Re: [asterisk-users] Morse Code
On Thu, Feb 25, 2010 at 8:00 PM, David Gibbons d...@videon-central.com wrote: Duh! How are we going to spread the word about how to take those alien bastards down if we don't keep morse code around!?!??! And what about if you're trapped in ship that sinks? What if the 3g coverage isn't good? Or you have no more battery? /r -- _ -- 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] [OT] Asterisk 1.6 and DECT Phones
On Tue, Feb 23, 2010 at 9:23 AM, Alan Lord (News) alansli...@gmail.com wrote: Another vote for the Siemens Gigaset range. Been using the S685IP almost since the day it was released here in the UK. Nice handsets, great voice quality, but as others have said the UI can be a bit slow. Alan, don't forget the link to the discussion on your excellent site: http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ Same experience here, we've been running our 2-person soho for a couple of years with one base and 2 S675IP handsets. - They look and act like a regular cordless phone to the average person who is not a telephony geek. - They work well with a bunch of SIP accounts and g729 if you have that possibility - common headset jack works with cheap headsets - landline connection that works transparently when the Internet connection is down - simple dialplan to route calls - Excellent battery life and talk time Ours have performed flawlessly. Yes, the interface is slow and so is the phone menu system. We just purchased another base and handset for our new office. I love this phone and wish I was getting a commission on the number of units I've probably sold. /r -- _ -- 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] [OT] Asterisk 1.6 and DECT Phones
On Tue, Feb 23, 2010 at 10:50 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: High quality to me means well built, reliable, good protocol support and above all a responsive manufacturer. Incidentally, I've dropped two of the S675IP handsets on the hardwood floor a few times, still working fine. Concrete may be a different matter. /r -- _ -- 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] Denying call transfer to certain extensions
On Tue, Feb 23, 2010 at 7:50 PM, Danny Nicholas da...@debsinc.com wrote: What I want is, if a call coming from a trunk 100 rings, and if the caller wants to be transfered to 101, the transfer is denied. In other words, 101 can't get transfered calls. WHat about using featuresmap to replace the usual transfer application with code that tests to see the origin of the cal ind if it is from the 100 do something else, otherwise transfer as expected. /r -- _ -- 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] sip.conf - sort order, does it matter
On Fri, Feb 19, 2010 at 8:54 AM, Olle E. Johansson o...@edvina.net wrote: You propably have a type=friend where the user part matches before you even hit the peer part, where the insecure configuration parameter matches. There is a confusion here on the From: username and the authentication username used, so there is a challenge sent. Is it just me, or would it be nice if a clear, understandable and unambiguous way to express codec desirata was invented? Is there a future iteration of SIP that deals with it? /r -- _ -- 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] Feb 19th @12 noon EST: Voxeo's Tropo
Hi, When Jason Goecke talks, VoIP ideas become reality, and this makes my day. On this call we’ll talk about the newest features in Tropo and how to get started with telephony apps in the cloud without adding new infrastructure. Here's a chance to speak directly to Jason (or JSON as we now call him) to ask your questions about VoIP cloud apps. Tropo is a cloud communications platform that to add voice, instant messaging (IM), and SMS to your applications, using the programming languages and tools you already know via web services API and JSON. You can get the date/time in your local time zone here: http://vuc.me/next For all other info, including SIP, Skype, PSTN and Ouija board dial access numbers: http://VoipUsersConference.org or http://VUC.me IRC: Join us on Freenode.net on #vuc channel Hope to hear you there! /r -- _ -- 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] Important security alert: update your dialplans now!
On Mon, Feb 15, 2010 at 9:51 AM, Olle E. Johansson o...@edvina.net wrote: To avoid extensive rewriting and fix the current issue. That works in countries where you have fixed-length numbers. Unfortunately, not every dialplan works that way, so that can't be a generic advice even though it may solve your problems. Thanks for your suggestion! Olle, this may be a stupid question, but shouldn't a native santitize function be urgently added to the code base in all versions or change the dialplan compîler to ignore dangerous characters? /r -- _ -- 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] VUC Friday Feb 12th: HD Communications Summit
Hi, Barring WW (wifi woes), I will be broadcasting live from the HD Communications Summit this Friday. Usually we begin at 12 Noon EST but we may start earlier so please check the site, IRC, Twitter or Facebook for the exact start time. If any of you are planning to be there, please email me if you'd care to have coffee or something stronger. Also, if you happen to be in the vicinity of Paris, we have an invitation left. Site: http://voipusersconference.org IRC Freenode.net #vuc Twitter: http://twitter.com/voipusers Facebook: http://facebook.com/voipusers -- _ -- 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] Losing local SIP phones when internet goes down?
On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson o...@edvina.net wrote: What I have seen on my asterisk box when I had a up/down adsl line was that the asterisk box couldn't do dns resolution and would hang( well no other internal calls could be made, seemed like some sort of semaphore was stuck) when the adsl came up and dns could be done, everything worked fine again Confirmed and experienced years ago in a release far, far away. Yes, that is the case. Asterisk doesn't have asynchronus DNS support, so in order to work when the link is down, you need a local resolver, like a caching BIND server, on the same host. The calls to DNS resolvers in Asterisk is synchronus, so Asterisk will wait for the response to arrive. IIRC, at the time I had this problem, asterisk did not answer analog phone lines either so as a company we had no phones and had to revert to regular telephones plugged into the wall. Even if Internet is working, if the configured DNS is down, you're still sunk. This sorely needs to be fixed IMO. /r -- _ -- 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] Losing local SIP phones when internet goes down?
Why not run a internal DNS with forwarders to your ISP ? That way Asterisk can still resolve itself and hosts internally. See above: you need a local resolver, like a caching BIND server, on the same host. Nice, but still, it ruins the all in one concept. Isn't there a lighter solution? /r -- _ -- 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] Losing local SIP phones when internet goes down?
On Fri, Feb 5, 2010 at 10:39 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: Doh! :) My philosophy has always been to install a local named server, whether it be for Asterisk or something else, as most of the time everything I do is behind NAT and I prefer to resolve internal addresses. This also help if you run your own mailserver and make extensive queries to RBLs etc. That last bit makes a good point. And speaking of RBL, is anyone doing a SPIT RBL? I was plagued by comment spam on an old forum I wrote in C years ago and I finally wrote a function to check projecthoneypot.org's httpBL. I feel like my days just gained an hour, the one I wasted every day modertaing useless spam comments. Are there lists to check for know pests in VoIP? /r -- _ -- 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] Losing local SIP phones when internet goes down?
2010/2/5 Vinícius Fontes vinic...@canall.com.br: Have you tried to set srvlookup=no on your sip.conf? I think that just stops SRV lookups, not regular DNS. /r -- _ -- 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] OT: VUC Feb 5th @ 12 Noon Open VPN
Hi all, OT but possibly of interest to many of you in the asterisk community, Markus Feilner is our guest tomorrow on the VUC: VPN Users Conference. Markus is an interesting guy. In a former life, Markus ran an asterisk box and used Sipgate.de. He works for a German Linux publication and just wrote a book about Open VPN 2.0.9 called Beginning Open VPN 2.0.9. Join us: http://vuc.me Skype:vuc.me sip:200...@login.zipdx.com IRC: #vuc on Freenode.net or http://vuc.me/irc best /r -- _ -- 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] VUC Today at 1 PM EST: Counterpath/Bria
Hi, In the aftermath of Digium's and Counterpath's Bria for Asterisk announcement, we're happy to chat with Todd Carothers, Counterpath Product Manager today at 1 PM EST. For more info, http://vuc.me Join us on IRC #vuc on Freenode.net or use the web client at http://vuc.me/irc Call in starting at around 12 Noon EST: sip:200...@login.zipdx.com Hear you there! /r -- _ -- 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] Anyone going to HD Communications Summit - Europe Feb 12th?
I realize that many of you are too far away to consider it, but I know of a couple of people who are considering going. Is anyone tempted? I am planning on going and have a promo code for you if you'd like one. r -- _ -- 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] Snom vs Polycom
The problem 'I can place calls but no one can reach me' is our number one support question. Advising the user to check the DND As a general comment, the DND button on a decent phone should LIGHT UP when it's in use. On the Polycom 650, it is very clear on the LCD screen with flashing icons, but it would be much better to have the button lit when in use, and perhaps add a broken dial tone as well. On the opther hand, the button is not under the transfer button. /r -- _ -- 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] GoToIfTime issue
On Fri, Jan 22, 2010 at 9:51 AM, Zhang Shukun bit...@gmail.com wrote: exten = 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1) but what should i do. if i want to set seperate weekdays,like mon,wed. not continuous weekday like mon-fri. I couldn't find any reference to multiple, non-contiguous days on a quick Google, but this would work at the cost of an extra line: exten = 222,1,GoToIfTime(11:00-14:00|mon|*|*?1:3,1) exten = 222,2,GoToIfTime(11:00-14:00|wed|*|*?1:3,1) -- _ -- 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] Snom vs Polycom
On Fri, Jan 22, 2010 at 1:26 PM, Julian Lyndon-Smith aster...@dotr.com wrote: Anyone got any subjective (!) views on the merits of these two ranges , using asterisk 1.4 ? The choice of phones is crucial. Setting aside my tastes, you really need to get a couple of typical users to try them before committing to buying a bunch IMO. I'll bet someone like e4strategies.com would work something out if you called and talked to them. Even the one-liners need to be ergonomic and so many SIP phones are horrible at that. /r -- _ -- 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] Snom vs Polycom
http://twitpic.com/z8n36 On Fri, Jan 22, 2010 at 8:11 PM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hey hey! Anyone got any subjective (!) views on the merits of these two ranges , using asterisk 1.4 ? I need to supply approx 30 handsets to a new client, with the senior managers (6) having some slightly more managerial phones -- _ -- 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] AstLinux 0.7.0 Released
On Wed, Jan 20, 2010 at 4:40 PM, Darrick Hartman dhart...@djhsolutions.com wrote: The AstLinux Team would like to announce that the 0.7.0 version of AstLinux is available for download. There have been many significant updates in this release including updating to the latest Asterisk Release (1.4.29), moving to DAHDI (2.2.0.2) along with several other system updates. To chat with the Astlinux people, join us this Friday at 12 Noon EST on the VUC: Web info: http://vuc.me IRC #vuc on Freenode.net The open mic mentioned here several weeks ago is also happening later on in the same call. Join us! /r -- _ -- 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] Friday Jan 15 @12 Noon EST: Hacking VoIP
Hi, Our guest this Friday is Himanshu Dwivendi, author of the book Hacking VoIP. You're welcome to come discuss it with us on the conference. Find your local time by going to http://vuc.me/next - the conference begins a little before 12 Noon Eastern Time. VUC has an IRC channel #vuc on Freenode.net and uses Skype for Asterisk to allow connections to listen and/or speak. SIP g722 : 200...@login.zipdx.com For more info : http://VoipUsersConference.org /r -- _ -- 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] Please remove me from the mailing list.
About what? On Fri, Jan 8, 2010 at 3:39 PM, Steve Howes steve-li...@geekinter.net wrote: On 8 Jan 2010, at 13:52, John Novack wrote: Steve Howes wrote: On 8 Jan 2010, at 02:28, John Novack wrote: Careful, or Steve will un top post YOU! I like it in the past. Leave me alone ;) Different Steve!! I agree with him though :P ___ -- 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] [VUC] Today at 12 Noon EST (6PM CEST, 9AM PST) iNum with Voxbone
Hello, In about one hour we should be chatting with Tim Behrins of Voxbone about their initiative, iNum. I say should because he's the scheduled guest, but I haven't heard from him today :) Next week, we'll be Hacking VoIP Feel free to top post your answers, it seems to stimulate conversation. /r http://VoipUsersConference.org for the usual data or jump on IRC #vuc on Freenode.net ___ -- 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] Please remove me from the mailing list.
On Fri, Jan 8, 2010 at 5:25 PM, David Gibbons d...@videon-central.com wrote: I would have read your message but I couldn't find it amongst all of this garbage... Funny I saw your right away :) Ok, all kidding aside, I really don't care where people post if only they'd clip all the garbage out, the footers, the greetings, etc and just left the points to answer and their answers. But heck, I'm being serious which is against all the RFC. /r ___ -- 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] How to dial a number make two phone Ring at the same time?
On Thu, Jan 7, 2010 at 2:38 PM, Zhang Shukun bit...@gmail.com wrote:
hi,
i want to dial a number to let two phone ring at the same time or
alternative ring,
how should i configure in asterisk? or how to right the Dialplan code?
exten = 12345,1,Dial(${PHONE1}${PHONE2})
each phone variable is defined as stated in docs depending on the
protocol, SIP, IAX2, etc
as in
exten = s,1,Dial(SIP/2000)
So PHONE1 would be SIP/2000
See
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.con
/r
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Re: [asterisk-users] How to dial a number make two phone Ring at the same time?
On Thu, Jan 7, 2010 at 3:07 PM, Zhang Shukun bit...@gmail.com wrote: Thank you! but how can i determine whether ring at the same time or alternative ring? BTW, the uri http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.con It got mistyped or cut, it's http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf The concatenation I showed was for simultaneous ringing of devices. For the rest, yuou will be best served by looking through the docsz on dialplan and possibly queues. Best, /r ___ -- 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] Really Silly Question From Total Newbie
On Wed, Jan 6, 2010 at 8:50 AM, Allann Jones allan...@gmail.com wrote: But jailbreaking increases the freedom to develop a application and Oh, I agree with you, but it's probably even better to make a decision to either buy into the constraints of Apple or find a better, free-er phone, which is what I hope a lot of people will be doing in the next few years. In a vain attempt to return to VoIP and Asterisk, let's hope that more future mobile OS will all allow multi-apps so that you can leave a SIP client running in the background. The big problem with jailbreaking is updates. If you have a lot of time and energy to manage that problem when it comes up, jailbreaking is fun in a geeky sort of way. /r ___ -- 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] Really Silly Question From Total Newbie
On Tue, Jan 5, 2010 at 8:53 PM, UIT DEVELOPMENT uit...@gmail.com wrote: I've been poking around the past few weeks, trying to familiarize myself with all of this. I am new to Linux, VoIP and Asterisk -- to be complete. This is my first exposure to all of these technologies. I think one of the best things to do is to read this book: http://oreilly.com/catalog/9780596009625 It will allow you to ask specific questions about stuff you may not get but in the meantime it will tell you all the basic things about what asterisk can do in terms a newbie can easily assimilate. There are also a lot of web sites out there with tutorials about the world of VoIP and Asterisk, and of course the IRC channel #asterisk on Freenode.net /r ___ -- 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] Really Silly Question From Total Newbie
On Wed, Jan 6, 2010 at 6:48 AM, Allann Jones allan...@gmail.com wrote: Jailbreak your iPhone and install Cydia to have a Unix like open source environment (based on Debian), then install Siphon SIP client, and have fun! There are at least 4 iPhone SIP clients available for $3-10 that work well and do not require jailbreaking the phone. http://www.voipusersconference.org/2009/sip-for-apple-iphone/ http://www.open-voip.com/blogs/blog1/2009/09/27/voip-on-the-iphone-and-ipod-touch-a-comp By the way, this thread make me realize that-two year contracts should have an unemployment clause that would allow the signee to trade the subsidized phone for a basic one and reduce to normal, inexpensive cell service. There should be a legitimate out for provable force majeur other than bankruptcy. The lack of this is just one of the reasons I have never signed a contract and stick to prepaid. But I digress... as usual! /r ___ -- 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] Friday Jan 1 Voip Users Conference
Thanks to Digium, the company, and to all of the fine people from Digium who participate in the weekly VoIP Users Conference conference! We will be live on Friday January 1, 2010 and there is also a reel of recorded greetings from people around the world wishing the VoIP Community a Happy New Year. You can hear this anytime during the year by downloading it from the site starting next week. Besides hangover remedies, live participants will be talking about the decade in VoIP and maybe what's to come. January's schedule has Tim Behrsin from Voxbone on iNum on January 8th, Hacking VoIP author Himanshu Dwivedi on January 15th and a guest from Plantronics on January 29th. Sometime in the coming weeks, Markus Feilner, author of “Beginning OpenVPN 2.0.9“. will be with us. When we have authors, their publishers usually give us a couple of books to give away as well. Until then, I wish all of you in this community the best of all possible combination of health, happiness, prosperity and minimal jitter. /r http://vuc.me Call (518) VUC-VOIP and say Happy New Year ___ -- 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] Live CD - do you think they are worth doing?
Hi, Curious, do many of you check out software or projects when they have a live CD or does that make any difference to you? Does anyone know if the general public (not reading this kind of list) is attracted to a Live CD more than an Install one? thx, /r ___ -- 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] Live CD - do you think they are worth doing?
On Sun, Dec 20, 2009 at 5:10 PM, jon pounder j...@inline.net wrote: Live usb sticks are another matter (assuming your bios actually reliably boots them) at least you can save your changes and pickup where you left off the next time. Excellent point, thanks! /r ___ -- 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] Live CD - do you think they are worth doing?
On Sun, Dec 20, 2009 at 6:09 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: However the writable storage also allowed us in our live CD to save some minimal configuration on the media. We have a CD version and a USB version of our live system, which are basically the same. The system configures a working Asterisk server with a web interface and a ssh server. Those two use some default user and password (which are hard-wired - after-all, some remote user has to use them, and we don't want to assume there's a local user at the keyboard). Would you say you've found it well worth making the Live CD available,then? (Sounds like you have.) By that I mean, there is a definite audience for it? /r ___ -- 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] Live CD - do you think they are worth doing?
On Sun, Dec 20, 2009 at 7:18 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: I am not sure if this is even on-topic for the biz list If in doubt, why not skip it and move on? I am asking people who offer asterisk-related products and voip-related products as Live CD, such as Asterisk Live... I am interested in the answers of the community and I encourage anyone with input to continue the discussion here. /r ___ -- 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] Friday @12 Noon ET: Kamailio, Open SER and Asterisk
http://vuc.me Kamailio, Open SER and Asterisk walk into a bar... The bartender is Alex Balashov, someone whose posts I have long admired on this list. Alex has agreed to take us through the following areas: - Relationship of Kamailio to OpenSER project history. - What is Kamailio/OpenSER? - SIP proxy - SIP server (for certain purposes, such as registrar, presence user agent, etc.) - Common uses of Kamailio. - Service delivery platform engineering and Asterisk scaling using Kamailio. - Some discussion of sip-router.org initiative. Any questions you may have are welcome either live or by voicemail (see below) if you can't make it. On this occasion, Packt Publishing has given the VUC two electronic copies of the book Building Telephony Systems with OpenSER by Flavio E. Goncalves to give away during our first hour on the VUC. Join us today! To find your local time for the VUC: http://vuc.me/next Join the call: - via g722: sip:200...@login.zipdx.com - go proprietary: Skype:vuc.me and Skype:ld.vuc.me (g729) - use the Phone from Here widget: http://vuc.me/call - use Talkshoe: 7463#2262...@proxy.ideasip.com - PSTN (567) 252-2286 - IRC on Freenode.net: #vuc - Voicemail/SMS: (518) VUC-VOIP ___ -- 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] iphone client app
On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad a...@samad.com.au wrote: Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone http://www.voipusersconference.org/2009/sip-for-apple-iphone/ I have not done any Asterisk-specific testing, I hope someone who has will chime in. /r ___ -- 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] iphone client app
On Sun, Dec 13, 2009 at 11:24 AM, Randy R randulo2...@gmail.com wrote: On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad a...@samad.com.au wrote: Got a new iphone, want to know about peoples experience with any apps that work well with asterisk and run on a iphone http://www.voipusersconference.org/2009/sip-for-apple-iphone/ I forgot to mention Ruben's post on this, a review of the apps he has tried. http://www.open-voip.com/blogs/blog1/2009/09/27/voip-on-the-iphone-and-ipod-touch-a-comp At some point I want to take the time to record a call from each of the apps to the same server from the same device and mic (I use an iPod, not iPhone) /r ___ -- 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] iphone client app
On Sun, Dec 13, 2009 at 8:45 PM, meetmecall i...@meetmecall.nl wrote: Siax is working great for me and as far as I know/remember well, you can get it from the app store for a reasonable price. It supports SIP and IAX2 and works easy with Asterisk. It looks like it requires a jailbroken iPhone, am I wrong? /r ___ -- 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] VUC Dec 11 @ 12 Noon EST: g729 transcoding, software hardware
Hi, We had a last-minute cancellation from Vivox for today's conference. It happens that someone suggested a guest idea, Howler Technologies CTO Jay Fenton, who agreed to join the call from the road. Anything you want to know about transcoding to and from g729 is out topic for the first hour. My pal David Duffet knows this technology well and has kindly signed in to help guide us through this as well. Just before this time in your local time zone : http://vuc.me/next (12 Noon EST) why not join us on IRC: #vuc on Freenode.net anytime SIP:200...@login.zipdx.com Skype:vuc.me (or skypeld.vuc.me for reduced bandwidth) PSTN: (567) 252-2286 Java web widget: http://vuc.me/call If you have a shipping address in North America, you can vie for a free Polycom ip335 : http://bit.ly/8px6al Independent of your location in the world, Howler Technologies is offering some free licenses for their g729 transcoding technology. See you there! /r ___ -- 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] g722 question
Hello, I am working with several SIP projects that use g722, or are trying to do so, with pjsip library. According to pjsip team's interpretation of g722, it works with 14bits PCM for input/output, so pjsip basically 'converts' the audio sample from 16 bits to 14 when encoding and vice-versa. Some implementations don't do 16-14 bits conversion, so when pjmedia talks to one of those the over-driven audio problems appear. What we need to know is what's the most used implementation: 14-16 bits conversion or not. Any pointers to help clear this up? We'd really like to see more g722-capable SIP clients for our own conference on ZipDX. Regards, Randy http://vuc.me ___ -- 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] g722 question
On Mon, Dec 7, 2009 at 3:23 PM, Kevin P. Fleming kpflem...@digium.com wrote: As far as I am aware, for ITU-T compliance the codec only cares about 14 significant bits, but the reference source code needs those 14 bits in the *top* 14 bits of each 16-bit word that it supplies/produces. The Asterisk implementation does not do any bit-shifting or masking at all, and seems to interoperate with quite a few endpoints just fine, so presumably that means it's the correct implementation :-) Hi Kevin, Thanks for sharing that. What I found odd was that so many clients got distortion. It then became apparent that most were using pjsip. However, since then I spoke to someone who is NOT and they are getting distortion as well, presumably due to a similar problem. I've no experience with the Asterisk implementation, even in pass-though. I've heard that 1.6 can do it, I think. Best, Randy ___ -- 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] Today in 30 minutes: VoIP on Social Networks
VoIP Users Conference begins in about 30 minutes to discuss the use of VoIP on social networks like Facebook. If you have any interest in this (or maybe you customers do?) please join us IRC anytime: #vuc on Freenode SIP see http://vuc.me for all the URI and PSTN numbers Skype:vuc.me or skype:ld.vuc.me (for reduced bandwidth) See you there. /r ___ -- 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] Asterisk registers with private IP
externip=123.123.123.123 On Tue, Dec 1, 2009 at 4:32 PM, Joao Gomes Pereira gomespere...@startel.pt wrote: Hello I'm trying to register an Asterisk working behind Nat. Here is the trunk: register=username:passw...@sip.startel.pt [startel] type=peer host=sip.startel.pt username=username fromuser=username secret=password qualify=yes disallow=all allow=ulaw allow=alaw allow=gsm insecure=very port=5060 nat=yes canreinvite=yes The problem is: Asterisk is registering with its internal IP (192.168.1.25), as you can see here: sip:s...@192.168.1.25 Q= Expires:: 81 Callid:: 480b40aa13ddd8707787b21a69656...@127.0.0.1 Cseq:: 103 User-agent:: Asterisk PBX How can I force Asterisk to register with its public IP? Is it possible to configure STUN in an Asterisk trunk? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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 -- France: 01 70 61 22 21 USA: (415) 727-0927 ___ -- 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] 1800 DID Provider - Suggestion
On Fri, Nov 27, 2009 at 1:54 PM, Marco Cordeiro marco.corde...@globalstar.com.br wrote: Do you guys suggest any 1800 DID Provider in the US ? We like OnSip.com / Junction Networks stable and various service levels from none of hosted pbx. You should post this to the -biz list. /r ___ -- 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] keep asterisk in RAM
On Tue, Nov 24, 2009 at 2:36 PM, jefferson alexandre jefferson.alexan...@gmail.com wrote: On Tue, Nov 24, 2009 at 11:21 AM, Jerry Geis ge...@pagestation.com wrote: Is there a way to keep asterisk in RAM and tell linux not to swap it out (ever). On a closely related note, has anyone built a normal (not embedded) system on SSD? It might help if it works well with linux. It seems to make a huge difference with OS like OS X. /r ___ -- 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] keep asterisk in RAM
On Tue, Nov 24, 2009 at 3:42 PM, Richard Kenner ken...@gnat.com wrote: On a closely related note, has anyone built a normal (not embedded) system on SSD? I've been running Asterisk on a 20GB SSD drive for a while now. And? Noticed any significant performance advantage? /r ___ -- 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] little boy on asterisk and Debian
On Fri, Nov 13, 2009 at 11:31 AM, Manu et...@manu-dpk.net wrote: Can you help me please? Thank you very much. Voici un meilleur site pour poser des questions de tout genre en français : http://asterisk-france.net/ /r ___ -- 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] VUC to...@12 ET: Allison Smith
If you missed @voicegal last time or didn't go to Astricon, join us today on the Voip Users Conference to meet Allison Smith, the voice of Asterisk. Or go listen to the FBI talk about security... http://VoipUsersConference.org for details. /r ___ -- 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] [VUC] Friday Nov 6 @ 12 Noon EST: Village Telco
Hello from http://VUC.me or voipusersconference.org This week on the VoIP Users Conference we welcome the Village Telco Project [http://www.villagetelco.org/ ] self-described as an easy-to-use, scalable, standards-based, wireless, local, do-it-yourself, telephone company toolkit. This is an inspiring project with the stated ambition: ... to render local telephony in developing countries to be so cheap as to be virtually free. Thanks to advances in Open Source telephony software and the dramatic decrease in the cost of wireless broadband technology, we think this is entirely possible. They are developing open hardware as well: In a nutshell, the Village Telco needs an affordable device to connect customers to the meshed WiFi network. The Mesh Potato will dramatically reduce the cost of a Village Telco startup Please join us this week to learn more about this original and worthy project. We have a very wide range of ways you can be a part of our community, thanks to the efforts of members of the same community: Your local time of next conference: http://VUC.me/next Audio live communication channels: SIP g722 wideband 200...@login.zipdx.com SIP g711 7463#2262...@proxy.ideasip.com Skype Call vuc.me or if your bandwidth is limited, low def using skype:ld.vuc.me POTS +1 567 252 2286 Text channels: IRC #voip-users-conference Google Wave search for VUC or try http://VUC.me/nextwave ___ -- 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] Fwd: Asterisk conferences
Hi, If by chance you should find your self in Paris or wish to be there to present... this is for you. Note they do NOT want commerical presentations and this is only about Open Source Asterisk http://www.astrieurop.com/en/ I am considering going. Digium being a premier sponsor, I imagine some of them will be there? r ___ -- 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] Forward DID to another server
Alex, You forgot to clip the extra from the quote, shame on you! On Mon, Nov 2, 2009 at 9:47 AM, Alex Balashov abalas...@evaristesys.com wrote: Tzafrir Cohen wrote: Top-posting, on top of your other sins. Please spare us this capital punishment. An entirely fair point. Nevertheless, I eagerly await your similarly convicted petitions aimed at curbing illiterate, obnoxious and indolent attempts to get others to do extensive work on one's behalf to fix a problem one has not done the due diligence to rudimentarily understand. ___ -- 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] deploying asterisk
On Wed, Oct 28, 2009 at 5:05 PM, Darrick Hartman dhart...@djhsolutions.com wrote: Let's be realistic here. You need to 'drink the koolaid' before you install it for a client. What I'm saying is you really need to install Darrick, No, he already drank the koolaid by believing in asterisk. Now he needs to install it in order to eat his own dog food. Keep that straight! /r ___ -- 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] Tim Panton Astricon Presentation recreated
Tim Panton went to the pains of recreating his Astricon preso today in the form of a screencast: http://blip.tv/file/2762980 Amazing future of the Google Wave / Voice combo IMO. /r ___ -- 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] AstriCon videos: a question of method
As others have said, John, Viddler is good. If you have any shorter-than 10 minute videos, you might put them on YouTube as well for the sheer exposure and then add something pointing to a Viddler URL for additional, longer content. /r ___ -- 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] Astricon
On Wed, Oct 21, 2009 at 9:57 PM, SIP s...@arcdiv.com wrote: Sounds like it wasn't a very interesting track. ;) Not sure, but I guarantee the previous night was interesting :) The VUC guys, sometimes led by Randal Happy Hour Schwartz, know how to party. One night I got two hours sleep and was operating in virtual mode during the day. Ok, to be clear that was a feeble attempt at humor. I was the M.C. of the Carrier/Call Center Track. There were many very good presentations, but no video camera. Alistair has already posted his slides. There were several others whose slides (I assume) will be posted but which, as I say, were not recorded. More summing up on VUC this Friday. Perhaps we can get Tim Panton to tell us about his excellent presentation. Michael Graves will hopefully be there as well. /r ___ -- 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] Astricon
On Wed, Oct 21, 2009 at 8:28 PM, Danny Nicholas da...@debsinc.com wrote: Is THAT a summary :)? As I said above (or below?) I we'll be talking about this on VUC Friday at 12 Noon. In fact, here's the whole spa^H^H^H preview: VoIP Users Conference (VUC) Astricon, Been there, Got the T-shirts Several VUC members were at Astricon (and there is a VUC T-shirt). Two of them did presentations, Michael Graves and Tim Panton. We'll be talking about the keynotes, the sessions, the Code Zone and the parties. More Info: http://VUC.me IRC: #voip-users-conference Google Wave: open to VUC members who have Google Wave ID, please contact anyone in VUC to be added to the VUC wave. Note that we do NOT have any \/\/ave invites. Technical Notes: John Todd will someday have a g722 device, but for now he is still the standard of voice quality by which all others are measured. The general consensus is that a Polycom top of the line g722 phone under normal non-airport noise conditions will produce a quality rating of 2 Jtodds. Last week's VUC was recorded with a quality of 0.5 Jtodds. John Todd is currently at 1.0 Jtodds. /r ___ -- 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] Astricon
On Wed, Oct 21, 2009 at 4:01 PM, Bob Pierce pier...@westmancom.com wrote: Or charge for full access! Leave a few teasers, and charge some amount to see them all. I would pay - even close to attendance price... could only help you get past break even ;) I agree, I would be quite willing to pay for full access to all the videos from the Conference. I missed the first part of this, but has anyone said: not all the presentations were recorded. /r ___ -- 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] Astricon
On Wed, Oct 21, 2009 at 7:46 PM, Barry L. Kline blkl...@attglobal.net wrote: Randy R wrote: I missed the first part of this, but has anyone said: not all the presentations were recorded. Hi Randy. Yes, that was mentioned. Actually, three of the four tracks were videotaped IIRC. Barry And I was in the one that wasn't. So I guess I'll have to summarize... except I was a sleep one of the days :) /r ___ -- 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] OT - DECT SIP Phones
On Sat, Oct 17, 2009 at 7:57 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: | I have three Snom M3s at the moment but getting pretty fed up with | the | issues :( I am UK based and would be interested to hear of other | peoples The S685IP has no headset jack AFAIK. If you want to use a headset or you don't need Bluetooth, get a S675IP. They're great, they do g722 wideband and you have plenty of company in the asterisk world to give peer support. We have a bunch of Gigaset owners on our weekly conference and we've even gotten Siemens Gigaset division to make a significant firmware change bexause they're listening to what we users have to say. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Slightly OT: Astricon and Google Wave
Looking at my shiny new Google Wave account, I was wondering if anyone else on this list is in the beta AND going to Astricon. Astricon seems like it would be a good test of the kind of collaboration GW is trying for. In any case, I'd love to try to do an Astricon wave so let me know if you're interested and we'll get together. I know at least two other people who'll be there presenting. This might be an interesting way for them to get feedback. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Invitation] VoIP Users Conference @ Fri Mar 7 12:00 - 13:00 ()
BEGIN:VCALENDAR
PRODID:-//Google Inc//Google Calendar 70.9054//EN
VERSION:2.0
CALSCALE:GREGORIAN
METHOD:REQUEST
BEGIN:VEVENT
DTSTART:20080307T11Z
DTEND:20080307T12Z
DTSTAMP:20080306T082900Z
ORGANIZER;CN=Randy R:MAILTO:[EMAIL PROTECTED]
UID:[EMAIL PROTECTED]
ATTENDEE;CUTYPE=INDIVIDUAL;ROLE=REQ-PARTICIPANT;PARTSTAT=NEEDS-ACTION;RSVP=
TRUE;X-NUM-GUESTS=0:MAILTO:asterisk-users@lists.digium.com
ATTENDEE;CUTYPE=INDIVIDUAL;ROLE=REQ-PARTICIPANT;PARTSTAT=NEEDS-ACTION;RSVP=
TRUE;X-NUM-GUESTS=0:MAILTO:[EMAIL PROTECTED]
ATTENDEE;CUTYPE=INDIVIDUAL;ROLE=REQ-PARTICIPANT;PARTSTAT=NEEDS-ACTION;RSVP=
TRUE;X-NUM-GUESTS=0:MAILTO:[EMAIL PROTECTED]
ATTENDEE;CUTYPE=INDIVIDUAL;ROLE=REQ-PARTICIPANT;PARTSTAT=ACCEPTED;RSVP=TRUE
;CN=Randy R;X-NUM-GUESTS=0:MAILTO:[EMAIL PROTECTED]
CLASS:PRIVATE
CREATED:20080306T082859Z
DESCRIPTION:Every week we try to get guests with ideas\, products and servi
ces you haven't had time to check out to come and talk about what they're d
oing. \n\nTomorrow\, Pika Technologies will be with us.\n\nFriday\, March 7
that 12:00 PM (Eastern US) 9AM PST\, 5PM GMT\n\n*** Call (724) 444-7444 o
r SIP:[EMAIL PROTECTED] ***\n\nAfter the call connects\, enter the conf
: 22622# and your_PIN# (or 1# if you have no PIN)\n\nIf ( (${You_are_Regist
ered}) (${PIN} == callerID) ) you will not need to enter an ID\;\n\nhtt
p://www.VoIPUsersConference.org for how to listen and join.\n\nAccording to
their site at http://www.pikatechnologies.com\, Pika offers reliable medi
a processing building blocks connect computer systems to TDM and IP network
s. Brand name companies design groundbreaking IVR\, call center\, custom PC
/IP PBX\, fax and logging solutions using PIKA Technologies' components.\n
\nhttp://food4wine.ning.com is the VUC Community Site (archive recordings\,
forum)\n\nIRC Freenode.Net #voip-users-conference is the channel to ask qu
estions if you can't call\n\nJoin us\, we look forward to hearing you. (Ech
o? I don hear no stinking)\n\n/r\nView your event at http://www.google.
com/calendar/event?action=VIEWeid=ZHM2bGkzM2VxaTZkZmdzaTJtZDNuMWk2dGcgYXN0
ZXJpc2stdXNlcnNAbGlzdHMuZGlnaXVtLmNvbQtok=MjMjc3BhbXN1Y2tzMjAwNUBnbWFpbC5j
b21kZmQ4ZWYzYjgyNDYzY2Y2ZmZmOTI3OWU2Y2RkNWZiMGViYjhiMWM4ctz=Europe%2FParis
hl=en.
LAST-MODIFIED:20080306T082859Z
LOCATION:http://voipusersconference.org
SEQUENCE:0
STATUS:CONFIRMED
SUMMARY:VoIP Users Conference
TRANSP:OPAQUE
END:VEVENT
END:VCALENDAR
invite.ics
Description: application/ics
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