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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 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 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 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

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

I'm looking for an inexpensive way to connect 20 analog phones to asterisk. I could get a bunch of Linksys or Sipura boxes but was wondering if there is a more cost effective way? I came across the Mediatrix 1104 and even the Mediatrix 1124 but that comes out to be almost $100/port. I might as well buy inexpensive IP phone. Does anyone have any suggestions? Thanks, Waldo

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

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%

If you ring into * and leave voicemail It does not reset the line Any ideas would be appreciated Regards Mick

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

Hi All, I'd like to setup asterisk for a small call center. I need basic functionality for a small call center, including but not limited to: 1) Multiple queues with different rules for each (e.g. some queues, people may leave messages in, others people will just have to wait until the call is answered) 2) Call monitoring (all call - inbound/outbound - must be recorded) 3) Agent Barge

This may be a stupid question, but I couldn't find anything on the wiki about it or on google. I have about 5 agents in my call center. I want them to login using agentcallbacklogin. The reason being is that I don't get so many inbound calls. We mostly make outbound calls. Therefore, during the times where we don't get calls, I want my agents to manually dial out. When calls

It's me again. I'm using XTen X-Pro softphones which have the ability to handle multiple conversations simultaneously. When my agents login and calls come in, everything works fine, as expected. Calls are directed to available agents, and when no agent is available, the caller sits in queue. However, since the X-Pro has the ability to handle multiple calls, how can I make the

Hi guys, I'm testing the sending of a URL to an XLite softphone when a call is in queue. See the output of the CLI below: -- Executing Queue("Zap/69-1", "q_sample|tT|http:// www.google.com/") in new stack -- Started music on hold, class 'default', on Zap/69-1 -- outgoing agentcall, to agent '1000', on 'Local/ 1000@agents-1b94,1'

Has anyone had any experience with the Soyo G688 phone? I'd like to use it as a agent's phone. Is it reliable? How well does it work with *? How's the quality? Features? Thanks, Waldo

Is there a way to define multiple contexts for agents/queues such that in a multi-tenant environment, there could be two different, say, Agents 1000? I'm setting up a multi-tenant configuration and I'm giving each tenant a web-based interface to define their own agents and I wouldn't want to restrict one tenant from choosing an agent because another tenant (which the tenant

I'm trying to figure out how much call load I can put on a Dual Xeon 2.4 Ghz Asterisk server acting strictly as an IAX2 call director, as show in the diagram below. The idea is that I have N number of gateway asterisk servers connected to the PSTN using T1 Digium boards. Then, I have M number of servers where my agents and/or telephone extensions (whether they are IAX or SIP

I've been reading a bit about Enum and DUNDi and still have something not very clear to me. This is a HYPOTHETICAL scenario: I have 4 asterisk servers. All of them are handling registrations of both SIP and IAX2 UAs. SIP agents are being load balanced by something like SER. I have another server in charge of load balancing IAX2 UAs registration (some sort of dynamic firewall telling