similar to: Asterisk Users Conference Friday @ 12:30PM EDT

Hi, Quick reminder that the conference is happening today at 12:30 PM EDT. I'd like to talk more about updating to 1.4. I now have a test box running asterisk 1.4.5, CentOS 5 and Lumenvox speech rec software. Seems to be fine except for some double NAT issues that could be router specific. Byran Johns from Shelton-Johns is our guest to share some of his extensive experience. More about him

Hi, I am going to be on the road for the next few days and with the variable delay on the mailing list, I am posting this now, 4 days before the conference. If you haven't yet listened or participated, please consider doing it. We have a great kernel of people at all levels of expertise and ideas and questions can be kicked around immediately (well, there's a few milliseconds lag). This

Hi, We're pleased have a 25-year telephony veteran with us tomorrow, Aswath Rao. Aswath maintains that "Trapezoidal VoIP is Evil". Join us and ask questions, make comments, argue about geeky details... and maybe win a Gigaset S675IP SIP/DECT g722-capable phone with an additional handset. Those of us who have these phones like them a lot. All dial in info is here: http://VUC.me -

Friday May 11, 2007 at 12:30PM EDT A short reminder that you can connect with others in the asterisk community by phone or SIP (or both obviously) during these conferences. Anyone interested in asterisk is welcome to join the conference. Details are found here: http://x2z.eu Past recordings are here: http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 If we're nice to the Digium guys, they

Hi, December 5th, 1999 was the initial release of Asterisk by Mark Spencer. We'll be celebrating this by gathering as usual at 12 Noon Eastern (9AM Pacific, 10 MST, 11 Central, 5PM UK and Western EU) for the VoIP Users Conference. You can get all the dial in information at http://VoipUsersConference.org including info on a SipAddHeader() kludge to avoid DTMF problems. IRC is Freenode.net

This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards,

AUC is Friday at 12:30 PM EDT. See http://x2z.eu Hi, One of our guests this week will be Jay Phillips to tell us about Adhearsion. Haven't heard about the open-source Adhearsion? Look here: http://www.linuxjournal.com/article/9519 Be with us to ask Jay questions. If you can't be there, download the recorded version (including last week's chat with Mark Spencer) here:

AUC is Friday at 12:30 PM EDT. See http://x2z.eu for how to join. http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 We'll be talking about SIP/IAX providers and I'd like to hear more about asterisk appliances like the Digium and the new D-Link. If the Digium guys are around as they usually are, you can also put any tech questions you may have directly to them, especially release

This Friday, part II of TDM solutions including ATA that do IAX and SIP without opening the box and installing a card. Your experience in this area would be appreciated. You can find us here: http://www.AsteriskUsersConference.org Also, a Google group has been created for discussions and scheduling of the conferences. If you feel like this is of interest, please join us:

The scheduled guest is JR Richardson of Nntegrated Solutions in Dallas. He wrote a widely-consulted white paper on the subject and I hope we can get background as well as answers to any questions we may have, so come on by: http://www.AsteriskUsersConference.org Here's a powerpoint of his presentation at Astricon:

For this week's conference, the two founders of Voicepulse, Ravi Sakaria and Ketan Patel, will be joining us. For those of you who are not aware, Voicepulse is an asterisk friendly VOIP provider that has won awards for service and innovation. We will also have Trixbox news, updates, as well as discount codes. Lastly, we are working feverishly to bring you more information regarding legal

I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work "as expected." I would like to be able to do something like: exten => _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal "*" followed by

Hello all, Note: This is still an asterisk-based discussion, but we widened the interests of the conference and also wish to avoid trademark abuse, although we continue work closely with Digium,hence the change of name. Today, we'll be listening to the latest on Astricon which is fast approaching. Our featured guest is Daniel Berninger from FWD. FreeWorldDialup.com has brought a lot of

As usual, we'll be jawing about any and all asterisk-related subjects with the usual gang and any new people are always welcome, regardless of your level of expertise. You can even come and ask questions, it's guaranteed to be a more pleasant experience than it will be on IRC ;) http://VoipUsersConference.org/topics.php IRC; Freenode.net #voip-users-conference

Hi all, We've had Yusuf Motiwala from TringMe on the VoIP Users Conference before when he annouced their Flash-based web phones. Now they've come up with something that tantalizes me, VoicePHP. Sure XML is a standard and fairly easy to implement, but not as easy as PHP, which I have used since version 2 (Yup, shadesof.phtml) Here's GigOm's take on it: http://tr.im/2y4g Check

Hello again, Mark Spencer will be joining us for questions on at least one of these conferences (we've discussed this and he is definitely onboard with the concept), but his schedule is such that we can't *promise* he'll be there this week. I haven't heard back from him about today, but I'll try to call him right before the conf to remind him. Mark's very busy, so thanks

Hi, this is to inform everybody that the translation of my new book (unstable version) is online at http://www.the-asterisk-book.com The book is a GNU FDL project. So everybody who wants to participate is welcome to do so. Also, everybody who needs material for his own work, feel free to take it as long as the new material will become GNU FDL too. I am glad that Stephen Bosch (who you

Yes, we mostly talk about asterisk, hardware, software, phones, people, events and things asterisk-related. Asterisk is a registered trademark of Digium. Asterix is a registered trademark of Ren? Goscinny. We never discuss that, though. Here's the short URL for sending out to others that might be interested: http://x2z.eu It lands here: http://www.VoipUsersConference.org Once again, I

You can listen or join the Asterisk Users Conference Fridays at 12:30 PM EDT Today's subject suggestions: FAX capabilities, what's your solution? Multiple asterisk server implimentation: ENUM, DUNDI or even two servers connected Your subjects? Share your ideas, ask your questions! See http://x2z.eu for instructions on how to join or listen

In several places online, and in the Asterisk F.O.T. book, there is a warning against using '_.' saying: "[it] should probably never be used". However, the need often arises act on numeric extensions that begin with *'s and #'s, and '+', and of course _X. does not match I have tried exten => _[0-9*#+]. but that seems to be the functional equivalent to _X.