Re: [Asterisk-Users] Introducing Firefly
Nice!! Have just tried it a bit, seems cool... Congrats!!! Will test it against my * box and will provide some feedback. Thanks! Sam\\\ - Original Message - From: Adam Hart To: [EMAIL PROTECTED] ; [EMAIL PROTECTED] Sent: Tuesday, January 27, 2004 7:11 PM Subject: [Asterisk-Users] Introducing Firefly After many months of development, I'm pleased to announced Firefly - an IAX soft phone and network. The firefly softphone - free, runs under windows, allows connection to multiple networks, skinable interface, connection to firefly network, IAX2 protocol, (SIP in next release), codecs supported - iLBC, G.711 ulaw/alaw, GSM. - contact lists, selectable ringtones. download from here - http://www.virbiage.com/firefly/ The firefly network - also free, runs on an enhanced version of IAX2 (simply uses IAX2 text messages for customised part), voicemail, text messaging, online presence, ability to indicate status (available, away, NA). I believe you can connect using a standard asterisk box but you'll miss out on the extended features. The network runs on iLBC so unforunately it won't work with most IAX2 clients (unless you get * to translate) Thousands of people have used it but it's still regarded in beta, as we are still in heavy development (so expect a few bugs). It doesn't use iaxcomm as we needed our own framework to support sip, including our own jitterbuffer. If you don't wish to connect to the firefly network, click cancel when it asks you. Coming soon featuresSIP - in alpha, few bugs outstandingmusic onhold - playing mp3s while the other party is onholdfast audio - will reduce the latency by 40-50msspeex - (if anyone wants it?) Feel free to contact me on or off the list to report bugs and suggestions. I'll post everytime we release a new version (probably every week), including fixed bugs and new features Our website is http://www.virbiage.com/
Re: [Asterisk-Users] New Windows IAX Client
Steven, This is really great!! Very professional and features comparable to what one find in major brands softphones. I wonder if it is possible/planned.. - Programmable Directory Number buttons (DNs linked to specific buttons)? - A special version capable to run from removable media (USB or FDD) with no need from setup installation? This is great for frequent travelers and for first contact on site demos. - video phone feature? Regards Sam\\\ - Original Message - From: Steven Sokol [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 6:01 PM Subject: [Asterisk-Users] New Windows IAX Client Announcing a new Windows-based IAX/IAX2 client. Please download it and give it a try. Let me know about any bugs, and any missing features. I have yet to come up with a catchy name for it, so at this point it calls itself IAX Phone. (Suggestions? Non-derogatory suggestions, preferably). Download: http://www.sokol-associates.com/Downloads/IaxPhone.zip Reference Support Page: http://www.sokol-associates.com/IaxPhone.htm Features: - Works correctly for both inbound and outbound calls! - Registers With Multiple Servers - 4 Line Appearances - Direct IAX URL Dialing (user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) - 20 Speed Dials - Native IAX Blind Transfer (no more wasted pound key) - Drag/Drop Transfer (right-click drag a line to a Speed Dial to transfer) - Last number redial - Message Waiting Indicator (Native IAX, no Manager configuration reqired) - Message Waiting Count * - Registration Indicator - Missed Calls Indicator - Mini-Mode for low screen-real-estate usage. - Multiple Audio Output Devices (use speakers for Ring, headset for audio I/O) - Supports IAX/IAX2 - Supports DSP Filters: AGC, Echo Cancellation, Denoise - Integrated call timer - Integrated with the Eutectics IPP200 USB handset - Direct dialing for IAXTel users (no need for [EMAIL PROTECTED]) - Intercom Calls (auto-answer, use speakers for audio output)** Coming Soon: - Call Log - Phonebook Dialing - Outlook Integration (dialing screen pops) - Programmable feature buttons (DND, Forward, Direct-To-Voicemail Transfer, etc.) - Documentation (sorry...;-) - Additional Screen formats (toolbar, icon-tray) - Keyboard Shortcuts - User-defined ring-tones (wav, midi, etc.) - Personalized ring-tones (assigned in phonebook) - Enhanced speed dial dashboard as DSS (Manager integration) for Operator console - Call rejection/Zapateller - Optional IAX debugging and logging - Auto-answer options (either all calls or by phone-book entry) - Integrated call recording Hopeful Additions: - Simple text messaging - Local conferencing - Screen-saver awareness (auto Away when screen saver kicks on) This is the very first release so there will be bugs (as always, use at your own risk). Please let me know what they are ASAP. I will hopefully have the next release (with many of the coming soon features) out later in the week or over the weekend. Thanks, Steve Sokol * Requires a small change to chan_iax2.c in order to return valid message counts. See the support page. ** Requires a special macro in extensions.conf that replaces the caller's real number with a special Intercom indicator ID. See the support page. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2 or EM for E1 CAS pbx to pbx link
Hi, So, u are using a new thread...it is better if you do notchange the message subject so we can trace u properly. Before dealing with R2 signaling which is register_signaling, you need to deal with CAS or ABCD signaling which is line_signaling and be able to establish (and drop) calls correctly. Once this is done, R2 signalingcan take place ---iffully available in*,and R2 tones will be able totravel between switches thru theestablished voice channel before callers be allowed to talk. So, again... you need de CAS table of the far_end switch or pbx. Get it and attach it to your post. Also, if u thing u will not able to negotiate with your far en party, ask him/her in advance what signaling and start arrangement should you use. Should tell you something like 4wire EM, IMMediate. This info is also crucial. Regards Sam\\\ - Original Message - From: M.A. Ali To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 5:01 AM Subject: [Asterisk-Users] R2 or EM for E1 CAS pbx to pbx link hi, thanx for the response. I just tried to work on R2 CAS but i found that the libr2 has not been implemented well and tested. I think in addition to EM, R2 can also be used in a pbx to pbx E1 link. what do tou suggest Sam ?? About the R2 implementation for asterisk i have seen in the list that steve has implemented 95% of that...but we dont see any release of that. any current info on R2 development?? and Sam you are right i don't have the CAS table of the other switch. But i think i can get one. help me out in this. I have to make a E1 pbx to pbx connection using CAS. thanks in advance janjua Help STOP spam with the new MSN 8 and get 2 months FREE* ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: Word-of-the-Day: wiki
hope this help some one else --as it helped me, in undertanding contributions to * wiki pages... Regards! Samuel - Original Message - From: whatis.com [EMAIL PROTECTED] To: whatis.com [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 10:31 AM Subject: Word-of-the-Day: wiki THE WHATIS.COM WORD-OF-THE-DAY January 6, 2004 wiki __ SPONSORED BY: Smart CRM Today, it's about smarter CRM. And that's exactly why SearchCRM.com and Peppers Rogers Group have joined forces to bring you the industry's definitive CRM conference that delivers on both the promise and challenges of enterprise CRM. Attend the Smart CRM conference in Atlanta, GA, February 11 - 13, and go home with tested CRM solutions and critical tips and techniques from the industry's leading independent experts, analysts, and the top principals from the Peppers and Rogers Group. Confirm your registration today and save $1,495. http://searchCRM.com/r/0,,19798,00.html?track=NL-34 __ TODAY'S WORD: wiki See our complete definition with hyperlinks at http://searchwebservices.techtarget.com/sDefinition/0,,sid26_gci943070,00.html?track=NL-34 A wiki (sometimes spelled Wiki) is a server program that allows users to collaborate in forming the content of a Web site. With a wiki, any user can edit the site content, including other users' contributions, using a regular Web browser. Basically, a wiki Web site operates on a principle of collaborative trust. The term comes from the word wikiwiki, which means fast in the Hawaiian language. A wiki allows a visitor to the wikified Web site to edit the content of the site from their own computer. Visitors can also create new content and change the organization of existing content. The simplest wiki programs allow editing of text and hyperlinks only. More advanced wikis make it possible to add or change images, tables, and certain interactive components such as games. A wiki provides a simplified interface. At any time, contributors can conveniently view the Web page as it looks to other subscribers, before and after the changes they have made. It is not necessary to know HTML (hypertext markup language) or perform work in HTML code. The best known example of a wiki Web site is Wikipedia, an online dictionary building collaboration. __ SELECTED LINKS: The Wiki Wiki Web is a Web site about wiki that is itself a wiki Web site. http://c2.com/cgi/wiki?WikiWikiWeb The Wikipedia site is also available. http://en.wikipedia.org/ __ TODAY'S TECH NEWS: WILL DAVID OR GOLIATH MANAGE YOUR WEB SERVICES? Before evaluating product features, companies looking at vendors to help manage and secure their Web services need to ask themselves a fundamental question. http://searchwebservices.techtarget.com/originalContent/0,289142,sid26_gci943203,00.html?track=NL-34 2003'S HIGH-TECH WINNERS AND LOSERS Take one last look at the world of IT in 2003. http://searchsap.techtarget.com/originalContent/0,289142,sid21_gci942124,00.html?track=NL-34 ENTERPRISES CRAVE STABILITY FOR LINUX IN 2004 Administrators and experts have had enough with SCO and lawsuits. They want to see Linux take real steps toward mission-critical functionality in mainstream enterprises. http://searchenterpriselinux.techtarget.com/originalContent/0,289142,sid39_gci942123,00.html?track=NL-34 Catch up on all the latest IT news at http://searchtechtarget.techtarget.com?track=NL-34 __ SECRET WORD-OF-THE-DAY | What is IT? You probably flip these things on a regular basis. In a telecommunications network, it's a device that channels incoming data from any of multiple input ports to the specific output port that will take the data toward its intended destination. Do you think you know the Secret Word? Click this URL and see if you're right! http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci213079,00.html?track=NL-34 __ RECENT ADDITIONS AND UPDATES [1] January Horoscopes for Geeks http://searchcrm.techtarget.com/sDefinition/0,,sid11_gci919205,00.html?track=NL-34 [2] up-sell http://searchcrm.techtarget.com/sDefinition/0,,sid11_gci942967,00.html?track=NL-34 [3] business metric http://searchcrm.techtarget.com/sDefinition/0,,sid11_gci940481,00.html?track=NL-34 [4] stove-piped development http://searchcio.techtarget.com/sDefinition/0,,sid19_gci942115,00.html?track=NL-34 [5] customer acquisition cost http://searchcrm.techtarget.com/sDefinition/0,,sid11_gci942906,00.html?track=NL-34 ::: WHATIS.COM CONTACTS
Re: [Asterisk-Users] CAS SF Inband tone signalling problem
Channel Associated Signaling is a simple 4 bits 'protocol' often used in digital non-isdn E1 connections to define/indicate the state of the line (or channel): sieze, sieze ack, idle, etc, at a given moment. Usualy, only 2 of those 4 bits are used. However, given that CAS tables may vary from manufacturer to manufacturer, the most important thing to start with in a PBX-to-PBXCAS connection is to identify the CAS table being usedat the far-end switch. If you can not easily obtain the tables on the far-end --which is the usually the case whenthe brand of your switch is not the sameas the one at the far end, or if you don't have previous experience with CAS, you may get stuck even if you have a CAS analyzer. Given that in PBX-to-PBX scenarios you usually go with EM signaling, you will also have to deal with the start arrangement; usually WiNK or IMMediate You have to have the same start arrangemnet for each direction (REC/XMT) as the far end switch. If Wink Start, you also have to deal with the length of the wink pulse. If you have the far end CAS table, just program yours to match their values. I am new with Asterisk and have never used zaptel.conf, but if u attach the far end CAS table to your post, we can be more precise in helping. Rgds Sam\\\ - Original Message - From: M.A. Ali To: [EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 1:05 AM Subject: [Asterisk-Users] CAS SF Inband tone signalling problem Hi, I am having some problem in defining CAS SF Inband signalling on a Digium E100P card. The problem is that the syntax given in the sample zaptel.conf doesn't work. Can someone provide me with an example of the syntax. I am trying to connnect asterisk with another exchange using a E1 CAS. I guess SF is the only option as the rest fxo,fxs,em etc are all subscriber end signalling. i need a CO to CO (or PBX to PBX)signalling...Somebody guide me on that too. Any help will be highly appreciated. Thnaks Sincerly, Janjua The new MSN 8: smart spam protection and 2 months FREE* ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Class features in dialplan ?
If what u mean by CLASS is Class of Service, ie: the ability to allow/denny access to users to/from resources like public network based on the number they dial, this can by nicely achieved by using a powerful tool that * calls context. Playing with contexts you can define several different class of service levels that can be separately applied to every phone and will work independently of the type of technology of the phone (SIP, H323, IAX, Legacy, etc). Sam - Original Message - From: Lance Arbuckle [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 16, 2004 8:58 PM Subject: [Asterisk-Users] Class features in dialplan ? hey guys I thought I was making progress on my dialplan when I realized that the class features that are available for zap channels aren't available for SIP channels. I see references in the archives to adding pattern matches in the dialplan for CLASS features which has raised a couple questions. 1. Is implementing CLASS like features via the dialplan the currently recommended way to do this ? 2. In general, are there any problems using non numeric characters in a pattern match with SIP phones (i.e. _*67) ? So far I'm planning to do Call Forward Unconditional, Call Forward Busy, Call Forward No-Answer, and Do not disturb and maybe some speed dials but I haven't thought that one through yet. 3. Anyone willing to share some of their cool features that they've come up with ??? I'd be most appreciative :) Thanks. -Lance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie ISDN question
Hi All, Have just checked kapejod's quadBRI specs and looks wonderful. I am not an expert on ISDN either, but seems to me that features and functionally worth the 600 EUR (almost US$600,right??) suggested price. However, from * stand point it seems --pls pardon me in advance if I am wrong, that kapejod's quadBRI card provides much more 'horse power' than * really needs in standard applications In our case (my partners and I), we are looking for an internal plain multiBRI EuroISDN card capable of taking the B channels and deliver them to the * in a duplex mode, so that * take care of everything else: Caller ID, Call Routing (switching), Conferencing, Least Cost Routing, Protocol Conversions, etc, etc .Our ideal multiBRi card needs to operate in TE mode only but would be great if it was capable to accept either U or S/T BRI interfaces --with S/T required only in situations where a NT1 + 2POTS box should go before * to provide dial tone even during shutdowns. Given that our * servers -- P4 Dell pe400SC, costs about US $300 here, it would be great if this ideal multiBRI would cost no more than the server, which has (we guess) plenty power to run the BRI's card load and a 4x12, 6x18 or 8x24 small office application. Does any one uses or knows any BRI card, like the one of our dreams... (hope this is not a silly dream). I know that some of you are successfully using a single port BRI card. It is a kapejod's card too, right?? Thanks a lot! Sam p.s. probably this should go in a separate thread... - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 4:54 AM Subject: Re: [Asterisk-Users] newbie ISDN question Hi Thorsten, the E100P is a PRI ISDN Card (S2M in Germany). You cannot connect phones to that card. The quadBRI card has 4 BRI ports that can individually be configured for TE mode (to connect ISDN lines) or NT mode (to connect ISDN phones). Please find the details at: http://www.junghanns.net/asterisk/page17.html best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ hi everybody, sorry for posting such a stupid question ;) i've managed to run asterisk* with my AVM fritz2.0 card and a some VOIP-softphones (SIP, H323). the functions of asterisk* really satisfied me ;))) now i want to run asterisk* istead of our old PBX. but it would be great to connect some phones directly to my box. how does a E100P from digium work. can i connect it to my ISDN-line and my internal phones (ISDN)? it would look like this: [PHONE2] / [PC]-[E100P] - [PHONE1] \ [ISDN-LINE] thank you for your help!!! thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.0
Any link where to directly find the main differences between 0.5.0. and 0.7.0?? Thanks a lot!! Sam - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 10:09 AM Subject: Re: [Asterisk-Users] Asterisk 0.7.0 On Tuesday 13 January 2004 02:27, WipeOut wrote: Tilghman Lesher wrote: On Tuesday 13 January 2004 00:10, Mark Spencer wrote: Okay, it's 15 minutes late, but it's out, thanks very much to all the people who worked so hard this weekend to make this possible! There is one bug so far and it's critical. It breaks includes and the GotoIfTime application. I'll own up to writing the broken code. The fix is very simple, though (attached). Why not quickly patch the source an release 0.7.1 if the bug is critical? We're planning to do that, but there's going to be a lag between planning a release and getting a release out. For people who want to use 0.7.0 right away, it's better to release news of the discovery of the bug right away, not wait for a new release. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Which ISDM BRI Card for Asterisk?
Take a look at: http://ns1.jnetdns.de/jn/relaunch/asterisk/page15.html Hope this can help, too... Samuel On Fri, 2003-11-21 at 16:22, Cees de Groot wrote: WipeOut [EMAIL PROTECTED] said: I would recommend you dump i4l and use a CAPI card with the chan_capi driver.. The cheap solution is a AVM FritzPCI card(this is what I use).. The other solution is the either the Eicon or AVM active cards.. I have experienced lots of bus hangups with the Fritz!, where the card doesn't see anything happening on the ISDN bus anymore. For my production system, I've now ordered a AVM B1, which hopefully works better. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
