similar to: PRI Lines not being answered (No User Responding)

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.

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. 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

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

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

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?

Hello, I have written a small function ('JostD' based upon a recent molecular ecology paper) to calculate genetic distance between populations (columns in my data set). As I have it now I have to tell it which 2 columns to use (X, Y). I would like it to automatically calculate 'JostD' for all combinations of columns, perhaps returning a matrix of distances. Thanks for any help

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

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

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:

Greetings - We've a pair of ISDN BRI that we use for dialtone, fairly happily except for the recent meltdown of one of our Netgear RT338's. We're in the middle of slowly migrating to a VoIP/Asterisk-on-FreeBSD based phone system. I had originally considered just buying a Digium TDM400 card and continuing to use the RT338's to bring out POTS lines from the BRI's, but the

We are in the process of installing a PRI line and we are going to connect it to an Asterisk Box. Verizon called us today to find out some information. I am surprised that they have never heard of Asterisk or Digium. But anyways, they needed some information in order to set up the circuit. Does the TE110P support NI1 or NI2? (I think the answer is both) What is the number of digits

We are connecting a new device to our voice system, and the vendor has a whole list of "supported" network devices and associated parameters for the Telco to adjust... Yes, it's the telcos issue... Anyway, out of all of the swtich types that dahdi can emulate, what is the most complete/tunable solution? The premise for those interested is a PRI connection to a Dialogic card,

--- span_1 = DAHDI/g1 1,1,dial(${span_1}/${EXTEN:0}) --- I can only presume some form of precedence overrides the group configuration in the recent asterisk installs and not for the servers set up earlier. On 26/5/09 4:01 PM, "Kal Feher" <kalman.feher at melbourneit.com.au> wrote: > Ok I've solved the problem. I do not think it was as switchtype issue after > all as

asterisk-users-request@lists.digium.com is believed to have said: >chan_capi does not set the NT-mode. Your cards driver need to do that. >E.g. for Eicon DIVA Server cards, you just set the '-x' option with divactrl >or set NT-mode in the config wizard. >chan_capi does not (need) to know anything about what protocol the card is >doing. CAPI is independent here. Ok.

-----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

My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. I have added

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

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