similar to: 4+ Port FXS Analog Device

Hi all, Assuming 1.0.7 is the latest stable version, how/where can I find out the different CVS revisions available and a description of what has been patched/updated in each CVS revision so I can decide whether to leave my 1.0.7 installation as is, or if I need (or think I need) to patch it with a CVS version? Thanks, Waldo

Hello list, We have some kind of a problem with our Asterisk installation. We want to be able to publish different caller id when placing outbound calls through the PSTN. We have Asterisk with TE410P and T1 from FDN Communications. The problem is that all our outbound calls show our main number, regardless of what we set with SetCallerID, even using CallingPres with all possible

I was playing today with the different queueing strategies in queues.conf when I noticed the following behavior. I have 4 agents defined in a queue in queues.conf. These agents login using AgentCallbackLogin. The strategy in the queue is set to leastrecent. I place four calls into the queue and * sends only one call to the least recently used agent. If that agent does not pick up, the

I'm wondering if anyone can recommend one over the other. I'm mostly interested in running open source solutions, so I would prefer if your recommendations are within the open source arena. Basically, I contemplated the idea of using SER as a NAT Helper and possibly as a SIP server for a portion of our user base. We prefer to have Asterisk in the mix because of the additional

Hi list (again), I have another question which I have not been able to resolve from neither the wiki nor Google. I've been able to configure a multi-tenant setup of asterisk for 2 small call centers with no problem, by simply playing with contexts (which I guess is how everyone else is doing it). The problem I have is that I've only been able to configure one global agents.conf

I have two asterisk machines. One of them has a Digium board (server A) and the other is simply using ztdummy (server B). Server A is running on Debian and Server B is running Gentoo. Server A is running Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 and Server B is running Asterisk 1.0.7. The problem I have is that when I try to transfer a call into a meetme room in server B, it simply hangs

Can anyone recommend a company that does professional Asterisk recordings for things like IVR, greetings, MOH, announcements, etc? Thanks, Waldo

I'm planning on setting up some remote agents and before doing so, I did some simple PING tests to measure latency. The average latency I got was 250ms. Does anyone have experience in terms of quality of calls when there is such high latency? Can anyone comment? Thanks, Waldo

I'm looking for a good, reliable, and cheap 4-port FXS ATA. Does anyone know of one that works with Asterisk? Thanks, Waldo

I was reading on the wiki that Asterisk runs very solid on NetBSD. Can anyone comment? What is the definition of solid? Who is running Asterisk on NetBSD and which version of Asterisk are you running? Also, I know there is limited support for Digium cards on NetBSD, but is there any support at all? Would a TE410P work in NetBSD? I want to build a very simple VoIP to TDM gateway. My idea

I was reading on the wiki different possibilities of automatically restarting asterisk every so often. In some places, people mention they restart it once a day other on shorter or longer intervals. I believe the main reason people are doing this is because of possible memory leaks. I'm running a system for IVR services. It's not a heavily loaded box, but there is almost always

I was browsing the applications developed in zaptel and came across zttest. After I run it, I get the following: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.000000% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%

Hi all. I'm evaluating a mediatrix 2fxs 1102. seems great (it has also supervised transfer, that's very needed in office environments and works well). the only I thing I cannot make work is the auth to my asterisk server. If I don't set a password into the mediatrix and *, I can call out, but still the registration goes wrong. using a password, nothing works. I've done some

Hi, I have a client that have currently 400 analog phones (all wired w/ Cat3). I need multi-ports FXS interfaces but I only find 24 ports FXS (like Mediatrix 1124) but it's a little bit expensive to get 15-16 box (~408 FXS ports). Someone have suggestion to link all theses phones to "channel bank" ? What equipment to use ? Currently all 400 lines are terminated in M-66 blocks and

I am attempting to use a Mediatrix 1204 to interface to multizone paging from Asterisk. I have 4 different paging interfaces and want to connect each of those 4 to an FXO port on the Mediatrix. The desired result is to be able to issue some SIP dial string from asterisk, seize the FXO port on the Mediatrix and then have a speech path. I am able to place calls over the Mediatrix when it's

Hi! Has anyone on the board successfully installed a Mediatrix 1102 or 1104 as a SIP peer on Asterisk? I'm trying to configure different user accounts on each FXS port, but I'm having authentication problems; Asterisk is saying the client is not authorized. Interestingly enough, I can dial a "9" and make a local call through the Mediatrix. Thanks! chris --------------

I recently purchased a Mediatrix 1124 from an auction of a company that went out of business. It came with nothing other than the unit itself. In digging thru the Mediatrix web site, and various google searches, it looks like it only supports SNMP setup, and only with their software (or the correct MIB). However, Mediatrix doesn't appear to let you download said software or MIB from

Hello! We are located in the UK, and we are planning to replace our old pbx with an asterisk based pbx. For outgoing calls our present pbx is connected to three PSTN lines which all have the same number. Internally, the pbx caters for quite a few extensions, and each extension can make outbound phone calls. Only very rarely does our call volume exceed three simultaneous connections (inside to

FYI, spoke with Grandstream this morning, the GXP-2000 release has been delayed again. Looking like April now before these hit the street. -- Cory Andrews Senior Partner VOIPSupply.com +++++++++++++ V: 800.398.VOIP X22 F: 716.630.1548 E: Cory@VOIPSupply.com

I'm proposing an asterisk configuration and considering the use of multiport SIP/FXS gateways (instead of T1 cards and channel banks). I'm looking for products similar in function to the Cisco ATA-186, but with more ports. I've seen the manufacturer's web pages for the Audiocodes MediaPack (http://www.audiocodes.com/) and the Mediatrix (http://www.mediatrix.com/) access devices.