search for: zapto

...n this? The mp3s play fine on any computer and haven't changed since they did work. Those wishing to hear for themselves, feel free to call extension 8800 at the number/addresses below. Thank you, Bryce Chidester Rhino Equipment Corp. bryce@rhinoequipment.com SIP: 305@rhinoequipment.zapto.org +1 (480) 940-1826 x305 IAX: guest@rhinoequipment.zapto.org/305 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050519/13cf4f29/attachment.htm

Hello group I'm working on getting festival installed and working on my FC1. I ran into a problem and after searching Google I found this message talking about a patch for Speech Tools and Festival http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html The above site does not have the files. Does anyone in the group have this patch? Marc Sutter & Reed Wade do you still

I was just browsing Digium's web site and noticed they are taking orders for the new IAXy. Has anyone purchased and tested one of these yet?? I have thought about buying one for testing, but want to make sure it isn't going to be a flop like the last one. Robert

I would like to setup a test number, that speaks back my phone number. How can I set this up? bye Ronald

I have asterisk with two zap channels which are analog ports off a T1. They each have a inward DID number If they are used for outgoing they show the T1 main number not the DID's number. Is there any way to send caller ID of the inward DID number not the main number Jeff

...to 10.215.0.0 hosts. So all seems fine except for the fact that I can''t access the shorewall router either from 10.215.144.48 (net) or 192.168.44.237 (loc). I tried ssh and http. However, pings to fw work from both net and loc. I placed a shorewall dump and some tcpdumps here: http://fhm.zapto.org/shorewall/shorewall_dump.tar.gz I would appreciate it if someone could give me a clue as to what is wrong. Thanks, Vieri ------------------------------------------------------------------------- This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual...

Hi ... probably one of those RTFM kind of questions (while I'd be happy to know where a good reference "FM" is :-) ) Has anyone an idea on how to disable the console sound driver. My problem is that a running asterisk is muting my speakers. Thank you in advance for your help Conrad

...s event notifications. > > I have also included a couple of sample plots (one that is a combined > plot of all my UPS and one that is an individual plot). If you'd > like to see how all of this fits together, the following URL is real > time: > > http://ewilde-webserv.zapto.org:8480/cgi-bin/upsstats.cgi > > You may click on the links to see how the plots change and there may > be a UPS or two with log info as well. > > If you'd like sample configuration files, let me know and I can send > you them as well. > > Thanks for all the work tha...

Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number

Anyone knows where can I find spandsp? Official site seems permanently down... TIA, Simone.

Hello All, I recently put in a zaptel 1fxo/1fxs card. I am experiencing heavy static. Even with the pots line disconnected, if I do a dial I still get static. This way I know it's not the line, but rather something on the card. I tried alternate pci slots. This card has a power connector, does anyone know what the power requirements are? The unit is in a small case with a 2.4ghz p-4 and

Hi, What do I need to do to get asterisk to NOT pickup a Zap channel when it rings? The channel in question is used for outbound calls only, and all incoming calls are answered by an analog phone elsewhere in the building that does not run through asterisk... so.. either make it not answer.. or make it delay for like 90 seconds.. I've tried wait's.. but it still seems to pickup the

Is there a softphone for windows that supports IAX? I can't seem to find anything out there...maybe im looking in the wrong places... Jeromy Grimmett VoipEmpire.com jeromy@voipempire.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/e668136a/attachment.htm

Hi, I have to connect 30 phone lines to my asterisk server, can somebody help on how I have to do it ? I have a TDM405P and one TDM400P with 4 FXO ports. Do I have to use 8 TDM400P ? Or, is there another way to do it ? Thanks, Angel.

Please can someone send me the .tar.gz file of spandsp, the site is offline and i didn't find it anywhere! Thanxxxx! Roland Zagler mailto:r.zagler@fog.at @fog smart partners

Hello, for the umpteen time Patrick Turley <pturley@rocksteady.com> sent: > <eom> Too bad this wasn''t on the, then I could set by watch to it. Steve _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

When I try to patch the Makefile for asterisk with the Apps_makefile.patch from Spandsp I get the following error. patching file Makefile Hunk #1 FAILED at 47. Hunk #2 FAILED at 76. 2 out of 2 hunks FAILED Has anybody seen this.

...######################## ######## with nat = yes ################################# -- (11 headers 0 lines)--- Destroying call '08e0cde70bda25a526f32d5a50727f33@192.168.1.200' asterisk1*CLI> <-- SIP read from 151.38.43.46:20300: REGISTER sip:192.168.1.200 SIP/2.0 Route: <sip:pasqu.zapto.org;lr> Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bKegpl50ntmlhc69ar0ackhhd From: <sip:208@my_ip>;tag=5vk550gjc5hc7o4v0ack To: <sip:208@my_ip> Contact: <sip:208@192.168.1.99>;expires=3600 CSeq: 1133 REGISTER Call-ID: 5hdiwAD3oIdgxgapi3CfvgbWvzQ0PU Supported: sec-agree Max-...

I'm trying to install the SPANDSP software into Asterisk to support incoming (mainly) Fax. I'm following the info in http://www.voip-info.org/wiki-Asterisk+Fax. I downloaded and installed the spandsp software from ftp://ftp.opencall.org/pub/spandsp/ and followed the directions in several documents listed on the on the Tiki page. I get down to patch < Makefile.patch that fails with

...gt; ------------------------------ > > Message: 5 > Date: Tue, 24 Jan 2006 13:27:40 -0200 > From: Pablo Lorenzzoni <pablo@propus.com.br> > Subject: Re: [Icecast] Total number of listeners > To: icecast@xiph.org > Message-ID: <20060124152740.GA6817@eriador.local.spectra.zapto.org> > Content-Type: text/plain; charset=iso-8859-1 > > You mean something like getTotalListeners() in > http://svn.xiph.org/websites/dir.xiph.org/index.php ? > > Since all our relays are known and authenticated, I think I could query > them all in a regular basis to check h...