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Hi all, What are the best and free tools for remotely adding, removing users on Asterisk server and also for monitoring the status of the Asterisk server, like how many users are logged on etc. I need tools for which I don't have to pay. Thanks, Zeeshan A Zakaria www.acabling.com <http://www.acabling.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050825/0b659ff8/attachment.htm

...already. Zach -----Original Message----- From: Dovid Bender [mailto:asteriskdigium@yahoo.com] Sent: Monday, February 06, 2006 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help on queues Yes. The wiki and voip-info.org --- Zach A <zeeshan@acabling.com> wrote: > Hi, > > Is there any detailed guide/tutorial source online on queues? > > Zach > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com > -- > > Asterisk-Users mailing list > To UNSUBSCRIBE...

Hi. I'm wondering if it is possible to make asterisk honor the Presentation Restricted bit on incoming PRI calls. Ideally I'd still like to see the number in the CDR but we can't let users hear restricted numbers in their voicemail messages, etc. The docs only seem to talk about outgoing calls. Thanks...

Hi, Is it possible to do this in extensions.conf to put a caller in queue and dial an agent's extension so that he knows that somebody is in queue waiting to be answered. This agent will be a remote agent and extension will dial his cell phone. Thanks Zach A. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ay, March 10, 2006 6:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] how to check if ztdummy is working properly? what kernerl are you using ? when did you did modprobe zaptel and modprobe ztdummy it loaded without a problem ? --- Zach A <zeeshan@acabling.com> wrote: > Hi, > > As I am having problems with MoH and have tried > everything to solve it > and nothing worked, I was thinking maybe the timing > source, i.e. > ztdummy, is not working properly and that is what is > causing problem. Is > there any test to ch...

...other phones, the you will need to figure out how to configure them to do the same...basically the phones will send out packets with the Internet Routable addresses and the port info configured for them. Kevin J. Steil Steil Technologies -----Original Message----- From: Zeeshan [mailto:zeeshan@acabling.com] Sent: Thursday, January 12, 2006 11:46 AM To: Asterisk User List Subject: [Asterisk-Users] How to register a SIP phone on Asterisk behind NAT Hi everybody, One of my client's Asterisk box is behind NAT. They have only one public IP on which they have their router. I can access the As...

Which models of Diva could work with CAPI and asterisk? Thanks Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di gw@adcomcorp.com Inviato: sabato 1 ottobre 2005 23.46 A: asterisk-users@lists.digium.com Oggetto: RE: [Asterisk-Users] Diva Nope. At least I tried and never could get it

...already. Zach -----Original Message----- From: Dovid Bender [mailto:asteriskdigium@yahoo.com] Sent: Monday, February 06, 2006 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help on queues Yes. The wiki and voip-info.org --- Zach A <zeeshan@acabling.com> wrote: > Hi, > > Is there any detailed guide/tutorial source online on queues? > > Zach > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com > -- > > Asterisk-Users mailing list > To UNSUBSCRIBE...

> Hi everybody, > > This question is confusing me for some time. From selling point of view > to a customer, calling asterisk a PBX doesn't look right. According to > the definitions of PBX or PABX, Asterisk is not just PBX but much more > than that. My question is, how should I introduce Asterisk to a > customer? I don't want to call it a PBX. > > Thanks >

I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I had to remove the hardware echo cancellation module from the card. We are only using the 1st span of this card right

Hi everybody After setting up trunk with FWD, all I get on my Asterisk box is message saying that all circuits are busy now, try your call later. Even 612 (time) says the same thing. Why is it that and how can I fix it. Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I can't figure out how to change User Login and Password for AMP. By default it is user:admin and password:maint. Anybody knows how to do it? Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050831/c8ef57b2/attachment.htm

Hi everybody, I've proxy server IP, user ID and password. Now I need to connect to a remote Asterisk server as a SIP using my Asterisk @ Home box. That Asterisk server will make PSTN calls for me. I think I am making mistake while setting up the Trunk because when trying to make calls, it give all circuits are busy error. When I setup Sipura adapter, which is relatively easier to setup,

Hi, I've Sipura 1001 phone adapter. In the settings it has separate Line 1 and Line 2 tabs, which apparently means it can control two separate phone lines. I've Asterisk@Home server and want to setup two different extensions for two phones, i.e. 201 and 202. After doing all this, I can see in Info tab that both lines are registered but only one phone gets the dials tone. Am I doing

I see this error when trying to make calls from my asterisk server. How can I solve this problem. Sep 2 08:25:03 NOTICE[1403]: -- Registration for '1096377@209.167.97.164' timed out, trying again

Hi, When a SIP client registers on Asterisk server, why it expires after certain amount of time?

Hi, What port does mpg123 uses to play music on when it starts MoH after asterisk has put called on hold? Zach A

Hi, I've set up FreeWorldDialup on my asterisk server but when I dial the service numbers, I get message '486 Busy Here '. When I dial any other number, it says 'All Circuits are busy now'. What is the problem with my settings. I've followed all the instructions step by step. Zeeshan -------------- next part -------------- An HTML attachment was scrubbed... URL: