or business implementation for really long time, now it is time to move on and go forward. Hopefully with help of business -oriented people from Asterisk community we will move this project forward. Also we will support Digium on it. Anyway, let's see how it will go. Regarding name, I agree ok , to don't make confusion or something like that, we will rename it to [EMAIL PROTECTED] All requests which came to old address will be added to list too. There is no need to resend existing registration requests!!! We will post additional notes later on Forums. So people who wants to join can use now [EMAIL PROTECTED] as a subscription address. Also forums, online presence will make big sense for it. Thanks, Alexander Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 7:22 PM Subject: Re: [Asterisk-Users] Asterisk Business discussion again Prehaps a new mailing list on list.digium.com if there's really a need for it, otherwise I bet it won't catch on. Also IMO, the word elite != elite, it means script kiddies who wish they were. good luck regardless, Adam The following is the original post, removing HTML seemed to remove the '' Hello all, Last couple weeks we had a lot of business discussions on mailing list, however some people don't like it, some people don't needed it, etc. I had couple discussions with Asterisk community members, who is interested to have business discussions about Asterisk, including but not limited to : business implementations, reselling , Asterisk commercial packages, IP phones, Asterisk One Stop Shop solutions, Wholesale termination, providers list, etc. Idea to which we come up with some members of our Asterisk community is to create Asterisk Elite Business mailing list. Which will be unofficial Asterisk list. This list will be moderated by couple moderators and will be really Elite List, we will not allow to access it spammers ,etc. During next couple days, we will publish some draft about our vision for Business implementation for Asterisk and related projects. We welcome anyone who is interested in business discussions about Asterisk and solutions based on Asterisk to join this Elite Business Asterisk List. We are also looking for people, individuals, independent consultants who has deep knowledge of Asterisk from technical side. If you feel serious about Asterisk, about great IP PBX software, please join us. Please be informed, it is not one more technical mailing list, we are not going to discuss all technical issues, main topics will be related to business, marketing, sales. It is just unofficial Business discussion list. Please send inquiries to [EMAIL PROTECTED] to get registered. Also we will be publishing archive and information from businesses-list in our Private Forum , on http://asterisk.xvoip.com under Asterisk Elite section, which will be hidden from people who is not registered. Please feel free to join us : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multi call iconenct?
http://www.xvoip.com Call as many as you want ;-) Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 11:57 AM Subject: Re: [Asterisk-Users] multi call iconenct? http://connect.voicepulse.com - Andrew Thompson - Original Message - From: Shoval Tomer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 10:33 AM Subject: [Asterisk-Users] multi call iconenct? Is there a service like iconnect that does allow dialing out more then one concurrent connection? Asterisk works great with iConnectHere, but they only allow one call at a time. I don't want to setup an account for each concurrent call, because it will make iConnect an expensive service, and besides, all of our calls combined doesn't reach 1000 minutes per month. Any ideas? Ë^®+$RÇ«²f¢-)à-+-Ë^®+$RÇ«²X¬¶Çb,+¦r?¡¶ÚþX¬¶Çb,+¦r?¿T¨¥T©ÿ-+-Swèý«-z¸¬'ë ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
Linus, We started this list on Forums : http://asterisk.xvoip.com So any body can post info about services, etc which are off-topic for this list.. Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: Linus Surguy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 12:08 PM Subject: Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) I can't remember who it was, but someone on this list was aiming to compile a list. We certainly replied, offering UK rest of world IAX (and SIP) termination. If that project isnt happening, it would be a great idea if someone else wanted to take it up. Linus - Original Message - From: Steve Sobol [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 3:31 PM Subject: Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) Low, Adam wrote: We can offer SIP based VoIP call termination in The Netherlands, Austria and Norway. If you'd like to speak to an account representative please contact me personally by email. Hmmm, this information should be on a website somewhere... -- JustThe.net Internet New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic)
DONE. Any other suggestions ? Please let me know if we need to add some additional sections. Any suggestions are welcome. Thanks, Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: Linus Surguy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 1:16 PM Subject: Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) - Original Message - From: Asterisk online forums [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 5:48 PM Subject: Re: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) Linus, We started this list on Forums : http://asterisk.xvoip.com So any body can post info about services, etc which are off-topic for this list.. Just a suggestion, but under 'Service Providers' why not start a subject 'Other Providers'? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * VOIP Terminator
Our forums are opened for anykind of discussions, including business opportunities, termination providers list, etc. Let's move some stuff to forums, so people who is interested can find and exchange information. Please join, at least we will empty this mailing list from some busienss discussions, because some people are just technical and they are not interested in any kind of commercial info so let's move on. We are also will be discussing commercial implementations for Asterisk projects. Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 12, 2003 8:05 PM Subject: Re: [Asterisk-Users] * VOIP Terminator I don't believe that this is particularly relevant to the Asterisk software -- perhaps another list can be created for discussion of what commercial services may or may not exist? On Thu, Nov 13, 2003 at 08:02:12AM +, Isamar Maia wrote: I am looking for a brazilian VOIP terminator for the states of Sao Paulo, Parana, Para and Bahia. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
As we know market has thousands of great free source products, but somehow most of companies are buying commercial software and paying a lot of $$$ . Question becomes why they need to pay so much money for something what can be taken for free ? Also, why all these software products are so expensive ? For example voip billing software from MIND CTI, it cost more then 100K . However you can find a lot of individuals on internet (read message boards, posts, forums) who can create normal billing software and it will cost you ... 1K.. Yes , 1 K . So why not to buy it for 1K ? Why to spend thousands of dollars ?? Software itself cost 0$ if it has no support, like in case of example of individuals who is selling 1K programs. If you spend 100K to software you know that it has : support, normal documentation, updates, patches, and in most cases it is Ready-to-Go solution. If you need some kind of customization you can get one from manufacture too, sometimes it will cost more $$$, but still you will get what you need. In order to provide support, documentation, marketing materials, etc software companies are offering high cost for software. Every company has 2 major departments sales/marketing and technical. We are in Open Source World (OSW) here, I would say we belong to technical department in this case. We just don't have sales/marketing department for Asterisk software in OSW. Let's take a look , who are members of Asterisk community : programmers, developers, engineers,sys. admins, net admins - 99.9% Sales/marketing guys they don't know what is UNIX/Linux, Open Source, Zaptel driver, XP100, Kernel . They have no idea about it. So, they are coming here let's say and look into mailing list archive. What they see ? kernel, driver compilation problem, zaptel info, etc. Yes, nothing else. Where is asterisk-business mailing list ? Where it is ?? When I started unofficial asterisk forums, I thougth people will come to discuss business issues and solutions. Take a look, we have 99.9% technia questions postings ...(http://asterisk.xvoip.com) Let's take a look into our engineers, developers, programmers .. They are just great guys , with excellent knowledge, experience .. mostly in technical field. It is hard for regular developer to create proposal about IP PBX/Asterisk and to bring it to management for discussions/implementation. In 99% management will refuse proposal, even without reading it. But if they will start reading it, what they see ? Guess .. zaptel driver, kernel, unix, xp100.. Proposal will go to garbage can. Everybody has its own responsibilities within company. Technical guys/developers has nothing to do with Marketing/Sales and this is why we don't see Asterisk today in slashdot or in New York Times. We are missing marketing/sales actions to bring Asterisk to community, to companies. And I agree 200% with Adam and John about it. For Linus Tovalds it took couple years to make people to believe into Linux . Today, we can see his results, IBM, Dell, Compaq everybody is using Linux, it is known , very well known and end result of his Free Open Source campaign for Red Hat Linux is next: Red Hat becomes commercial software (commercial support per workstation.etc). Asterisk IP PBX project is great !! It has so many features, it is fantastic, but ... it is missing normal business/marketing/sales part. Without it , it will be very hard to bring it to community, to make it very well known. We need Asterisk Pro commercial package to be developed. We need to create nice documentation, faq, knowledgebase. Good software packaging. One more example about packaging, you can buy so called clone for X100p card and imagine it has beautiful colored box, CD , 30 pages colored nice manual and all these for 10$. If I will take X100p card from Digium and clone card ... I will choose clone card. So others will do. Asterisk Project is free source project and for someone who is using it at home environment and who is developer it is fine, they don't need anything else. But majority of people here represent companies, they can bring Asterisk to company as a solution. They only need help to do it. We can help them to do it, it will increase Digiums sales and will bring some profit to community participants. I think it is time to start commercial Pro version (not expensive !!!) of Asterisk. In my company we already made decision to do it, to offer people ready-to-go solution. But is is hard to do anykind of such product without Digium and Mark's support. Mark I think you are very overloaded with all projects, maybe we can help with Asterisk project. Asterisk Basic will stay as it is now, but we will be developing Asterisk Pro. This community has excellent talented people, just go to IRC and participate in chat .. You will see how helpful are people. And everything is free.. Some of guys spends couple hours to fix someone's problem, to write to someone AGI script, etc.. and everything
Re: [Asterisk-Users] IBM to Run VoIP On Linux
Robert, You are right about licensing issues. But I have mentioned in my email that we need direct support from Mark and Digium. It might be direct license or something else. But we need Digium and Mark to participate in this project. Now about commercialization : Idea is to create enhanced version of *, which will include customer support, documentations, manuals, new additional services (like voice termination), etc. Asterisk as it is now will stay as it is.. But enhanced version will be commercial, and again we need to calculate all fees and expenses involved. But commercial version will help community and Digum and all of us to make nice famous product. For someone who can't afford fees for technical support we will open Asterisk Knowledgebase .. my idea is to have it in normal way and open... But to create one, to make it nice and informative we need funds. It is clear that Digium is not going to sponsor the project (maybe I am wrong, this is why I need some feedback from Mark). so we need to find funding by ourselves. My company is willing to participate in project and ready to bring some parts into it. Once again, I want to make clear one thing. Nobody is talking about complete commercialization of Asterisk, I am talking only about enhanced version. I can bet that we will have customers willing to pay money to have system with enhanced features, support ,etc. We just need to start it. If any questions ,let's discuss it. Also thank you Robert that you published your info on forums, for someone who is not part of mailing list it bring information :) Thanks, Alexander Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: rnc Info Lists [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 08, 2003 5:55 PM Subject: Re: [Asterisk-Users] IBM to Run VoIP On Linux I think it is time to start commercial Pro version (not expensive !!!) of Asterisk. In my company we already made decision to do it, to offer people ready-to-go solution. But is is hard to do anykind of such product without Digium and Mark's support. Mark I think you are very overloaded with all projects, maybe we can help with Asterisk project. Asterisk Basic will stay as it is now, but we will be developing Asterisk Pro. Correct me if I am wrong, but unless you have a license from Digium directly then you must sell your Pro version software under GPL. What you do for documentation/packaging is probalby not covered under GPL. You make some good points but I think that the solution is not to commercialize everything. There is starting to be a trend of businesses (and governments) turning away from commercialization (ever so slowly but it is in that direction). Pick something that is missing and contribute that to the community. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best place to order Cisco ATA 186
Hello Al, Please let me know how many of them do you want to order, we are distributors of Cisco and can help you . Als owe are providing International/Domestic calls termination to more then 260 countries worldwide. Thanks, Alexander Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: Al [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 05, 2003 7:36 AM Subject: [Asterisk-Users] Best place to order Cisco ATA 186 I want to setup an Asterisk network using the Cisco ATA 186. What is the best place to order those devices? I'm not finding them anywhere. Al __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk FAQ
James, You can make mirror of your site at our facilities. To support Asterisk community we can host mirror of your site, or make it primary hosting whatever is more convinient for you. We can do it duting this weekend. Let me know Alexander Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: James H. Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 01, 2003 6:25 AM Subject: Re: [Asterisk-Users] asterisk FAQ I'll see what I can do to upgrade the speed of www.voip-info.org Traffic has been going up as it gets more popular. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Michael Wood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 31, 2003 6:44 AM Subject: Re: [Asterisk-Users] asterisk FAQ It has been extremely slow for me too. Regards, Mike On Fri, 31 Oct 2003 15:46:49 +0100, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote : I just went there. Do they share a single isdn B channel with 50 other servers? it was sloow. I'll put it there, eventually On Fri, 2003-10-31 at 15:21, Rich Adamson wrote: Roy, I've started to write an FAQ for asterisk, available here: http://asterisk.pronto.tv/faq.php Please help me fill it up with the good stuff :) Why don't you put it here: http://www.voip-info.org/tiki-index.php and folks can updated/edit online? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank with E1
Sergi, I would say it depends of your budget. You can find on market different channel banks. Some of them are very expensive and have all fancy features, some of them are not so expensive and of course are missing some features. We are using CAC and New Bridge chanell banks, they are working good and no problem. When you are looking for channel bank, make sure it supports Answer supervision, it is very important feature. But I don't know what exactly are you going to do within your netowrk for 100 phones.. Do they need all features to be trasmitted like Calle ID , from outside world ? Also you can take a look into Adtran or NewBridge. IF you have more specific questions about them, please let me know. Thanks, Alexander *** XVOIP network is lunched, get your +1 777 number today. [EMAIL PROTECTED] *** Unofficial Asterisk Forums *** URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register *** - Original Message - From: Sergio Serrano Revuelto [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 8:44 AM Subject: [Asterisk-Users] Channel Bank with E1 I need connect up to 100 analog phone to a H.323 network through *. I think use TE410P, But I need to know what channel bank is better. I use E1 lines Any idea? Thanks in advance, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de DUSTIN WILDES Enviado el: miércoles, 29 de octubre de 2003 14:30 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Answering Machine Detection Thanks for all the info! So I take it I would need to either build an additional APP to asterisk like (voice_detection) or into an AGI and have that application or AGI run after the call is Answered? Fortunately it's not a telemarketing system! :-) It's an appointment reminder system for some of our employees. Calls them up and reminds them of important tasks like meetings and stuff. -Original Message- From: Michiel Betel [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 8:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Answering Machine Detection See http://resource.intel.com/telecom/support/documentation/unix/SR50_linux/ html _files/vox_feat/contents.html#TopOfPage chapter 2 for a basic insight on Dialogic does it... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: woensdag 29 oktober 2003 3:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Answering Machine Detection Humans tend to say Hello? (short burst of audio followed by silence), and answering machines tend to say I'm sorry I'm not here right now, please leave a message after the beep (long burst of audio followed by a beep and silence). So, basically you need to decide 1) what is audio and what is background noise and 2) how long should there be audio followed by silence. On Tue, 2003-10-28 at 19:25, Alastair Maw wrote: On 27/10/03 21:57, DUSTIN WILDES wrote: Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? There's obviously no nice way of doing this. If you're doing telemarketing, and you're playing pre-recorded audio, which of course is a nasty thing to do, the algorithm is something like: 1. Dial out. 2. Wait for answer. 3. Start playing audio. 4. If you hear something that sounds like a beep, either hang up and try again later, or stop the audio, pause for two seconds and start playing it again. 5. Hang up when finished playing audio. Step 4 is accomplished by doing a FFT on the incoming audio into frequency buckets and taking a rolling average of the mean and standard deviation, such that you can detect when a fixed monotone beep occurs at the other end. If you don't want to play audio files and wait for beeps, and want to connect real humans to each other, then there's no decent way to do this, as the only difference between humans and arbitrary answering machines is that the answering machines give you a beep prompt to record your message. Regards, -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk
Re: [Asterisk-Users] Answering Machine Detection
some information can be found here about algorithm and descriptions of method being used. http://citeseer.nj.nec.com/393112.html Regards, Alexander *** XVOIP network is lunched, get your +1 777 number today. [EMAIL PROTECTED] *** Unofficial Asterisk Forums *** URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register *** - Original Message - From: Ray Burkholder [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 9:17 AM Subject: RE: [Asterisk-Users] Answering Machine Detection Might want to write a new energy detector algorithm in dsp.c though based on a wideband/low Q resonator approach (move the pole way in towards the origin) as opposed to narrow band goertzels (pole on the unit circle). More robust for this type of work. Where does one go to learn this terminology and the math to implement it? -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card
We are going to make pilot project for one of our customer, based on QuadT1 card. I will let you know after we finish test. Regards, Alexander Unofficial Asterisk Forums *** URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register *** XVOIP network is lunched, get your +1 777 number today. [EMAIL PROTECTED] *** - Original Message - From: Russ Beaupre, P.E. [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 9:38 AM Subject: Re: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card FW: Voice/Data mixed routing over Digium E1/T1 CardWe are using it in three sites where the T-1 is pure IP and calles are routed in/out over SIP IAX2 and then to a channel bank. As a router the T400/T100 works great; I would highly recommend it. As voice server it works great. As a combined router/voice server (using only IP voice trunks) we've been having some issues (which are being worked on...Thanks Mark) Russ Beaupre, P.E. BoTech Communications Corp. - Original Message - From: Ray Burkholder To: [EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 9:01 AM Subject: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card The documentation mentions that the Digium channels can be split into some voice channels and the remainder of the channels used for routing IP traffic. Does any one have this in use in conjunction with Asterisk? Does it work well? Would you recommend it for a production server? Obviously, if this works, this makes for a cost effective platform where you obtain one E1/T1 to a provider, and they can provide TDM and data over the one circuit. No separate router required. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses dangerous content at One Unified and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 16xE1 solution based on *
We are looking to utilise Digiums QuadxE1 cards for one of our European customer, he is looking for 16xE1. Carrier has no VOIP support currently, so we need to organize by ourslef VOIP-TDM interface, to get them connected to our network. Idea is to try Digium/Asterisk solution. It will be some kind of VOIP gateway application, call from PSTN will come to Asterisk via Quad E1 and will be forwarded to our centralized Nextone switch to handle digits manipulation , routing and termination. I am curious how big server should be to handle 4 QuadxE1 of traffic. Definately it is not good idea to put all cards into one box and to have one point of failure. So I was thinking about 2 QuadE1 cards in each box. Does anyone has experience with real time traffic on such volumes ? What is your experience? What kind of hardware do you run ? Also how long usually takes to receive Quad port card from Digium, do they have usualy csuch cards in stock ? Regards, Alexander Unofficial Asterisk Forums***URL : http://asterisk.xvoip.comRegistration is : http://asterisk.xvoip.com/profile.php?mode=registerXVOIP network is lunched, get your +1 777 number today. [EMAIL PROTECTED]
Re: [Asterisk-Users] 16xE1 solution based on *
It happens after someone is just hungry and thinking about food ! ;-) Heh ...typoe happens.. Let's don't make from it story ... - Original Message - From: Linus Surguy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 10:24 AM Subject: Re: [Asterisk-Users] 16xE1 solution based on * XVOIP network is lunched, get your +1 777 number today. [EMAIL PROTECTED] XVOIP is lunched? Is this what happens after one is breakfasted? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SS7 signaling/Softswitch
OpenSS7 project mentions Asterisk also. I think project will bring something what we all really need - SS7 support for Asterisk Take a look : www.openss7.org Regards, Alexander Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register - Original Message - From: Ray Burkholder [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 27, 2003 8:01 PM Subject: RE: [Asterisk-Users] SS7 signaling/Softswitch You are correct in regards to what SS7 is and does, I thought it would be helpful to bring other users on the list up to speed. ;-) Some additional SS7/VoIP integration info from a 3Com perspective can be found at: http://www.c7.com/ss7/whitepapers/3com_ss7_intelli.pdf What I was inquiring about was Gus' comment about a PRI treated as route on a 5E. I'll have to defer to Gus for an answer on that one. I'm also trying to find out what types of SS7/AIN features may be available over a PRI D channel. For instance, message waiting indication (MWI) signals are sent interoffice over SS7. Could one formulate a packet that's sent over a PRI D channel that would end up in a remote switch via SS7? The MWI you mention is probably part of CLASS services, and is probably a function of AIN on an SS7 SCP (Service Control Point), to which a Telco's switch is connected. For some light reading on AIN, SCP, TPAN and related bits, this page has some interesting info: http://www.ulticom.com/html/products/ss7/ain.asp It doesn't directly answer your question, but I would guess that the Class 5 switch has to make some sort of translation between what happens in the D channel on a PRI and what it needs to communicate over its backend SS7 network. I see two proper solutions: a) implement SS7 directly so you have access to the signaling network for your application, or b) just handle the communications over the ip network in a converged network scenario. By the way, why do you ask the question of the D channel message? What is your application? So, the proper answer is that if you really want to implement this PRI - SS7 - PRI message, you should really be talking to your nearest CO Engineer or Telco Enterprise Business Office where they handle this all the time for enterprise call center applications. On the other hand, maybe Gus could contribute a regular tutorial on how he's got various things interconnected. The more the info, the better. Gus once asked if we want the plethora of info he can provide. I vote yes. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
