search for: metrotel

Hi, I have installed Elastix 1.5.2 (Barranquilla, Colombia (TELCO: METROTEL)) with a TE220P (2xE1) and TDM2400P (16FXS), openr2 is included in 1.5.2 version. The outcoming calls are ok, but with incoming call i have an error: ERROR*[*14972*]* chan_dahdi.c: Chan 2 - Protocol error. Reason = Multi Frequency Cycle Timeout, R2 State = Seize ACK Transmitted, MF state = Categor...

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 --

...another trunk group and different T1's that go to a MAX TNT first: PSTN->_T1->_MAX_TNT->_VOIP->_ASTERISK->_VOIP_PHONE ECHO ON VOIP PHONE g711 PSTN->_T1->_MAX_TNT->_VOIP_G711->_ASTERISK_IAX_GSM->_ASTERISK_IAX_GSM->_VOIP_PHONE_g711 NO ECHO -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956

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

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

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

Greetings, I'm ready to start and setup Asterisk. Any preference on which Linux to use? Windows & FreeBSD in use here. I'd like to get up and running as quickly and easily as possible. Thanks -- James Taylor jltaylor@metrotel.net 903-793-1953 --

...t; 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 are passed: server ID message. James Taylor MetroTel 903-793-1953/1956 ________________________________________________________________ Sent via the KillerWebMail system at mail.metrotel.net

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

...ers] Another Asterisk Certification > > > > Alternate Certification > > > > For those of you who can't (or won't) shell-out the $3000+ for the 5 day > > certification class, > > here's a quicker way AND IT'S HALF THE MONEY! > > > > www.metrotel.net/asterisk.htm > > > > Asterisk is a good product. > > Some people "need" certification. > > > > A mature "product" needs "certified" professionals. > > Asterisk is maturing. > > > > Remember the Certified Novell Engine...

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 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 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

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 added the three lines to the sip.conf file based on the latest changes from broadvoice. I can receive incoming calls but cannot place any outgoing calls. The error I get is: *CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569 -- Attempting call on SIP/Broadvoice/5068012 for application Playback(demo-congrats) (Retry 1) Mar 8 08:35:21 NOTICE[29290]:

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

...erisk 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 (other than the hardware). Compensation for incoming calls can offset most or all of the T1 monthly expenses. -- James Taylor jltaylor@metrotel.net 903-793-1953 --

Hi There I am trying to get the following setup going. Two PCs (gateways) running Linux FC2 both have aDSL connections running. And both run the Asterisk PBX software, I defined the IAX trunks in the iax.conf on both sides, the systems seem to call eachother but never get the handshaking completed ... The consoles keep reporting errors on both sides. Is there a example for doing this ?? --

Hi everyone, Could someone help me on make my Asterisk registers to a Gatekeeper. I have compiled the chan_h323.so and it seems to be working. What I want to know is how can I "route" my SIP clients to a single account on a remote Gatekeeper. I have tried a lot of conbinations but nothing happend. For example: my account number: 123456789 ;extension.conf exten =>

I was ZapBarging an extension the other day (sorry, CVS Head as of last week) and "forgot" to put down the phone when the conversation ended. To my consternation, I then started to listen to a totally different conversation - I think that the zap channel got re-used. Is there any way to force the zap barge to drop when either party hangs up ? Thanks. Julian