similar to: Question about STREAM FILE.

Hi all, Now I'm working with a E400P, and I don't now if it's possible to do the following. I want that and AGI script (Perl) recieve a call, and the user introduce the date, the time and the destination phone number (where the temporized AGI must call). Before an AGI script will call to that number in the date and time introduced by user. That's possible, and it's how can I

Hi everybody, Firstly I'm going to describe the scenario where I'm working. I use a E400P with Asterisk CVS-05/22/03-11:14:50, and I'm working with asterisk trow AGI scripts (Perl). The configuration of extension.conf is: exten =>_s,1,Answer exten =>_s,2,AGI,script.agi Inside the AGI script is call Dial application as follows: print "EXEC Dial

Hi everybody, I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae In * one sip.conf register =>usuario1:pass1@<public_ip_2> In * two sip.conf [usuario1] type=friend username=usuario1 secret=pass1 host=<public_ip_1> dtmfmode=inband Logs in * are the followings In * one logs: Sip

Hi All, I was able to track down what I believe is a bug when using AGI services. This bug may crash your system if your extensions.conf script is intensive in using AGI services. Depending on your system's ulimit, * keeps opening files until it reaches the system limit and then stops responding. Function app_agi/launch_script seems to leave an open and unused file. Can someone confirm this?

Some of you have been following our progress on http://farfon.convergence.com.pk as we blundered our way through the development of a low-cost ethernet IP phone that does IAX and augments the client options currently available for the kick-assterisk server. With help from the denizens of #asterisk and kind words of advice from Mr. Spencer and the rest of the gang ... we're proud to have

Hi, I am having a weird issue with samba where once a week approximately at the same time users will lose connectivity, if i run wbinfo -u all users are displayed wbinfo -g all groups are displayed However running getent passwd only shows local-users, no remote users are shown.. To fix the issue I have to change the name of my idmap config and restart samba and winbind and everything works

Hi All, I have a doubt regarding join tables I''m having 2 models 1)Fac 2)Cont and both models have " has and belong to many" relationships so there are 3 tables 1)facs 2)conts 3)conts_facs then i''m fetching the data in controller as @conts=Cont.find(:all]) @cfacs=Fac.all(:joins=>:conts, :select=>"facs.name") but i dont know how to get the

<DIV><FONT face=Arial size=2>hello All,</FONT></DIV> <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV> <DIV><FONT face=Arial size=2>our Asterisk pbx is sitting behind a normal analog hardware pbx, we have to dial 9</FONT></DIV> <DIV><FONT face=Arial size=2>to take an outside call through the hardware pbx, our

Hello, One quick question. Does anyone has experience implementing unified messaging (UM) using Asterisk. Does Asterisk has support for UM ? Thanks, Tarun ___________________________________________________ Medicine meets Marketing; Dr. Swati Weds Jayaram. Rediff Matchmaker strikes another interesting match !! Visit http://matchmaker.rediff.com?2

Hi, I'm interpolating a list of syncronous accumulated precipitation observations collected over a number of raingauge stations sited over land, over a regular lat/lon grid using akima's interp(). Then, I plot and locate geographycally the resulting field with a filled.contour() and a call to map(). Everything is fine but I need to "blank" (in Golden Surfer nomenclature)

Hi, I am a beginner to Asterisk. Can anybody clear my following doubts regarding the configuration needed? 1) What is the ideal system configuratin required?(like processer, RAM, h/d space etc) 2) How many connections it can handle at a time? 3) How many Virtual PBXs it can handle? 4) Whether Postgres or Mysql is best suited? 5) How many IVR's it can handle simultaneously? 6) How many

hi, I have a problem in call time out, An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server. But when i do a Dialout(from both E1s)the calls do not timeout. For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering. even after 20 seconds, call doesn't get timeout pls gv me a solutions.. its really urgent.. Surajee

http://www.voip-info.org/tiki-index.php?page=Asterisk+multi-language By trial and error and a lot of ancient nordic magic (reading the source) I found out that Asterisk does not look for language-specific sound files with the -cc extension, cc being country code. Asterisk looks for files first in a "cc" subdirectory, like "se/vm-login.gsm", then in the default directory.

Hi all and thanks for your time in advance, I can't figure out why summary.lmrob complains when lmrob is used on a zooreg object. If the zooreg object is converted to vector before calling lmrob, no problems appear. Let me clarify this with an example: >library(robustbase) >library(zoo) >dad<-c(801.4625,527.2062,545.2250,608.2313,633.8875,575.9500,797.0500,706.4188,

Hello, I am having some trouble with the Asterisk::AGI perl library. It seems that the AGI->Exec() command is causing me a problem. Here's the line in my AGI code: $AGI->exec('Record',"$vmfile:wav, 30"); I'm trying to record voicemail to the file name stored in $vmfile with a silence timeout of 30. However, this is not being parse by AGI or Asterisk correctly,

Hello all, Can I execute linux command like(ls, mkdir) through the Manager interface? I can't seem to access the manual at digium.com. I keep getting 'Forbidden' error. Looks like they are upgrading or something. CF

Every half year or so, I probably will repost this list, adding and subtracting as the community makes advances (or ignores what isn't required.) >Date: Thu, 9 Oct 2003 04:51:23 -0400 >To: asterisk-users-lists.digium.com >From: John Todd <jtodd@loligo.com> >Subject: Sasquatch, the Loch Ness Monster, UFOs and... > >Mythical Asterisk Creatures, oft-discussed, rarely

Hi all, I''ve installed paperclips but paperclip is not saving the files my model has has_attached_file :attachment,:styles => { :medium => "300x300>", :thumb => "100x100>" } and i have db migration as class AddAttachmentToPolicies < ActiveRecord::Migration def self.up add_column :policies,

ok what softphone i should use to fit windows and linux supporting iax,thanks in advance. _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar – get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of