similar to: Help with Congestion error

It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Matt Florell Sent: Monday, March 13, 2006 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hey; I figured I would ask here as I seem to get better results. I am using the voip.ms <http://voip.ms/> VoIP service. I have no problem configuring my Asterisk server 1.8x to dial out with my Softphone. HOWEVER, for some reason, I cannot get inbound. All that I hear is a busy signal. I know this is not much for you folks to go on, but what would be a good place to start

Hello, I''m trying to get lossless VoIP traffic over my 3000k/500k ADSL line. Shaping outgoing traffic is no problem: I set total ceil for outgiong device (ppp0) to 450kbit and put VoIP into highest prio class. Even during full upload the voice is clean on the other end. Now I tried to get the same result for incoming data. I attached HTB to eth1 where the incoming voip traffic is

Hello Clair vulnerability scanner considers the latest version of CentOS mariadb vulnerable, because of RHSA-2019:3708 It states, that mariadb must be updated at least to the version "10.3.17-1.module+el8.1.0+3974+90eded84". CentOS' last version is "10.3.17-1.module_el8.1.0+257+48736ea6". Rpm/yum considers CentOS' version older, than RHEL's. % rpmdev-vercmp

Hi there, I've created a litle video explaining on how to setup some group policy to redirect folders using Active Directory Tools and Group Policy Management tools, it works and I've tested it so i decided to include it on my small youtube Linux collection. Hope you enjoy it and like; might be it can make it to the wiki video library. Thanks for Samba 4, it really _IS_ the thing.

Hi All, Please pardon me if it is the wrong list to ask how-to etc. I am having an issue with the Signed SSH keys. I am being asked for the passphrase for my signed public key, even though I don't have any. I am running CentOS7 with OpenSSH_6.4p1, OpenSSL 1.0.1e-fips 11 Feb 2013. 1) I have ca server with ca user keys (ca-user-key.pub) 2) I created user ssh rsa keys (user-id-org and

On Mon, 16 Mar 2020 at 12:17, koka miptpatriot <miptpatriot at gmail.com> wrote: > Hello > > Clair vulnerability scanner considers the latest version of CentOS mariadb > vulnerable, because of RHSA-2019:3708 > It states, that mariadb must be updated at least to the version > "10.3.17-1.module+el8.1.0+3974+90eded84". CentOS' last version is >

Hi , I open secure Session with X11 forwarding. when run any standard X applicattion like xterm xclock it come up without any error. I start my user X application it fails. On User application side it fails when "xtOpenDisplay" system call with localhost:10.0. On SSH server debug message are : > debug1: X11 connection requested. > debug2: fd 20 setting TCP_NODELAY >

Was this ever solved? How do I load a UTF-16 xml file into REXML? Thanks! Naren -- Posted via http://www.ruby-forum.com/. -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFF+G/Ez6ZCGd0@public.gmane.org To unsubscribe from this group, send email to

Hi, I am using a 30 VIP phone and a 12 SP+ phone with Asterisk. When I complete a call outside through the ZAP device, the phone does not go back to dial tone, even after I hang up. The line gets disconnected as per Asterisk console. But the phone stays in the same state like it is connected. The ZAP line is freed up and I could make calls from other phones. Only this phone just remained in

Hi, i am currently working on a problem where i need to plot latitude and longitude data on a respective county of UK. After this i want to plot altitude data to make a 3d surface on which then i have to plot my corresponding data. Please Help... thanks in advance Narender [[alternative HTML version deleted]]

I have a select element where the id field of a db table is populated. When I select an id from that select tag I want to show all the related names with that id populated in another select tag element in the same page.please help me. Thanks & Regards INDRANIL MUKHERJEE -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To

I cannot find any means of passing my own Callerid using Voxee. It always comes across as NO ID, or nothing, or unknown. I could not find anything on their website about setting your own caller id in the system either. (their web account pages). Is anyone here using their own Callerid information through Voxee? thanks

Alright, folks. I just deployed * into full production at my office. We have around 50 7905's, 5 7940's, and a handful of soft clients. We run a call center with around 15 agents. I also have a queue set up for the receptionists so that they don't get bombarded with calls. Everything seems to be working with a very few minor glitches. I firmly believe that the few problems we are

hi i am installing scalix mailserver while installing RPM i am getting this error: Failed dependencies: compat-libstdc++-296 is needed by scalix-server-10.0.1.3-1.fc4.i386 libcrypto.so.5 is needed by scalix-server-10.0.1.3-1.fc4.i386 libgssapi_krb5.so.2(gssapi_krb5_2_MIT) is needed by scalix-server-10.0.1.3-1.fc4.i386 libical >= 0.24.RC4.20050413 is needed

I am getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if

Hello, I'm planning to convert my phone system to Asterisk, as I've outgrown my TalkSwitch system. I have a few questions for experienced * users, most of which can be answered yes/no. Current Setup: - Talkswitch 48NLS (4CO/8Ext) phone system. - One CO line, two Vonage lines, one Voicepulse line connected to phone system - A third Vonage line directly connected to a fax machine - A

Hi I just signed up with voicepulse's voice connect service. then emailed me over configs for my extentions and iax i enter in all the info and when i start up * and do show registry it seems to be rejecting my login. Has anyone seen this before.. Any further insite will be greatly appreciated. thanks frankie (aim)cronparser (irc)crontibs 17006240093 -------------- next part

Anyone having problems with voxee since last few days or is it just me ? In peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or even LAGGED ( also shows high in peak time even when no high latency ). No problems with any other provider . Anyone else having same problem

We are trying to determine how to build out an IVR system we are working on. The system needs to be able to handle probably at most 5-10 concurrent calls at peak times. Other times we are just looking for a reliable service. For incoming calls we've been using BroadVoice VOIP and before that VoicePulse VOIP. VoicePulse's IAX service started dropping DTMF inputs that we were processing