search for: aheeva

Just looking for unsolicited thoughts on the Aheeva product? Anyone have anything to say? Thanks, Steve Totaro http://www.asteriskhelpdesk.com

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

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 asterisk restart Any help would be appreciated, sorry if it is a newbie question. Regards, Am -------------- next part -------------- An HTML attachment was scrubbed... URL:...

...===?ok i have an issue with this kernel ==============> # yum install kernel kernel-devel kernel-smp-devel kernel-smp======? there is no package available when i make (./configure make ) in order to install (zaptel-1.4.12.1) i have this message make[3]: Leaving directory `/usr/src/aheeva/zaptel-1.4.12.1/menuselect/mxml' gcc -o menuselect menuselect.o strcompat.o menuselect_curses.o mxml/libmxml.a mxml/libmxml.a -lncurses make[2]: Leaving directory `/usr/src/aheeva/zaptel-1.4.12.1/menuselect' make[1]: Leaving directory `/usr/src/aheeva/zaptel-1.4.12.1/menuselect' Gen...

...because we're zombie or need a soft hangup: c0=Zap/52-1, c1=Agent/1400, flags: No,Yes,No,No Nov 23 09:08:36 DEBUG[-1274020944]: Bridge stops bridging channels Zap/52-1 and Agent/1400 Nov 23 09:08:36 WARNING[-1274020944]: No class: none Nov 23 09:08:36 VERBOSE[-1274020944]: == Spawn extension (aheeva, 5142232986, 5) exited non-zero on 'Zap/52-1' Nov 23 09:08:36 DEBUG[-1274020944]: Expression is '1' Nov 23 09:08:36 VERBOSE[-1274020944]: -- Executing GotoIf("Zap/52-1", " 1?2:3") in new stack Nov 23 09:08:36 VERBOSE[-1274020944]: -- Goto (aheeva,h,2) Nov...

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

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 Reg...

...s still not working. Does anyone know how to get echo cancellation to work? We have Asterisk 1.0.7 and Zaptel 1.0.9 with 2 PRIs using a TE410P board. Here is the output from the CLI: zap show channel 1 Channel: 1*CLI> File Descriptor: 25 Span: 1 Extension: Dialing: no Context: aheeva Caller ID string: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Owner: <None> Real: <None>LI> Callwait: <None> Threeway: <None> Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law:...

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 tremendously IMPROVED their call quality with IAX2. Headsets are Plantronics H251N tops with DA60 USB adapters. All Desktops are at least 2.0 GHz P4 with 512MB RAM -...

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:

...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. Right, I wouldn't suspect otherwise. > 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 tremendously IMPROVED their > call quality with IAX2. I wonder what the hell is going on then, that is definitely something strange. > Headsets are Pl...

...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 our main office which has over 100 agent seats. http://astguiclient.sf.net/ There is also Aheeva- http://www.aheeva.com/ for a commercial Asterisk call center solution that offers a ton of functionality for a price. That's about all I know of for full Asterisk call center suites. Let me know if you have any questions about astGUIclient. MATT--- -----Original Message----- From: Erick...

...ot of work as far as I am concerned, for free it is > OK I guess. I think using meetme conference rooms for everything is a > kludgy hack, and the UI is less than nice (if you are into UIs). > > I suggest you continue on your own custom development if you have the > time. Check out Aheeva for inspiration. > > Thanks, > Steve Totaro > > On Fri, Oct 17, 2008 at 1:31 AM, ram <talk2ram at gmail.com> wrote: > > look at Vicidial > > > > ram > > > > On Thu, Oct 16, 2008 at 4:46 PM, yavuz yildirim <yvzyldrm at gmail.com> > wrote: &...

Hi, I was just wondering if anyone knows how predictive dialers detect voicemail and answering machines, and if they could explain to me how that works. Thanks! Brian.

Hello. Second question. Should I be asking this on the dev list (that's not the question by the way). Q. - there are several mentions on the list that asterisk :- "can interoperate with almost all standards-based telephony equipment" "interconnection with digital and analog telephony equipment" "visual message waiting indicator" etc. etc, That seems

Guys, is there any way to detect voice when calling a zap channel? For example, if you want to send out or playback a recorded message, you need to wait for somebody to actually answer the phone before playing starts.. Anyway to detect this?

...orm - Internationalizing Asterisk - Asterisk In The SOHO Environment PEOPLE: - Mark Spencer - Serge Kruppa - Kevin Fleming - Jason Goecke - Matt Riddell - Tim Panton - Dave Troy - David Zimmer - David Duffett - Randy Resnick ASTERISK EXPO: - Digium - Alliance Systems - Aheeva - Mix Networks - Audiocodes - AudioCodes - Kirk Telecom - Xorcom - GES - Axialys - Citel - Ranch Networks OTHER EVENTS: - AstriCon Party - dCAP Testing - Product Demos - Coding, debugging, etc. Please see the AstriCon site for a complete schedule for all th...

what is the general view about the versions of the packages that are used with asterisk. lame asterisk asterisk-addons dahdi libpri i like to say on a version and not upgrade due to my experience with Linux and upgrading screwing up things. When it comes to Asterisk i have only one server build under my belt and I have had issue along the way. What do most people do with the software

A friend of mine runs a small office building, 10-15 tenants. Each have their own company, their own thing, renting space from him. His main PBX is getting dated and his tenants are complaining. I was telling him about Asterisk but his main concern is he doesn't want to have to always be the one to add/remove extensions, or change the IVR hours or whatever. Does anybody know of a free or

Hello, I'm trying to sum up current options for doing small (up to 20 agents) inbound-only CC. I've found: astguiclient, maybe there are some other CC solutions? And on the other side: witch is better to have 20 PC w/ softphones or one T1 channel bank and normal phones with hands-free sets. (whitch set whould you recomend) ? kd, -- Krzysztof Drewicz Affordable 2/4 span E1/T1