search for: mstorck

I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk

Hello, we have set up e164.lu as a test zone, as the delegation for 2.5.3.e164.arpa hasn't been completed yet. For all those who want to call the numbers currently availble directly via SIP, please use the zone name in your enum.conf. If you decide to use the zone, please tell me at mstorck@luxadmin.org, so as soon as the 2.5.3.e164.arpa zone is ready, I will mail you, so you may disable querying e164.lu. Best Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, rout...

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2659 0873 -------------- LuxAdmin powered service --------------- http://www.Gateway.lu Your gateway to the net Adv...

Hey gang, There must be any easy solution for this but my mind is frazzled on compiling 2.4 with RTC as module. Bleh. Currently extension 9000 is our VoicemailMain(@company) line. Some employee's are complaining that the old system was better because you didn't have to enter your mailbox number and that instead the old system took you right to it. I figured there was something similar

...), the same is true if the users uses the redial button. So my question is, how can make asterisk or the ZAP channel wait a little bit longer before he claims a match against a number in the dialplan... Thanks, Marc -- CTO Marc Storck MS Networks SA mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -------------- LuxAdmin powered service --------------- http://www.Gateway.lu Your gateway to the net Ad...

...iddle of 1 debug packet, to continue over 20 seconds later (if it continues). The actual CPU load is "load average: 0.00, 0.00, 0.00". I cannot find the problem, maybe someone over can help me! Thanks, Marc -- CTO Marc Storck MS Networks SA mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -------------- LuxAdmin powered service --------------- http://www.Gateway.lu Your gateway to the net Ad...

...ovided number (3) '061706161' ] (in this case TON = 2) Does a variable like ${TON} exist??? Or how can i read that number? If this would have to be implemented I'm willing to fund a bounty! Regards, Marc -- CTO Marc Storck MS Networks SA mstorck@msnetworks.lu Internet Service Provider http://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 ------------- MS Networks powered service ------------- http://www.Gateway.lu Your gateway to the net...

...g to server 1 and the RTP stream to server 2. How do I configure asterisk to work with that type of installation. It seems they are using NexTone as SIP Signaling and RTP servers. Can someone help me??? Regards, Marc -- CTO Marc Storck MS Networks SA mstorck@msnetworks.lu IT Service Provider http://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --------------- MS Networks powered service --------------- http://www.LuxAdmin.com Hosting and housing solution...

Could anyone quote a price for the following project. We should be able to generate a specific (say 12Khz) signal at certain intervals (calculated using a price/rate table on a mySQL database) DURING an ongoing conversation. The conversation is to be marked (start and end) with specific signals as well. This is a requirement for special hotel applications where a device counts the signals to

I had this idea, and after looking for something like this already in progress, I found another guy who tried to start it... But I was unable to contact him, and his project seems to be dead. But, I believe it is possible, and I wanted to know the opinion of the experienced... So, let's go: I got an asterisk server setup to receive free calls from US to Brazil. The problem is that at my work,

...sion (CONTEXT, h, 1) exited non-zero on 'SIP/1.2.3.4-08442ea8' > cdr_odbc: Query Successful! When I call from another VoIP device it just works fine! I hope someone has some help! ;-) Regards, Marc -- CTO Marc Storck MS Networks SA mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -------------- LuxAdmin powered service --------------- http://www.Gateway.lu Your gateway to the net Ad...

At time to time I receive some junk faxes from some advertising companies that play smart and don't provide any TSI number so I can not bock them by the number in Hylafax. Despite calling their Fax Removal Service 1-800-... number several time they refuse to obey my request. So I would like to setup a small script or context loop in extension.conf if possible and maybe run it overnight; maybe

Does any one know what happened with voice conduits? I have been trying to reach them for nearly three weeks now. Their voice mail boxes are full and writing email to them does not get any returns. Thoughts or sightings are appreciated. -- R.J.

I have been using Sixtel from the beginning of the year and service was getting worse and worse. Yesterday I tried to access the website to get the CDR and I got an error saying that the domain no longer exists. I checked the whois and it says that the domain is on hold. Have they finally folded? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel:

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Hola! I'm using asterisk as H.323 -> PRI gateway. First call goes thru ok, second concurrent call fails with: Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to context [ip2pri] -- Executing Dial("H323/ip$192.168.32.25:60271/984", "Zap/1/9541163107100") in new stack Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to

Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same

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.

...nstead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart.... Currently I will have to answer the line to do that. Is there a way to do this with asterisk? Regards, Marc -- CTO Marc Storck MS Networks SA mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 ------------- MS Networks powered service ------------- http://www.Gateway.lu Your gateway to the net Ad...

...;all the T1 crap then pass calls to asterisk as SIP/IAX, that would be >>awesome in our situation. >> >>Right now our only solution is 8 T1s into a 5300, then SIP to asterisk. >> >>-Matthew >>----- Original Message ----- >>From: "Marc Storck" <mstorck@luxadmin.org> >>To: "Asterisk Users Mailing List - Non-Commercial Discussion" >><asterisk-users@lists.digium.com> >>Sent: Monday, December 13, 2004 8:09 PM >>Subject: Re: [Asterisk-Users] Ethernet Channel Bank (Comming Soon to a >>NOCNear You!) >&...