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...X100P card - TDM400P card with 1 active port - Voice-mail - Use of Asterisk through the console I am using Mandrake Linux 9.1, in case anyone needs to know what Linux distribution this test used. I wish success for those that are having problems with the recent CVS. -- Thanks, Timothy Soos XQL, LLC

http://www.nero.com/us/631911127302064.html Have you all seen this? Its a SIP softphone put out by the people that do the CD burning software Nero... Check it out it works with * Dave

I've spent some time on the Wiki adding documentation on all asterisk applications from the cli 'show application xxxx' commands. I've also added some cross references and pointers. http://www.voip-info.org/tiki-index.php?page=Asterisk If you find this useful, please go there and help us build a reference database. A Wiki is a wonderful collaboration and documentation tool -

Hello All, I am a newbie looking to learn about Asterisk. I'm new to IVR and all that goes with it. I would like to know if it is possible to grab the number of an incoming call, have Asterisk, or third party software return the call with an automated voice message allowing the original placer of the call to select another person to call (eg "Select 1 for Bob") then have

Hi I read through the archives but could not find much reference to * using SIP on TCP instead of UDP for signalling. Can * be configured and if so how. My service provider will only accept SIP signalling on TCP. Thanks Master -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello. Well I finally rx'd my dev kit (new batch of TDM's apparently. I'm on Mandrake 9.1 There were no hardware install instructions, it would have been nice to know whether the 4-way power connector was to be used or was for some other future or expansion purpose. It came with a floppy disc, no label and it wasn't even write protected. The only readme file was

...yping the numbers ? My dect does give me > the dialtone then. > (allthough i don't have the problem that digits go to quick, but maybe you > can tweak that in an advanced menu setting on the phone). > > Tj > > > --__--__-- > > Message: 6 > From: Timothy Soos <XQL@americanisp.net> > Organization: XQL, LLC > To: asterisk-users@lists.digium.com > Date: Sat, 13 Sep 2003 05:13:54 -0600 > Subject: [Asterisk-Users] Monitoring an active channel > Reply-To: asterisk-users@lists.digium.com > > Hello All, > > I am still having some diffic...

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

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I've put together the first attempt to defining an RDF metadata vocabulary for use with the CD Index/MusicBrainz/OggVorbis. If you care about metadata issues, please take a look at: http://www.cdindex.org/MM I've included a section for video specific stuff, but everything that I originally had in there is being covered by the MM:Contributors section. The Contributors stuff will allow

I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indications.conf from /etc/asterisk -

Hello, I would like to request opinions for: What is the best SIP phone? Please give a rating from 1 to 10 (10 being the best) for the following categories: Reliability Ease of deployment with Asterisk Sound quality Please be sure to supply the manufacturer's name and the phone's model number. If you have an opinion on this topic, I would like to hear it. -- Thanks, Tim

Hello All, Is it possible to monitor and record a SIP to SIP call? If so, how? I gathered from some previous posts this would not be possible. -- Thanks, Tim