similar to: Agent Queues/XTen X-Pro/Multiple Call Appearance

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

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'

Same here Michael and the PocketPC version seems unaudible with any codec; early days trying that though. Simon -----Original Message----- >From: "Michael Van Donselaar"<mvand@neb.rr.com> >Sent: 22/04/03 04:10:24 >To: "asterisk-users@lists.digium.com"<asterisk-users@lists.digium.com> >Subject: Re: [Asterisk-Users] Xten - Free windows SIP

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

I've tried every softphone available for linux and I've found that Xten X-pro for Windows seems to load fine under the WINE API, the problem is - it wont work the sound... how do i get the sound to be emulated so I can place and receive calls using this software? the program's options seem to list my audio devices. keep in mind i have to use a microphone and sound output. is there a

Anyone used the combination above? We are and it sounds like crap. The audio drops out in regular intervals which suggests that someone's g729 isn't doing its job correctly. I'm blameing XTen cause when I make a ulaw call that gets converted to 729 using digium's 729, calls sound fine. Anyone else? Similar experiences? -Matthew --

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'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

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'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 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 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 have a tdm 3xfxs and 1xfxo, aslo I have a setting with 1 snom 190 and 2 xten line. I can call from the snom to the ptsn line at the fxo port ok. I can call from the ptsn to the xten lite phone. I can call from the xten lite to snom but what I CAN`T do is; Call from xten to ptsn. When I dial from the xten, I can hear the dialed party, but he cannot hear me... Tips? Help? What I'm

All, I have my * box NAT'd with all ports forwarded that are SIP related (based on Wiki). I also have nat=yes, externalip=WAN address of firewall, internalip=LAN network of *. I have my Xten soft phone on a PC which is NAT'd behind firewall with ports forwarded. I have also followed instructions on Wiki for Xten. I can authenticate fine, and sip show peers shows my extension is OK,

Ok I have a question. I have Xten-lite working with our Asterisk system and I am able to make and get calls. But the main problem is the sound is very choppy and sometimes it cuts off words. I have tested it with ulaw and alaw as well as GSM. They all do the same. ulaw seems to work better. I also have an ATA-186 which works great without this problem. Here is my Sip.conf settings.

[this post is just as much for the archives as it is for immediate consumption] Somehow I had completely missed the announcement that XTen Networks had released a SIP client for Mac OSX. Well, they have. It seems to mostly work, even! Very cool. The User interface leaves something to be desired, with it's menus trying to force me to use cell-phone sized windows (!!!!) to manipulate

hi there! I have a couple phones connected to a sipura ata and if I go into *- IVR, I press options on the regular phones and it all works fine and dandy. then I connect an xten softphone, a new extension in my dialplan, I dial the ivr, * asks me to dial something to go through it, I press keys on xten, but nothing happens, * just times out through as if I did not press anything! is there some

Waldo, How do you let your customers manage 'their' PBX. I too have a setup like you. However, I installed a * server for each customer, via vserver. I'd like to now what kind of software/webbased package you use for this. I also have SER installed as a front-end server for the * servers. But, as I'm still not very into SER, don't know exactly how this fits in. Should I use

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