similar to: Meetme and option c for announcing user count

Hello, Is there any way to force more ringing time on incoming lines? I have heard of the route of answering and then playing a ring tone for a set amount of time, but is there any way to actually have the line ring from the outside for a set amount of time within the dialplan? I thought this would work, but no luck exten => 9999,1,Ringing(5) exten => 9999,2,Answer *CLI> show

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

Hi I have a situation where I want to be able to count how many times a caller goes round a loop of "Please hold...", "please continue to hold". I have found an example on voip-info but I can't get it to work. Not sure if I've got some syntax wrong somewhere? All that happens at the moment, is I hit is the playback of "som-debug" at 9999. Any ideas would

Hi, I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions. Any ideas?? --------------------------------- How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos -------------- next part -------------- An HTML attachment was scrubbed... URL:

Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However, every 4 or 5 times, we get an error back from the provider that says "The number you have dialed.....

Hi, at the moment we have in Avaya Integral PBX with german pri (30 lines). We want to smouthly migrate to an Asterisk server. For this reason: Is it possible to route the external german pri (E1) through Asterisk server to that Avaya PBX? I think at first we need a Digium e1 card 4-Port. But how do we have to configure the routing of the whole PRI? I really would appreciate any sample

Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306

Hi *, I want to integrate the Eicon Diva 4Bri Card to Asterisk. Eicon drivers and capi is installed. I use the latest dev version from eicon compiled and installed for my fedora 2 system. I found the chan_capi for asterisk from www.junghanns.net. Also loaded the patch and applied to the chan_capi source tree. I changed the Makefile to include the capi20.h from eicon:

Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first only rings at my phone and in the second i can see the callers number before i am connected. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib

I have my sip phones going into the context [from-sip] and would like to play an introduction message and then have the caller enter the extension. The message (dir-info was picked just to have something) doesn't play. Maybe I misunderstood the "s" extension. According to what I read it is executed everytime something enters the context. Obviously something was misunderstood. The

I've had slew of problems with my Bell Canada Single Number Reach (SNR) dropping in the past couple of months. Another outage Monday for several hours has me wondering if there's a way to 1. Make a call out of my system via a PSTN back to my SNR line, say every 30 minutes (this I'm sure is easy enough via the call file...however...) 2. Track the outgoing call and match to an

Folks, I'm trying to get Asterisk to load my voicemail configuration from MySQL. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database I restarted Asterisk, but no luck: the voicemail.conf does not get updated. I started with a sample voicemail.conf that I found on the Wiki. Or was it from Voicepulse? I can't remember. For initial testing, I

Hi all, I use the G option in my dials for redirect both parties in the conference. There is a way for auto-include in a conference other parties that first two without using AGI? I try with: [from-internal] exten => 9999,1,Dial(IAX2/DIP02/9999||G(fromiax^9999^1) [fromiax] exten => 9999,1,MeetMe(9999,qdxAa) exten => 9999,2,MeetMe(9999,qdx) exten =>

I'm having problems getting the dial format of Zap/g1/1... working. Here's my zapata.conf: [channels] group=1 context=default ;switchtype=national switchtype=5ess signalling=pri_cpe callerid=asreceived channel => 1-23 and here's my sample extension: exten => 9999,1,Dial(Zap/g1/15102996116) exten => 9999,2,Congestion If I replace "g1" with "1" it calls

On 12 Eki 2012, at 20:00, Jim Grosbach wrote: > > On Oct 12, 2012, at 7:07 AM, Baris Aktemur <baris.aktemur at ozyegin.edu.tr> wrote: > >> Dear Tim, >> >>> >>> The JIT sounds like it does almost exactly what you want. LLVM's JIT >>> isn't a classical lightweight, dynamic one like you'd see for >>> JavaScript or Java.

G'Day All; I am a little unsure on how to get Music On Hold to work. Please critique my extensions.conf. ????? Thanks ; SIP 5001 exten => 5001,1,Dial(SIP/5001) exten => 5001,2,Voicemail(u${EXTEN}) exten => 5001,3,Hangup exten => 5001,102,Voicemail(b${EXTEN}) exten => 5001,103,Hangup Thanks -------------- next part -------------- An HTML attachment was

Hi, we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our old PBX. So now we could migrate to the * server. But, there are two things we can't live with: 1. A call from the outside to the old PBX is missing a leading 0 before the number. Ex: caller has number 0123456 -> * routes to old pbx -> old pbx sees 123456 as caller number. 2. A call made from a SIP

Hi all, I have indexed a huge amount of data with text from several european languages. In the index are values like Georg Friedrich H?ndel. I would like a search phrase like "Georg Friedrich Handel" to find records with the real spelling of H?ndel but it doesn''t seem to work. Can anyone give me an idea of what I need to do to make this happen. A bit lost here and

--l4Siqd0eqV Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Hello, I run the following job. Please, compare the results of summary and table concerning berufl. From similar SPSS/PSPP runs, the result of table is correct. Did I misunderstand anything or is there a bug? What does the difference come from? What does '(other)' mean? What about the strange

Hi, i am running * 1.0.9 with a newer version of the TE405P. Modprobe wct4xxp and ztcfg are OK. zap show channels only shows me the following. my zapata.conf: [pstn] switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown usecallingpres=yes busydetect=no ; not need on pri callprogress=no ; was yes but wiki says experimatley could be produce hangups callwaitingcallerid=yes ; show