similar to: Round-robin with Queue

I have been getting that message also. I have been using various versions of CVS head since Feb. 2005. -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jeremy Gault Sent: Friday, October 14, 2005 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,

We have a TDM400P installed here with four FXS modules. It works well except for a couple of issues: First, I have a Panasonic KX-TG2431 telephone (so others can reach me when I am in o ther parts of the building) hooked up to one of the FXS ports. When the other end hangs up, I get the usual CPC disconnect signal. After the CPC, sometimes it will go to a dialtone, and other times a

Hello, I run Asterisk in a 100% VOIP installation with the Polycom IP-500 phones. Every once and a while I have problems with either dropped calls between Asterisk and my provider, or invalid RTP audio streams with phones behind NAT. I have had a few Asterisk developers look into my installation and even my provider check my setup but still am having problems. They tell me that I need to

I asked my telco to release caller name on the PRI. Earlier they were releasing only the phone number. I still did not see the name, but only the number in caller id. Actually I now see number twice. When I inquired with them this is the response I got: "I ran a trace on your TG. I see that your switch is picking up the call so fast that it is not able to pick up the name. The

Does anyone know of any plans to add an intercom/all-page feature in *? The few SIP phones that have auto-answer capability would be better if Asterisk could broadcast one leg of a channel to many legs at one time. Thank you, Steve Maroney

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

Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I am running into some problems here with PrivacyManager. We used to use it without any issue, but now there seems to be several problems. We are currently running Asterisk 1.2.4. First, it seems that if the user does not press the pound (#) key after entering their number, PrivacyManager will fail. I have the minlength set to 10, and

All, Quick question that hopefully someone out there will know the answer to... We were previously running Asterisk 1.4.(something) (I forget which one) on Debian. Due to an office move, I am temporarily routing our calls through an Ubuntu box that I have. It runs Asterisk 1.4.17-dfsg-2ubuntu1 (basically, what came with Ubuntu.) Here's the problem I am having: We are using Polycom

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

> From: Martin Devera <devik@cdi.cz> > Subject: [LARTC] (E)SFQ suggestion > Hi, > just simple note. Maybe it is already in progress :) > > There are attempts to replace hashing routine in SFQ to > consider IPs or ports. > What about to use HRR - roundrobin around bunch of IP > adresses and then smaller WRR for ports per IP ? > It would solve both

Hi, I dont really know if this is the right place to post this question..If its not pl let me know... Here is what i am looking for. I have two machines with two ethernet cards.One of these machine has an ftp server(vsftpd). When i request a file from the first machine i want that machine be able to start an ftp, and when each of the packet of the file go to one of the ethernet cards, i want

All, I have a weird situation here and haven't been able to turn up any useful information in searches, so I thought I'd post to the list. Essentially, I have a customer who wants us to forward some of their calls to various cell phones. Normally, I'd use FollowMe() for this (that's how most of our customers are set up.) However, this particular customer has some requirements

Hi, I have configured a call queue as follows: [sales] music = default strategy = roundrobin timeout = 20 retry = 5 reportholdtime = yes announce = queue-sales member => Agent/9021 member => Agent/5901 member => Agent/9020 member => Agent/1114 Now, I would expect all calls to this queue to be delivered first to 9021, then to 5901, then to 9020, then to 1114 (all agents are logged in

Hi there. Currently, our network design has two ISP lines and 3 subnets for LAN. Below are some details :- eth0 - isp1 eth1 - isp2 eth2 - subnet1 eth3 - subnet2 eth4 - subnet3 What i wanted to do is to assign incoming port 80 to our local squid server running on the firewall itself and assigned it to eth0(ISP1). I think it shouldnt be a problem as /etc/shorewall/rules provides a sample of the

Hi, I had saw a few patches to be applied in order to get the firewall to block based on the application. At the moment I am having problem to block ICQ and Yahoo Msgr. Even I only open a few ports like, the ICQ still able to connect through port 80. Any add-in module for shorewall or anyone able to do this? Regards __________________________________________________ Do You Yahoo!? Tired of

Hello all. I''ve been struggling to QoS VoIP at our site and have a successful implementation at this point. Basically I had to set aside enough bandwidth for VoIP by placing all other traffic behind an HTB (multiple classes and queues behind it). Everything is fine. Here''s the diagram: ------- | eth | ------- | --------

Trying to install samba 3.0.3 on my Linux server, to serve 4 PCs, all in my house, serving my family. Each of 4 users will have a home directory, plus I'll have at least 2 read-only shares. I have basic connectivity, but can't get some of the clients to behave the way I want them to. Specifically, I want to have full admin access for myself for all directories. Everyone else

Hi,  I tried fitting loglinear model using the glm(catspec). The data used is FHtab. . An independence model was fitted. Here summary() and fitmacro( ) give different values for AIC.   I understand that fitmacro( ) takes the likelilhood ratio L2(deviance) to calculate AIC and uses the formula AIC= L2- d.f(deviance)*2 and this AIC is used for comparison of nested models. (Am I right?)   The value