similar to: Any Aheeva Users?

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

I have echocancel=yes in zapata.conf but when I do a zap show channel 1, I notice echo cancellation is turned off. I followed the article that talks about the order in which the statements need to be in zapata.conf to get echo canceling to work: http://lists.digium.com/pipermail/asterisk-users/2005-June/110615.html But it is still not working. Does anyone know how to get echo

Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service

Hello All, I do have asterisk installed for a call centre with aheeva application and i would like to know how to configure the sound for the inbound calls and if there is any possibility for agent to receive a file with the phone number and name of clients: For your information there is no problem related to the outbound call An help would be appreciated Kind Regards Salah. --------------

All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4

Hi, For a customer, I am looking for a good and reliable Asterisk based system. Five servers will be installed at different locations and will be linked together with each other. This system will work as a call center as well. It has to be a stable and reliable. Customer also needs GUIs for system administration and agents call activities. He also wants video conferencing Please help me select

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 11:36 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio? On Tuesday 21 March 2006 11:19, Adam Robins wrote: > All switches and routers give

As I understand it the current plan is to make it possible to losslessly transcode VP3 video to Theora video. In my experience, one of the "features" of VP3 is it drops frames in the event that there is little/no movement in a frame, or if "drop frames" is enabled, to drop frames if the data rate is getting too high. I understand that the way that VP3 does this in

hello all, I have installed centos 5.5 ( linux text) and I have updated it with # yum install bison bison-devel================?ok # yum install ncurses ncurses-devel==========?ok # yum install zlib zlib-devel===============?ok # yum install openssl openssl-deve=======?ok # yum install gnutls-devel============ ==?ok # yum install gcc gcc-c++============?ok # yum install newt

Hi all, We are running Asterisk 1.0.0 with a TE410P. Very often we exerience calls dropping in the middle of the call. I enable the full logging and saw a couple of suspicious messages right before the hangup. Thos could happen on a Zap-IAX2 bridge as well as on a Zap-Agent bridge... I see Nov 23 09:08:36 DEBUG[-1274020944]: Bridge stops because we're zombie or need a soft hangup:

Hello! Continuing my PRI saga - I have a PRI setup and appears to be answering calls OK, but my carrier is cutting all the calls after 15 seconds. For example, when I call from my cell phone, it goes straight to a busy signal - however the CLI shows the call coming in and being answered. Additionally, when I call from another ground line, it will ring once or twice, again show as answered, but

Before studying your configs, what have you tried so far? Did you change this? Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4 to span=2,0,0,ccs,hdb3,crc4. Here is the documentation on voip-info for why it may be the cause of your issues http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax span definition format:

Hi, I have questions about the Polycom 601 and side card.... 1) In the side card the lights all time off... But all functions it's ok. I need help with extension module of polycom... All works fine... But lights not work.... So... I don't know when any person or extension is busy... Any ideas? , Olger On 3/27/06 11:34 PM, "asterisk-users-request@lists.digium.com"

What kind of call center: inbound, outbound or both? how many lines per agent will you have? what kind of trunks will you be using? do you need to tie into an existing database? do you want screen-pops? MATT--- -----Original Message----- From: Zeeshan [mailto:ztahir@gmail.com] Sent: Tuesday, August 02, 2005 7:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Dear All, Does someone do anything on your Linux (or UNIX) mail servers to convert darn proprietary MS Outlook winmail.dat attachments your users may receive (occasionally if lucky) into readable e-mail format. I know quick answer to my question (good quick answer would probably be: just trash them). Still, being involuntarily immersed into corporate world (even at Educational institution),

Hello, We use and develop the astGUIclient suite. It is Open-source(as in GPL) and offers Inbound and Outbound call center functions with reports, ACD, monitoring, recording and very basic IVR scripts. Complex IVR functions need to be custom programmed within Asterisk but that is not really that hard. It works across multiple Asterisk servers and we are using it currently at 5 locations including

Dear All, we've been having some intermittent problems with students who's home directories are on a Sun E450 running Samba 2.0.6. I'm not totally convinced that the problem is caused by the server, but may be the network. However, I've had a look at the log.smb file and I'm seeing some error messages that I've not come across before: [2001/02/23 18:20:40, 0]

some of you may find this helpful: I've uploaded a short (5 second) raw clip in yuv4mpeg format, associated audio, and batch files to exercise the encoder & decoder examples. In addition I've included the file as compressed (test.ogg), and a longer version as well to test playback sync. Notes: to use MPlayer with the -vo yuv4mpeg option, you need to get the latest release and compile

We manage our own VOIP solution using Asterisk. Has anyone had success with an IP-PSTN provider? I'm looking for someone to terminate SIP calls to the PSTN in the Seattle, Washington area. (vice-versa as well if possible) Yes, I could do it myself via asterisk and digium cards but I would like to consider other options. Any opinions? Thanks, Chad -------------- next part

I haven't actually had many questions up to now. Despite the all-encompassing CP_INSTANCE monolith, things have been relatively easy going. Now we get into real grit: First off, CBitman looks like it has endianness issues; it's packing into host-order 32 bit arrays (the comments and symbol names seem to indicate that this was originally byte-based packing code that got upped to 32 bit.