similar to: rmirror.digium.com host unreachable

Can I use queuemetrics with asterisknow? I mean, if I modify the dialplan to use queuemetrics (I still don't know if it's possible), will I loose my changes when the time comes to do a conary update of the asterisknow package? thanks, -- ------------------------------------------------------------ Erick Perez ------------------------------------------------------------

Hello Asteriskies, Has someone tried www.asterisknow.com ? What is the package manager used? And what is the added value compared to the well maintained debian based asterisk ? Thanks, -- Cheers, Maxim Veksler "Free as in Freedom" - Do u GNU ?

When I bring up the Asterisk GUI in AsteriskNOW, using IE7, it displays a message at the top "Your browser is not supported by this version of GUI!", and "We recommend using Firefox". Does this mean that it is known NOT to work under IE7, or just that it is insufficiently tested to be guaranteed? It's easy to ask techies to use Firefox, but if we are trying to sell a

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Has anyone else experienced a problem with the LED for span 1 on a TE410P or TE405P? I had a TE410P on which the span 1 LED would not light red, but once the span was connected, it did correctly light green. I RMAed the board to our UK distrbutor and received a replacement. However, the replacement board displayed the same problem! Wondering if it was related to the computer I was putting it

Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer

Hello, Trying to upgrade (from asterisk18-1.8.6.0-1) to the latest RPM version from Asterisk repo I found that asterisknow-version is needed by package asterisk18-core-1.8.7.0-2 How could this be explained? Best regards, Ioan ######### [root at localhost ~]# yum update asterisk18* -x asterisknow-version Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * base:

I posted a message to this list about 50 minutes ago. I received a posting acknowledgement pretty quickly, and it showed up in the mailman list archives, but I still have not received a copy. Looking at some of the other recent messages I have received, they have also suffered a delay of 20 minutes or more on lists.digium.com before being sent out again. Are there any plans to beef up the

Hi, I currently have some systems on AsteriskNOW 1.7 & have been happy with its clean simplicity & reliability. Are many people here using AsteriskNOW 2.0.x? How do you feel about it? Did Digium stick with their previous philosophy of keeping everything very vanilla & making it clean & simple for someone who understands how to manage CentOS, FreePBX, tftp, ntpd, etc. but

Hello, Is it possible to set things such as parts of config files are edited though AsteriskNOW GUI while other parts remain "hand editable" ? AsteriskNOW website include screenshots but not much information (such as user manual) beside that. This thing was the one that kept us from using freepbx (let me say I don't mean it's not possible with freepbx : I mean we couldn't

I have some rpath/foresight machines I''d like to puppet''ize but AFAIK no one''s made a conary package provider ... I suppose what I most care about are config files for now -- because we are using the "conary way" of keeping our packages updated. That said, it would be nice to know that there''s something out there? If there isn''t one ... I

Hi, I'm a complete newbie to Asterisk and have been reading through documentation and sites for the last couple of hours trying to understand what to do to start my learning curve with Asterisk, and am very confused. For starters, what is the difference btwn the 1.2 and 1.4 branches of Asterisk? I can't seem to find a document that describes the changes. Secondly, what is the best

I just downloaded and installed the AsteriskNow appliance (http://www.asterisknow.org) . This looks like it has lots of promise. Anyone know what the secret is to being able to actually login to the root console? thanks, Geoff

I currently have a customer that a previous employee setup with Gentoo\Asterisk. I'm looking to migrate to AsteriskNOW. They have a custom menu, which I would assume is easily replicable in AsteriskNOW. The only other thing I can think of is the sound bites for the menus. Does anyone have any advise or migration recommendations for this move? -------------- next part

I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error

Hello all. I was hoping someone out there might have some advice or suggestions regarding an upgrade from an archaic Asterisk version. I've been given the daunting task of upgrading a very old Asterisk-1.0.x install to a recent LTS version. I'll also need the install to have high-availability and failover support. From my research, it would appear that AsteriskNOW-3.0 might be my

Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository:

I am currently using AsteriskNow v2. What I am trying to accomplish is having all calls from an area code go directly to the person responsible for that area. While searching for a solution for this I did come across a post that had a few examples. So Josh at extension 1902 would receive all calls from the 808 area code. exten => s,1,GotoIf($${CALLERIDNUM:0:3}" = "808?1902|1)

Hi list, Anyone knows about the date of the official (stable) release (v1.0) of AsteriskNOW??? It's supposed to be at the end of this year, which is very close now with no signs of it. Thanks... Raul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071218/c543b407/attachment.htm