similar to: single dialplan for multiple Asterisk machines

This is a resend - customer complains that some phone systems they call don't respond to key tones, please advise. Thanks Rob. ----- Original Message ----- From: Softprofit Solutions To: asterisk-users@lists.digium.com Sent: Monday, November 24, 2003 7:34 PM Subject: tone detection problem Some commercial systems don't hear the tones dialed from an analog handset, same

Hi All, I hope this is the right list for this sort of request. I'm wondering if you all could recommend (or are) an asterisk integrator. I've been following the lists, etc, and have played with the software, but just don't have the time to really figure it out, nor to deliver a solution in a fixed time. I need someone who can help me spec the hardware and configure * for a small

Hello, Good news on the Polycom front for those that are interested. It looks like we may get a dedicated Engineer for Polycom/Asterisk!!! Happy Day! Here's the message I got tonight: Matt: I heard back from our VP of Engineering- she is prepared to have an individual dedicated to working on the Digium- Asterisk project. Can we discuss again Friday or mid next week? Scott Willard

I am having a problem with softphone registration, having read the list and watched it for a while for similar problems I just cant seem to figure out the problem. Using SJPHONE or DIAX , I can make outgoing calls but I can't get them to register with asterisk, I have other sip devices registering OK-7940's. From the list and the digium web site this seems to be a straight forward set up

Hello, Just confirmed this on my end, because of the massive changes that have been made to callerID handling in asterisk 1.0.5 many of the features of the astGUIclient suite will not work on this new version. The latest stable version recommended is Asterisk 1.0.3. We will work on trying to find ways around the new callerID rules that the asterisk developers have put in place and hope to have

Hello, We've released another update to our Asterisk GUI Client suite: http://astguiclient.sf.net/ Screen shots: http://astguiclient.sourceforge.net/screenshots.html The client suite runs on both Windows and UNIX and includes a dialer (the suite is not an asterisk configuration tool) In addition to the usual bug fixes, this is mostly an update for the VICIDIAL dialer application.

Has anyone on the list been able to locate and try out the 1.1.0 firmware? It was released in November, but I have yet to get my hands on it. The Polycom site has way more docs online, but the link to the firmware only brings up the release notes. -sb

I have a TDM400P with one FXS and one FXO.. how many liscence(2) I will have to buy? Thanks

I have a couple of live T1s sitting around and they are not ISDN(like most of the people that are using Asterisk seem to be using), they are regular old 24 channel, robbed-bit, E&M wink start, D4AMI T1 circuits. Can I get these T1s to work with a T100P Digium card and asterisk? Searching through the lists and the documentation I haven't seen any examples of how to configure this kind

Hello, I have received yet another new phone today, the ClipComm 101 (http://www.clipcomm.co.kr/eng/e_product/e_product_voip_ip_phone.html) I bought it for $165 directly from the Korean Manufacturer(No US distributer yet). Here are the features: - Built-in NAT functionality, you can switch from Hub to Nat, great for home DSL/Cable users - This includes some limited port forwarding

Hello, Would you please suggest me, where can I buy g723.1 liscence in cheap. I might need a liscence for 10-50 channels. Thanks,

Hello, I recently tried an upgrade of CVS on my test server today and found that the res/res_parking.c file is completely gone. This is where I had to go into the code every time I do an upgrade and change the code to allow for doublehash transfers instead of single hash transfers: That means that you need to hit the pound key twice to initiate a transfer instead of once. Because of our inbound

We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer

I hope this question is adequate for this list I use the nnet code from V&R p. 348: The very nice and general function CVnn2() to choose the number of hidden units and the amount of weight decay by an inner cross-validation- with a slight modification to use it for classification (see below). My data has 2 classes with unequal size: 45 observations for classI and 116 obs. for classII With

Hello, Has anyone on the list been able to successfully setup a Polycom Soundpoint 500 IP phone? I am getting failed registrations, and the Polycom documentation is not very precise. Their web interface isn't helping much either. Thanks in advance, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I was using the doublehash.patch that Iain Stevenson had created back in August to change the transfer key from a single hash "#" to a double-hash "#". It always patches properly, but when I went from CVS 2004-01-12 to Asterisk 0.7.1 it doesn't seem to work anymore. I've attached the patch to this email and I use the following command to patch it: patch -p1

Hello list, I am having trouble listening to GSM files created by Asterisk using a browser. I am noticing that some of my users succeed in listening to them and some others don't. I guess it is something of a codec problem that does not seem to be installed on all machines (though they are all WinXP). Anybody knows what one should do to listen to GSM files? I send files through the

I''m having some trouble with the instructions at the link below: http://instantrails.rubyforge.org/tutorial/restarting_the_dev_env.html I had started at the tutorial at http://instantrails.rubyforge.org/ tutorial/index.html and had gotten as far as creating the cookbook2 exercise and had gotten to the "Welcome aboard you''re riding ruby rails" screen in explorer, but

Hello. Was just curious to know if anyone is working on a graphical interface to Asterisk using X windows, or something else similar. Thanks! David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031113/2718fd7e/attachment.htm

Hello, In the last week I've been getting a lockup about every 2 days. during the lockup the people that are on the phone can keep talking, but noone can initiate any kind of call internal or external. I went into the manager interface and tried a Action: Hangup and Manager gave me a Success message back only to see that the Zap channel was still active in the "show channels"