search for: xvoip

Hello all, Our Asterisk unofficial online forums (http://asterisk.xvoip.com) are moving forward and are alive. Thanks to all Asterisk guru's who helped to achieve it. Everyday more and more new members are joining this community, however not everyone is using mailing list or knows how to use IRC, this is why they are coming to web-based forums. We believe that...

well, asterisk unofficial forums are online since Nov 2003. http://asterisk.xvoip.com We have 500 registered members as of today and forum is alive. About 1000 uniqe visitors are hiting forum everyday. We will post link to your forums to provide users with more info. http://asterisk.xvoip.com I agree, the Wiki is Step #1 for any Asterisk user. The Wiki is a great ref...

Did anyone here use the * forums over at asterisk.xvoip.com? I've been unable to connect for a few days now and was wondering if anyone knew if they're down for good. It'd be a shame if they are since * newbs like me need every resource we can find. Joel Moore

...t 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 asterisk-elite@xvoip.com 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 : asterisk-elite@xvoip.com R...

I've set-up a mirror at ftp.gbnet.net/pub/digium It's (sort of) available via www.gbnet.net/public Mirror's the whole of ftp.asterisk.org/pub Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19

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

Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example: If * detects Answering Machine or Voicemail, hangup call & the AGI will log (ANSWERING MACHINE DETECTED) and at that point,

...config.conf to: ; Extconfig.conf for realtime configuration [default] voicemail => mysql,my_db,voicemail_users The error goes away, but the config still does not work. Can't find anything on the new Wiki pages on the subject though. --------------------- Also posted here: http://asterisk.xvoip.com/viewtopic.php?t=764&start=0&postdays=0&postorder=asc&highlight=

...ly name of person and we only possible way to do it , via mailing list. Thanks, Alexander E-Mail: akandelaki@stealthtele.com ************************************************************************ ***************************************** Unofficial Asterisk Forums : http://asterisk.xvoip.com <http://asterisk.xvoip.com/> ************************************************************************ ***************************************** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail...

...ly name of person and we only possible way to do it , via mailing list. Thanks, Alexander E-Mail: akandelaki@stealthtele.com ************************************************************************ ***************************************** Unofficial Asterisk Forums : http://asterisk.xvoip.com <http://asterisk.xvoip.com/> ************************************************************************ ***************************************** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail...

Hi all, I can see huge traffic here over 400 post in 4 days. My proposal is to create asterisk newsgrup proposal or phpBB forum what do think about it ? BR, Corvin btw. I'm admin of phpBB Forum (slackware forum - polish language), nearly 900 users. I think if someone will prepare it good it can be great project. (but I have 7 person team).

I have been using Asterisk 1.0-RC2 successfully with a channelized T1 circuit for quite a while now but after upgrading to 1.0.0 callerid no longer works properly. Debug output from a channel shows what actually is received through DTMF from the carrier: << [ TYPE: DTMF (1) SUBCLASS: * (42) ] [Zap/9-1] << [ TYPE: DTMF (1) SUBCLASS: 2 (50) ] [Zap/9-1] << [ TYPE: DTMF (1)

...config.conf to: ; Extconfig.conf for realtime configuration [default] voicemail => mysql,my_db,voicemail_users The error goes away, but the config still does not work. Can't find anything on the new Wiki pages on the subject though. --------------------- Also posted here: http://asterisk.xvoip.com/viewtopic.php?t=764&start=0&postdays=0&postorder=asc&highlight= ===== Regards, Jason Goecke jason@goecke.net ----------------------------------------- NL Mb: +31.622.471.436 US Ph: +1.360.526.0542/+1.720.946.6451 US Fx: +1.801.409.4351 UK Ph: +44.844.986.9270 DE Ph: +49.2...

...tn | s | 2 | Hangup | | +----+----------------+-------+----------+-----------------+-------------------------------------------------------------------------------------------+ Any ideas, does Realtime have problems with Goto? Or am I just overlooking something? Also posted here: http://asterisk.xvoip.com/viewtopic.php?t=806

I keep getting the following Auto-congesting message whenever I try to dial from an X-Lite SIP phone to another one within my LAN. It's a real basic configuration but I am unable to figure out what is happening: localhost*CLI> -- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in new stack -- Called jtest May 2 11:47:58 NOTICE[1133742896]: chan_sip.c:1019

Hi All Total noob on the list so all help appreciated.... I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones.... The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......