search for: dbc_asterisk

My cell vm goes to asterisk, not the carrier. Apparently MWI is turned on/off with specially formatted SMS messages. Anyone know how to do this on a Treo 600? Having the phone light from Asterisk would be HUGE ... not to mention extremely cool. dbc.

I upgraded my 7960 to sip v 6.3 and my display time has now disappeared from the top left corner. Loadid: SW: P0S3-06-3-00 ARM: PAS3ARM1 Boot: PC03M030 DSP: PS03AT38 Here is the section dealing with time in my SIPDefault.cnf file. Does anybody see anything wrong with it or have any other ideas? # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)

Asterisk as a phone system makes perfect sense in a condo. You can get all the DID's you want and eliminate costs for the owners. You can offer standard FXO for people who don't care and IP sets for people who want to "upgrade" to feature sets. Your door openner is a piece of cake. 1. Create an option in your dialplan only in the "from-access-door" context that reads

I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxxxxxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it "forget" what it has programmed and

Kevin Walsh noted that his SPA-2000 takes time from his local NTP server in a post back on Fri June 25. Q: Where do you tell it to use NTP? I'm a bit confused as to where my SPA-2000 is currently getting its time. I told it GMT-5 in the misc section but it doesn't really tell me where its going for this. Is it just broadcasting looking for ntp? The net of my problem is that it is 1 hour

Did the console "Dial" command go away in recent releases? I just went to use is on my server (now at 1.0) and its gone. Help from the CLI doen't show it. -- David Cook

I'm looking to give the SPA-3000 a whirl as I'm having too much difficulty with the irq sharing thing inside the box. I'm reading the book but without having one in-hand to play with it appears a little obtuse at this time. Before I drop down my money can someone with some hands-on with one of these confirm if the SPA-3000 can: a) detect inbound distinctive ring (this looks to me like

It looks like this meetup group is becoming the venue for contact as most "Toronto" respondents have signed up here. Can I request that the remainder who responded to the "Toronto" call signup? Shidan is the organizer and has proposed a date. I'd hate to confirm it without the remainder having input - especially those with travel times like Andrew K. out in Listowel.

> Lito Lampitoc wrote: > > thanks for enlightening. So you mean, if I have 3 lines when the > caller > > dialled the first line and it was busy, the call will be diverted > to the > > next two available lines in random? > > > > I don't think it's random. I think its just sequential. If main > line > is busy, try second. If that is

On Wed, 2007-08-22 at 08:50 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Tue, 21 Aug 2007 21:01:50 -0400 > From: "David Cook" <dbc_asterisk at advan.ca> > Subject: Re: [asterisk-users] 99 bottles of beer > To: <asterisk-users at lists.digium.com> > Message-ID: <012501c7e458$05919370$10b4ba50$@ca> > Content-Type: text/plain; charset="us-ascii" > > On 8/21/07, Steve Edwards <asterisk.org at...

I am still haveing problems (echo) with my X100P but I'm thinking it has more to do with the server it is in which is not a negotiable item at this time. My question then is to the use of SPA3000's as a replacement from the FXO standpoint. 1. Can you setup the FXO port to recognize distinctinve ring and call a different context like you can do with Zap channels? Being able to call a

> From: Ronald Wiplinger <ronald@elmit.com> > Subject: [Asterisk-Users] Is there hardware to remote control > available? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <41C6D43F.5070201@elmit.com> > Content-Type: text/plain; charset=us-ascii; format=flowed > > I am looking for a

...ascii; format=flowed Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd ------------------------------ Message: 14 Date: Mon, 18 Jul 2005 12:03:28 -0400 From: David Cook <dbc_asterisk@advan.ca> Subject: [Asterisk-Users] IAX register confusion To: asterisk-users@lists.digium.com Message-ID: <1121702608.42dbd2d07532c@advan.ca> Content-Type: text/plain; charset=ISO-8859-1 I have been unable to understand the connection between an IAX registration for dynamic IP assignment...

..."ManxPower" Wieling) 24. Re: 99 bottles of beer (Russell Handorf) 25. Re: Saftware RAID1 or Hardware RAID1 with Asterisk (Steven) ---------------------------------------------------------------------- Message: 1 Date: Tue, 21 Aug 2007 21:01:50 -0400 From: "David Cook" <dbc_asterisk at advan.ca> Subject: Re: [asterisk-users] 99 bottles of beer To: <asterisk-users at lists.digium.com> Message-ID: <012501c7e458$05919370$10b4ba50$@ca> Content-Type: text/plain; charset="us-ascii" On 8/21/07, Steve Edwards <asterisk.org at sedwards.com> wrote: >...

Hi folks. I am using a X100P card and after some random amount of time of correct operation, say 8-20 hours, the card starts acting up and producng horrid sound quality which is all broken up. All other channels appear to work fine. One thing I noticed, is that "zap show channel 1" always shows the "Actual Hookstate: Offhook" as soon as the telco line is plugged in. Is this

The wiki page on Asterisk + Nat (http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions) lists the possible types of server/client relationships with one most probably interesting to us being #3. <snip> 3. Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk <snip> Then it goes on to say: * #3 Works with port forwarding and some header mangling

Quoting asterisk-users-request@lists.digium.com: > > Would it be considered trolling to start a thread on Cleaning Maple > > Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi? > > Let's not forget the weekly "tooques and telephony" segment, and a > review of > the best block heaters for your wi-fi fones. > Oh, we're gonna have a good time next

Quoting asterisk-users-request@lists.digium.com: > Date: Mon, 14 Mar 2005 22:23:54 -0700 (MST) > From: Greg Hill <gregh-asterisk@hillnet.us> > Subject: Re: [Asterisk-Users] How NuFone.Net's customer service > works. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: >

Regarding the SMP and interrupt issues. I know my machine is not running APIC now, but how do I determine if it is capable? Can I find out from the running system or is this something I need to know from the mfg? Currently the X100P shares IRQ with the secondary SCSI (yeah, go ahead and laugh). The box is a Dell PowerEdge 1400SC. Apparently the "SC" means Simplified Configuration and

I have been unable to understand the connection between an IAX registration for dynamic IP assignment and and the host definition. I have signed up with an ITSP for a DID. My ip is dynamic and although I have a dynamic DNS name, we are registering and outbound works fine. I'm at a loss to understand the relationship between the registration and the [section] definition in iax.conf that will