similar to: Is Asterisk ready for Prime-Time?

What does it take to get ztdummy to work correctly? I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel 1.4.9.2 Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs. Problem is playback() does not work. So then I stop zaptel, asterisk runs and playback() now works. However, meetme()'s dont work. I need ztdummy I'm pretty sure for that. I am

>> All too common and largely undocumented. I had this same problem. >> >> Installing ztdummy changes Asterisk to use it for timing of playback, >> apparently. Removing ztdummy "fixed" the problem. To get it all to >> work, I had to upgrade to to at least kernel 2.6.23.11 (previous >> versions are either missing options are just broken.) > >

All too common and largely undocumented. I had this same problem. Installing ztdummy changes Asterisk to use it for timing of playback, apparently. Removing ztdummy "fixed" the problem. To get it all to work, I had to upgrade to to at least kernel 2.6.23.11 (previous versions are either missing options are just broken.) After doing this, I recompiled ztdummy and it worked. Note

I'm trying to understand something that just doesn't seem to compute. How can companies like Cisco justify selling their hard phones for as much as they do? I know there is a matter of recouping R&D costs but when you look at the iPhone with all its amazing features for less than $500.00 it just doesn't make sense. Am I the only one that thinks this? Roy Anciso Director of

I've been evaluating Asterisk for a while, and things seem to be going very well. The issue of redundancy and automatic fail-over is now on my mind. I searched the archives and googled for solutions, but didn't really come up with much. We'll be using queues (modified), which precludes some of the standard redundancy solutions, since the queue needs to know all the agents

Hello All, We have been experiencing some ongoing reliability problems with Asterisk for quite some time, and I am trying to find out if anyone else has experienced the same problems. We are running asterisk 1.4.17~dfsg-2+b1 on Debian Lenny, with a Digium PRI card, and have approximately 120 sip peers, mostly Snom 360s, with a few Grandstream GXP2000 and a handful of Handytone 486 units. The

Is anyone aware of a service where we can lookup phone numbers to determine a name and/or name + address available in bulk? We want to look up every number called to our call center, so it will be tens of thousands per day. Services that charge 3 to 5 cents per lookup will get way too expensive very quickly. Thus, I'm looking for a service that can either license a database or

Hi * users, I would like to extract the information present in the SIP REFER message that comes to asterisk. Would SIP_HEADER() allow me to do that ? I have used SIP_HEADER() for extracting the to and from SIP headers previously. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

Hello all, i read a few articles online about the possibility to setup a "buzzer" door system to PBX using asterisk! currently my setup contains asterisk of course, and a sipura 3102.. what do i need to get such a feature done?! or should i ask if its possible?! _________________________________________________________________ Connect to the next generation of MSN Messenger?

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

I've encountered an odd situation with Asterisk 1.4.19 that I can't figure out. If I dial an extension via a Cisco AS5400 with the "g" option to come back, when I then Dial another extension after that, we don't get audio from the caller. There are no firewalls, no routers, no anything but a network switch between. The calls come in as SIP from the Cisco and

Has anyone got a Premisys Slimline channel bank working with a Cisco AS5400 or similar? I'm not sure if my unit is bad, or what. I'm using FXS Loop Start. Calling the port connects immediately without ringing the attached phone. If I pick up the phone, it's connected and I can talk to the caller. Hanging up has no effect. I can see the bit transitions (0101 to 1111 when I go

On Mar 5, 2008, at 5:46 PM, asterisk-users-request at lists.digium.com wrote: > If you are running a call centre (large or small) using Asterisk, > I'd be > interested to know how you log your agents in & out: > > E.g. > > - Do you use AgentLogin (to force calls onto the agents, perhaps)? > - Do you still use AgentCallbackLogin? > - If you use

I've been searching to a solution to this for a while and can't figure it out, perhaps someone has done something similar. I have a Cisco AS5400 sending SIP traffic via PCMU / ulaw directly to my Asterisk (1.4.19-rc2) box. Jitter and latency are incredibly low on my lightly loaded switched gigabit ethernet network. One Asterisk uses Zaptel and a Digium card, and DTMF recognition

On Oct 16, 2008, at 2:36 AM, asterisk-users-request at lists.digium.com wrote: > I want to call an extension like 88888 and invoke an external C > program upon > calling, pass an constant integer like 1 to the C program. > > What I have done is: > > /etc/extensions.conf: > exten => 88888,1,system(/usr/local/src/parallel/fire 1) > exten => 88888,n,

I've had it with Dell server garbage. They seem to change RAID controllers as much as I change socks, and then the controllers don't work with Linux, unless you load a new driver. They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot! Their sales folks are dumber than rocks, and they change them more often than I

Note that the newish SPA962 has 6 appearances and a color screen. I've noticed that the bright color screen does impress people when they first see it. PoE is also very nice and web provisioning was quite easy. I've yet to try a more automated provisioning method on it. I know that getting the polycom's to auto provision wasn't very straight forward. I do provision some

Jeremy wrote: > What kind of luck are people having with the Web-MeetMe control? The > condition of the page on the voip-info wiki makes me a bit nervous about > putting Web-MeetMe into a production environment. Use of MeetMe has > really taken off here since installation and I need a scheduling and > provisioning system for PIN numbers etc. Are there any other solutions > out

What's the consensus? Should winbind even be considered for production use? Looking back through the archives of the Samba lists, there's a lot of doubt about it. Where people have had problems, there are more often than not no solutions given. When I look at the man page in 3.0.28, there are litterally blank spots awaiting completion, and lack of documentation of sometimes-essential

Hi all, I've started playing with kvm on CentOS 5.5, with not much success so far. In a nutshell, I have the same problem as http://lists.centos.org/pipermail/centos-virt/2010-April/001854.html I followed the RHEL5 virtualization guide to set up a bridge interface br0, and then used virt-install (rather than virt-manager - I like automation) to set up the vm. It hangs at some point.