similar to: Asterisk help please

I need to monitor the states of my T1/PRI Zap channels. Specifically, I need to be able to programmatically determine whether a channel is unused, carrying an inbound call, or carrying an outbound call. Using the manager interface, I can easily tell whether a Zap channel is used or not by looking at the results of: Action: Command Command: zap show channel <x> Or: Action:

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

I am having two problems, possibly related, while performing pre-deployment testing of a Samba/OpenLDAP PDC with data that was vampired from an NT4 PDC. The Samba server fails to become a local master browser, and password change attempts (from a Windows client) fail. I followed Samba-Guide/ntmigration.html (taking some liberties with various items of configuration), ending with step #19.

ast_waitfordigit that accepts milliseconds as input. Douglas Garstang wrote: > Admittedly I have not used the ExternalIVR app. Is it any good? > > I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, > it can do it, but boy it is UGLY. There's also the fact that you can't > call Backgound() in a macro, which forces you to use Read() which >

Is this completely open source? And all it's dependencies? https://github.com/trustmaster/SugarAsterisk Is it free as in free speech and free as in free beer? I know there are a few variations of SugarCRM. We're currently using an Asterisk hosted PBX with Cisco hardphones. Conceivably, we could run something like SugarAsterisk on the local network, and it would connect with the

Groups for implementing =91GSM Gateways=92" which leads me to believe (or h= ope at least) that more than one phone can be paired to a dongle. Dongles are so cheap I guess it doesn't really matter other than more complexity. Anyone know for sure? --=20 Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) --001636c92545598447046efba265 Content-Type:

Hello, I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks

I'm using SixTel as a test (Opened account w/ $10) and am happy with them so far... In their basic service package, they don't charge a monthly fee, and it's outbound only, and you get charged for every minute. I paid for a DID, which is $1.50 or so per month, and it lets me receive inbound calls, which I also pay for by the minute.. I don't mind this for a service like this

n active project, than a dead one. Otherwise who is going to patch vulnerab= ilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What

with an active project, than a dead one. Otherwise who is going to patch vulnerabilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

hi all, how to set the caller id facility for the TDM400p card. Please help me thanks, sandeep.s ----- Original Message ----- From: <asterisk-users-request at lists.digium.com> To: <asterisk-users at lists.digium.com> Sent: Tuesday, January 15, 2008 3:09 PM Subject: asterisk-users Digest, Vol 42, Issue 51 > Send asterisk-users mailing list submissions to > asterisk-users at

Zitat von Steve Totaro <stotaro at totarotechnologies.com>: > Are you using the wifi on on the cellphone? The peer IP is showing as > 192.168.200.3 which is not a routable address. Unless things have changed, > double NAT configurations do not work. Hi Steve, My Asterisk is behind a NAT, but my cellphone was NOT in NAT, but direct in Internet. But maybe my Provider does a

On Sun, Jun 7, 2015 at 10:05 AM, Luca Bertoncello <lucabert at lucabert.de> wrote: > Zitat von Steve Totaro <stotaro at totarotechnologies.com>: > > Are you using the wifi on on the cellphone? The peer IP is showing as >> 192.168.200.3 which is not a routable address. Unless things have >> changed, >> double NAT configurations do not work. >> >

I use the E-series Nokia phones on my Wireless LAN. The e series have sip agent On 8/20/07, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

BRIStuff would work on, but does not because US BRI is a bit different than other BRI around the world. About keeping secrets, which he doesn't, he has said that he had these patches for a long time now. At any rate, it is his code to whatever he wants. If he wants to charge for it, do consulting, or keep it a secret, it is not your business nor do you have any say what he wants to do with

Great for Chan_Mobile and GSM modem for SMS in Kannel or if Asterisk supports SMS over GSM modem. I know chan_mobile had SMS in the future at one point but have not revisited the project since. "America Movil's MVNO TracFone Wireless quietly unveiled a prepaid, nationwide unlimited offering for $45 per month that includes unlimited text messaging and 30 MB of data."

This is concerning an Asterisk 1.4.18 server. We have approximately 70 DID numbers. Incoming calls are placed into the "incoming" context (by zapata.conf) and are routed based on the dialed number. I want to do some manipulation (CallerID name override) to all incoming calls before they are routed. I would prefer to avoid duplicating the necessary code in each DID extension stanza,

Hi, Quick question to the real world. Approx what specs would I need on server to handle 95 ZAP or Dahdi -> SIP gateway using G729 on the SIP to carrier side (nothing else, just media conversion)? Does the latest Asterisk/DAHDI significantly improve these numbers over say, Asterisk 1.2.X? Sure, there is plenty to read but nothing I could find quickly on my exact needs that was clear and I