similar to: New Member

Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a "basic" Linux/Asterisk system? Just re-boot and config. -- James Taylor jltaylor@metrotel.net 903-793-1953 --

Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

I've seen the embedded posts. Is anyone running Asterisk on the MINI ITX? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

Hey all, quick question: does asterisk work okay in a Cygwin environment? I want to install it on my cygwin setup for local testing/demoing and save me the hassle of using a pure linux machine........ As long as it doesn't take a huge huge performance hit from running out of Cygwin, then I'll have a go there for a start confirmation appreciated! thanks -- C h r i s E a r

I've got echo problems. *** I'm looking for paid support. *** I'll accept free support, but don't mind paying if someone really knows what they are doing. I've read the wiki, etc. Played with the settings in zapata.conf Using V400P PSTN->_T1->_ASTERISK->_BROADVOICE->_PSTN ECHO ON CALLED PHONE PSTN->_T1->_ASTERISK->_T1->_PSTN NO ECHO

Arrrrgh I hate trying to sift through all these messages and keep track of the various threads going on ......... Who else on here prefers the newsgroup/threaded approach? If you haven't already, check out news.gmane.org for mailing lists turned into newsgroups readable by news readers....... only problem being that this list requires list membership before

I have a client that would like to use asterisk to link their multiples locations together. However, if a person in the remote office dials 911, How can the 911 operator determine WHERE the emergency is?? Since all calss would be going out of the PRI in the main location, the police/fire trucks will show up at our COLO!! I know that there are some that are doing this multi site setup, how did

Hello, I am trying to solve a problem that I can foresee when I deploy Asterisk into a few SOHO situations soon. In Nebraska and in my area of Western Pennsylvania we have violent thunderstorms in spring and summer and sometimes very heavy wet snow in Winter. Both type of event will take out the power of a period of 30sec to 36-hours. So no UPS system would be able to handle a system for over

Anyone done a chat line app? -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956

What started out as a good thing for the community has veared it ugly head and will come back to bite us in the ass. I give my respect to the two companies that decided to put themselves 'out there' and attempted to bring 'real world' certifications of knowledge in an area that is unregulated, open, and currently has no measurement of how much one person has over another. What

Anyone out there have Verizon or SBC as the local exchange company in your area? I'm in the process of negotiations for additional interconnect agreements with these carriers (we're a wireless carrier). I'm looking for Asterisk folks who want a T1 (or several) for a PSTN gateway and would like to "network" their Asterisk boxes. This can take place at little or no cost

Does anybody developed Predictive Dialer using Asterisk/Digium PBX? Another question: does anybody developed an Dialer using the X100P board? Julio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030718/be051d49/attachment.htm

Robert, > Is there a product to simulate a PRI trunk? (Robert > Goodyear) TDMoE emulates a T1. ;) Once the TDMoE link is up, Asterisk just sees 24-lines that appear to be a T1 instead of having to deal with all of the complexities of VoIP. This is useful, since probably 75% of the utility of VoIP is really just the fact that it can run over a network. It's also handy because it

With the increased traffic as of late, I'm wondering if it is time to split the list again. Specifically I am wondering if it should be split along the various VoIP protocols and zap hardware, then leave a general list that does configuration other than VoIP related? The hope is that those asking SIP or H323 questions could get help from the various supporters while the main list can deal

I configured this once now I forgot what I did. Two Broadvoice accounts. Incoming is simple - just use the phone numbers. Outgoing: Dial out on a specific line and/or set up the groups and select the other "line" if the first one is busy? -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956

I am looking for help in writing the scripts for numeric paging via Asterisk. Here's the general flow: answer lookup D-I-D if "on" prompt with "please enter your telephone number" store in variable prompt "please hangup now" Shell to SNPP executable and send message to server. I have the code in "C" for the command line SNPP client. Three parameters

Hi- Can someone help me conceptualize how I could setup an * box which features both English and Spanish prompts? I know it's possible to configure the box to use one or the other, but how could I setup a multi-lingual box? Any suggestions? Thanks in advance, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, Is it possible to implement an application that satisfies the following scenario using agi and php? - user picks up phone - he wants to redirect all his calls to the cellphone - he dials *400 for example and all the calls addressed to him are diverted - he comes back to office next morning, dials *500 and his redirection is gone Note: my extensions.conf is entirely rebuilt at 5

Hello All, I have 4 Broadvoice lines. If I call any of the lines it shows that is coming from the first line. exaple register=XXXXXXXXX1@sip.broadvoice.com:passwd:XXXXXXXXX1@sip.broadvoice.com register=XXXXXXXXX2@sip.broadvoice.com:passwd:XXXXXXXXX2@sip.broadvoice.com register=XXXXXXXXX3@sip.broadvoice.com:passwd:XXXXXXXXX3@sip.broadvoice.com

Hi, I have succeed using Asterisk on VMWare on an Athlon@1GB with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem