Re: [asterisk-users] AGI asterisk high balance

Rizwan Hisham wrote: > Well database really is a bottleneck for me. I am currently trying to do > rating stuff in agi using perl. What im doing is i lookup the rate of > every dialed code for every call from the mysql database using the > longest match technique. The longest match technique cost

Re: [asterisk-users] New generic sounds

SIP wrote: > Tilghman Lesher wrote: >> We're about to do another batch of sounds, and I see by my word count that we >> have some extra time left over. So, suggestions will be entertained for >> additional prompts in English, Spanish, or French. The only rules are: 1) >> the >> prompts have to b

Re: [asterisk-users] Manual Wardialer

Andreas van dem Helge wrote: > Does anyone have a script for manual wardialer for asterisk? not sure > if "wardialer" is the correct term but basically I want to call X > number say 555- through 555-0050 and be able to listen to each > call and when I hang up or press a key it will dial the

Re: [asterisk-users] Digium B410P or Sangoma A502D?

Andres wrote: > We have tested both and they work fine. The Sangoma is much easier to > install as it does not depend on any other driver, you just run > 'setup-sangoma' and follow the instructions. You don't have to fiddle > with the linux kernel or zaptel or chan_misdn. It just works. Plu

Re: [asterisk-users] GoToIfTime problem

Lee Jenkins wrote: > I'm having a problem at a custom site where GotoIfTime doesn't seem to be > working for some reason. I had putty running and logging CLI output and > below > is the call data: > > -- Executing Answer("Zap/3-1", "") in new st

[asterisk-users] GoToIfTime problem

I'm having a problem at a custom site where GotoIfTime doesn't seem to be working for some reason. I had putty running and logging CLI output and below is the call data: -- Executing Answer("Zap/3-1", "") in new stack -- Executing Ringing("Zap/3-1", "") in new stack -- Executing Wait("Zap/3-1"

Re: [asterisk-users] DUDE!!!!! was RE: Dialplan Visualization(Extensions.conf or Dialplan Show)

Brent Davidson wrote: > John Signorello wrote: >> excuse me... >> >> But did you not just post >> >> [asterisk-users] OT Nice IBM 1U Server Gets Along w/Old and New >> Digium Boards Cheap X305 $199 >> >> Did you not provide a link to a COMMERICAL entity? >> >> Wasn't your a post a unsolicited p

Re: [asterisk-users] Drag and Drop transfer application

Steve Edwards wrote: > On Wed, 16 Apr 2008, Bob G wrote: > >> Why the guy asked a question? >> >> From: "Lee Jenkins" >> >> Bob G wrote: >> > >> > Introducing Click-to-Call >> >> I think you're going to get yel

Re: [asterisk-users] Drag and Drop transfer application

Bob G wrote: > Introducing Click-to-Call > > Posted: 16 Apr 2008 9:55 AM PDT > > The 1EZphone browser softphone has created so much buzz in the media > that a lot of individual users and companies who have a web-presence; > Websites, Online Advertising, Blogs, Customer

Re: [asterisk-users] Drag and Drop transfer application

Lee Jenkins wrote: > Al lists wrote: >> Hi list, >> Any good drag and drop transfer call application for windows based >> systems you can advise ? >> Something like HUD perhaps? >> >> > > Yes. > > Maestro Control Panel (I authored this one) >

Re: [asterisk-users] Drag and Drop transfer application

Al lists wrote: > Hi list, > Any good drag and drop transfer call application for windows based > systems you can advise ? > Something like HUD perhaps? > > Yes. Maestro Control Panel (I authored this one) http://www.datatrakpos.com/pos/datatalk/maestro.aspx. There is also the nice flash base

Re: [asterisk-users] PBX Console

an see it by going to: Admin > Users > Edit Users and selecting my record :) ==> Lee Jenkins. Are you having trouble entering the software? If so, it's a good chance you may be running Vista or a nicely locked down version of XP. The original installer saved Maestro's (fir

Re: [asterisk-users] PBX Console

Darryl Dunkin wrote: > FOP works for us, no need for X: > http://www.asternic.org > > If you need to avoid using a mouse, you can use the Polycom attendant > console instead: > http://www.polycom.com/usa/en/products/voice/desktop/soundpoint_ip/sound > point_ip_attendant_console.html > We recentl

Re: [asterisk-users] app_swift v1.6.1 released for Asterisk 1.6

Kai-Uwe Jensen wrote: > An app to invoke the Cepstral text-to-speech engine. > > On Tue, Apr 15, 2008 at 11:46 AM, Zoa <[EMAIL PROTECTED] > > wrote: > > > What is app_swift ? > > Zoa I've written an AGI wrapper for it as well, in case you don't want to re-co

Re: [asterisk-users] Phantom Rings

Brent Davidson wrote: > I'm having a major problem at one of my branch offices with "Phantom > Rings" on their asterisk-based phone system. The system was originally > built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC > card. The upgrade severely increased the frequency o

Re: [asterisk-users] Web page to show online extensions?

Vincent wrote: > On Thu, 3 Apr 2008 10:51:15 -0430, Earl Terwilliger <[EMAIL PROTECTED]> > wrote: >> http://www.micpc.com/eventmonitor/ > > Thanks guys. I was also thinking of stand-alone apps like Jabber or > something. The call is simply to know if an extension is on- or > offline. > > N

Re: [asterisk-users] AsterPas ObjectPascal Based FastAGI Server goes Open Source

Lee Jenkins wrote: > Announcement: > > We are pleased to announce that we have released AsterPas FastAGI ObjectPascal > Script Server for Asterisk PBX under license. > Oops. That should be LGPL license ;) -- Warm Regards, Lee "Everything I needed to learn in lif

[asterisk-users] AsterPas ObjectPascal Based FastAGI Server goes Open Source

Announcement: We are pleased to announce that we have released AsterPas FastAGI ObjectPascal Script Server for Asterisk PBX under license. What is AsterPas? AsterPas is a FastAGI server which allows real-time scripting of Asterisk PBX call flow using ObjectPascal based scripting. AsterPas

Re: [asterisk-users] Limit calls when using autodial

Tong wrote: > That might work. > > I'll give that a shot. > > Doug Lytle <[EMAIL PROTECTED]> wrote: >> Tong wrote: >>> Is there a way to limit outbound calls when feeding files to the outgoing >>> directory in asterisk? I several thousand files i need to feed asterisk, >>> hoping to cop

Re: [asterisk-users] MeetMe option b

Jerry Geis wrote: > Jerry Geis wrote: >> I am running asterisk 1.4.18 trying to use MeetMe and option b. >> >> I am getting permissions denied failed to execute conf-background.agi >> on the CLI >> >> lrwxrwxrwx 1 root root 37 Mar 17 10:11 conf-background.agi -> >> /home/silentm/bin/conf-bac

Re: [asterisk-users] Telemarketer Torture....

James Finstrom wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Anyone have the telemarketer torture prompts? I would seriously like > to revive this. > > - -- I wrote one a while back that uses Cepstral TTS, but the mechanics are simple. When a telemarketer calls, I say "hmmm,

Re: [asterisk-users] Callerid Error- Causing All Zap Channels Busy

John Meksavan wrote: > Asterisk Users, > > I am running Asterisk-1.4.11 on a Debian "Etch" system. On an > occasion, when customer calls into my Asterisk Box, I get this error > messagefrom Asterisk "CallerID returned with error on channel Zap/3-1" , > causing all my zap channels to be busy.

Re: [asterisk-users] Logs for Call generated by Manager API

Mark Hamilton wrote: > I don't think the link that Lee gave works. > Oh boy, you were talking about the link to download the software and I completely misunderstood. My mistake, the link is fixed to download the software. Here's the direct link: http://www.datatrakpos.com/pos/datatalk/downlo

Re: [asterisk-users] Logs for Call generated by Manager API

Mark Hamilton wrote: > I don't think the link that Lee gave works. > > > Also, I wrote a Windows based utility for viewing AMI packets and testing > AMI > commands. It's Freeware: > > http://www.voip-info.org/wiki-Asterisk+GUI#OperatorManagerInterfaces > Look for Manager API Test Utility > >

Re: [asterisk-users] Logs for Call generated by Manager API

[EMAIL PROTECTED] wrote: > I am generating an outbound call through the Manager API and bridging it to > an internal Extension, my problem is I am not able to find the logs for the > call generated by the Manger API, Since on the same Asterisk server there are > many users connected and I am rec

Re: [asterisk-users] Logs for Call generated by Manager API

[EMAIL PROTECTED] wrote: > Is there way to get the logs of the call generated by Manager API, or is > there some other way to achieve same scenario so that I can get the status of > the call generated by me. > > > Actually I have a scenario where I have to call customers and play a message, >

Re: [asterisk-users] Asterisk monitor() zap channel problem

Raul Alarcon wrote: > im trying to use monitor() aplication with "b" option, to start the > recordigin just once the conversation has actuallly begun. > > It works fine with a sip extensión, but when i use a zap channel, it > records all the channel bridging, including the ringing sounds... > >

Re: [asterisk-users] Attended transfers through a GUI

Chris Bagnall wrote: > Greetings list, > > I've been playing around this afternoon with Flash Operator Panel, trying to > get it to do attended transfers. I am running the latest version. > > Has anyone managed to get this working reliably, and if so, would you mind > sharing how you did it pleas

Re: [asterisk-users] Converence/Meetme with Manager API

Mitchell Jackson wrote: > Hello! I am having problems figuring out how to do something, and any > help would be much appreciated. > > I would like to use the manager API to take an existing call on a > specific SIP extension, dial and conference in a third party. > > From what I can tell, the

Re: [asterisk-users] Variable setting in AMI Originate

Anthony Messina wrote: > On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote: >> Anthony Messina wrote: >>> Working with asterisk 1.4; using the AMI Originate command, it is >>> possible to do something like: >>> >>> Variable: CDR(accountcode)1234

Re: [asterisk-users] Variable setting in AMI Originate

Anthony Messina wrote: > Working with asterisk 1.4; using the AMI Originate command, it is possible to > do something like: > > Variable: CDR(accountcode)123456 > > Or must the variable names be "var[n]" where n is a number? > > I'd like to set the accountcode for a Local channel that originate

Re: [asterisk-users] Asterisk Manager and Visual Basic

Bill Andersen wrote: > Has anyone tried to used VB6 to communicate with the Asterisk Manager? > > If so, would you be willing to share some basic code showing your > approach to getting connected and parsing results? > > I've got a Telnet control that is allowing me to connect, authenticate > and

Re: [asterisk-users] Monitor Asterisk using C

Soumya Kat wrote: > Hi, > > I have installed Asterisk 1.4 along with net-SNMP 5.4.1 in my Fedora 8 > system. Asterisk works fine for me and I can log into Asterisk-GUI and > monitor asterisk. > > What I would like to know is how to get information such as SIP users, > number of SIP connections

Re: [asterisk-users] Higher level API on top of AMI and AGI (was Re: "Real" API for Perl?)

Moises Silva wrote: >>> Asterisk 1.6 includes a new feature that allows using AMI as a transport >>> for AGI commands, there abstraction becomes even more important. >>> For Asterisk-Java I am currently adding support for that in a way that >>> allows the developer to run the same "AGI" code either

Re: [asterisk-users] Higher level API on top of AMI and AGI (was Re: "Real" API for Perl?)

Stefan Reuter wrote: > Lee Jenkins wrote: >> I thought that the OP was asking for something to perl what Asterisk-Java >> does >> for java coders. I would definitely consider Asterisk-Java to be a >> framework, >> though not so much with PasAGI which i

Re: [asterisk-users] "Real" API for Perl?

Alex Balashov wrote: > Well, no, there really aren't any prebuilt high-level frameworks for > approaching Asterisk through the Manager API or AGI. Instead, there are > just AGI bindings that allow you to integrate dial plan logic with > outboard code. > I thought that the OP was asking for so

Re: [asterisk-users] QueueMember event/LastCall Variable - Format?

Lee Jenkins wrote: > Jared Smith wrote: >> On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote: >>> What format is the LastCall variable of QueueMember event? I'm looking at: >>> 1201897536 for instance. >> Unix epoch format, or number of seconds

Re: [asterisk-users] "Real" API for Perl?

Alex Balashov wrote: > Ken D'Ambrosio wrote: > >> Hi, all. I've used the perl/AGI interface, and... well, I found it kind >> of hokey. Granted, this was in 1.2 days -- perhaps things have changed. >> Regardless, I guess I have two questions: >> 1) Has the Perl/AGI "binding" improved since then?

Re: [asterisk-users] QueueMember event/LastCall Variable - Format?

Jared Smith wrote: > On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote: >> What format is the LastCall variable of QueueMember event? I'm looking at: >> 1201897536 for instance. > > Unix epoch format, or number of seconds since Jan 1, 1970 UTC, as I > recall.

[asterisk-users] QueueMember event/LastCall Variable - Format?

Hi all, What format is the LastCall variable of QueueMember event? I'm looking at: 1201897536 for instance. Thanks ! -- Warm Regards, Lee "Everything I needed to learn in life, I learned selling encyclopedias door to door." ___ -- Bandwidth and

Re: [asterisk-users] [AGI 1.4] C sample?

Vincent wrote: > On Sun, 27 Jan 2008 09:09:59 -0500, Lee Jenkins <[EMAIL PROTECTED]> > wrote: >> Sorry, I don't have a sample for you as I write mostly in Freepascal/Lazarus >> these days and use my own library for AGI/FastAGI. That said, did you try >> saving

Re: [asterisk-users] [AGI 1.4] C sample?

Vincent wrote: > Hello > > I'm pretty much a newbie when it comes to C, but I have to use > this language to write a couple of AGI proggies because I need them to > be statically compiled. > > Strangely enough, Google didn't return much when looking for the > "Hello, world!" of AGI in C. >

Re: [asterisk-users] New Polycom Provisioning Tool Released with BugFix

Lee Jenkins wrote: > Doug Lytle wrote: >> Michael Munger wrote: >>> Polycom Provisioning Tool Updated. >>> >>> >> >> Looks like a Windows only tool. Shame it doesn't work under Wine. >> >> Doug >> >> > >

Re: [asterisk-users] New Polycom Provisioning Tool Released with BugFix

Doug Lytle wrote: > Michael Munger wrote: >> Polycom Provisioning Tool Updated. >> >> > > > Looks like a Windows only tool. Shame it doesn't work under Wine. > > Doug > > Looks like it was written with VB.net. Not sure where Mono is as far as VB.net goes, but if I'm not mistaken, once i

[asterisk-users] AMIProxyPal - AMI Proxy Project

After having misunderstood some key elements of AstManProxy, I started to write my own proxy server for Asterisk AMI. I was under the impression that it required a mysql database to cache its data for some reason. (Is there another AMI proxy that uses a mysql database?) At any rate, I had wr

Re: [asterisk-users] ProxyPal for AMI Proxy Development

Julian Lyndon-Smith wrote: > Lee Jenkins wrote: >> Julian Lyndon-Smith wrote: >>> astmanproxy does this already, I think .. >>> >>> Julian. >>> >> Of course ;) AstManProxy is a great product from what I had read up on it. >> >&g

Re: [asterisk-users] ProxyPal for AMI Proxy Development

Julian Lyndon-Smith wrote: > astmanproxy does this already, I think .. > > Julian. > Of course ;) AstManProxy is a great product from what I had read up on it. One thing is that it requires (if I'm not mistaken) an mysql installation which is too heavy of a dependency for some applications th

[asterisk-users] ProxyPal for AMI Proxy Development

Hi all, I'm writing a real-time (not RealTime) proxy server for the AMI interface. Although I'll be using it for some commercial products, the proxy software itself will be released under GPL. I was wondering if there would be any interest in testing it from the community? I don't have ac

Re: [asterisk-users] zaptel programming

Philipp Kempgen wrote: > Bhrugu Mehta wrote: > >> I am new to zaptel programming. >> can anybody help me how to start this. or any ref. site or matirial >> availabel. >> i want to use c lang. for this. > > > Some tutorials: > http://www.google.com/search?q=learn+c+in+21+days > When done ask

Re: [asterisk-users] [Zaptel] Why no port to Windos?

Vincent wrote: > On Fri, 14 Dec 2007 10:51:10 -0500, Lee Jenkins <[EMAIL PROTECTED]> > wrote: >> I have to reboot my desktop xp box daily for it to run well. > > I haven't rebooted my XPSP2 in months, and I let it run 24/7, with a > bunch of apps open at all times

Re: [asterisk-users] Text-To-Speech synthesizer--help required

srinivas Antarvedi wrote: > Hello users, > > Actually i wanted to implement Text-To-Speech engine > from cepstral voice using swift application > > i tried the documentation of doing this and i was unsuccessful > at doing this work with asterisk > > can anybody please help me out finding the sol

Re: [asterisk-users] Asterisk.NET API --help required

srinivas Antarvedi wrote: > Hello all, > > Here is the requirement from my side > to use Asterisk.NET API to generate > an automated call (outgoing) from asterisk > and then link to one of the extensions which > plays a sound file for the callee. > > For this i have worked out in the follwin

Re: [asterisk-users] [Zaptel] Why no port to Windos?

Doug wrote: > At 19:55 12/13/2007, Vincent wrote: > >Hello > > > >I was wondering why there doesn't seem to a Windows version of Zaptel, > >making the Digium and its clones unavailable for a Windows PBX. > > > >Is the Zaptel/Zapata combo too *nix-centric? > > > >Thanks. > > Windows is a ha

Re: [asterisk-users] Affordable SIP Trunk for Home PBX ?

Steve Edwards wrote: > On Sat, 13 Oct 2007, Lee Jenkins wrote: > >> I have been using axVoice.com for some about 9 month to a year now and >> their service is pretty damn good. For home users they have unlimited >> plan for around 22.00-24.00 U.S. per month. > &

Re: [asterisk-users] Affordable SIP Trunk for Home PBX ?

D4rk F1ber wrote: > So I have my asterisk box up and working internally at home and all is > good so far. The next thing I wanted to do was make and recieve calls > to regular land lines now. > > I don't have a POTS line and was looking for probably a SIP trunk. > > I have seen mentions of Skype

Re: [asterisk-users] How to use an Application from inside an Application?

Pirlouwi wrote: > Hello, > I wonder if there is a way to build my own asterisk application (let us > say apps/app_myappl.c), > and to launch other existing applications from it (for example, doing an > apps/app_dial.c, or others). > > Could someone highlight me on that? > thx > Pirlouwi. > Eve

Re: [asterisk-users] Using PHP to reload extensions

Mojo with Horan & Company, LLC wrote: > No, because then asterisk would be presented three arguments: '-rx', > 'extensions', and 'reload' -- as 'extensions' is not a command by > itself, and the 'reload' appears superfluous to asterisk, this would not > work as desired. > > Asterisk needs to be

Re: [asterisk-users] Using PHP to reload extensions

Michael Munger wrote: > I am trying to use PHP to reload the extensions in an Asterisk > installation. I keep getting this error: > > Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) > when I run the script by visiting the URL; however, if I run the script > from the com

Re: [asterisk-users] extensions.conf vs. AEL

Yehavi Bourvine +972-8-9489444 wrote: > Hello, > > I see that most people are using the extensions.conf syntax (most of the > examples and questions here use that syntax). recently I've translated all my > dial plan to AEL syntax and I find it much easier, especially when you need > IFs. > >

Re: [asterisk-users] Stable-Stable Asterisk

Steve Totaro wrote: > David Gomillion wrote: >> On 8/23/07, *Ed Pastore* <[EMAIL PROTECTED] >> > wrote: >> >> Hi, folks. >> >> I've been on the Asterisk Announce list for a while now, and it seems >> to me that the release versions of Asterisk are a bit bleedi

Re: [asterisk-users] Which GUI for ACD edition ?

Olivier wrote: > Hello, > > I want to safely delegate ACD edition to a system administrator who has > no knowledge of Linux nor Asterisk. > More precisely, I want him to be able to edit and change menus such as : > "Type 1 for management; 2 for support; 3 for sales department". > > I could teach

Re: [asterisk-users] RAW asterisk!

Doug wrote: > At 19:35 8/17/2007, Lee Jenkins wrote: > >Bill Andersen wrote: > >> I'm a network admin that maintains 3 commercial Asterisk > >> servers for my employer. > >> > >> I am wanting to move away from the "pre-packaged"

Re: [asterisk-users] RAW asterisk!

Bill Andersen wrote: > I'm a network admin that maintains 3 commercial Asterisk > servers for my employer. > > I am wanting to move away from the "pre-packaged" commercial PBXs > to a more "pure" asterisk setup. The systems I have utilize a nice > web GUI to make changes, but it really limits wha

Re: [asterisk-users] Dialplan / AGI autoanswer question

Matthew Harrell wrote: > Hi. I've got a working dial plan on my home system but there are problems > with it and I was hoping someone more comfortable with dial plans might be > able to help. In a nutshell here's what I'm currently doing on an incoming > outside phone call > > [default] >

Re: [asterisk-users] CallerID Error causes problems for Polycom phones

Anselm Martin Hoffmeister wrote: > Am Mittwoch, den 15.08.2007, 10:14 -0400 schrieb Lee Jenkins: >> Hi everyone, >> >> I have been dealing with a certain issue with a particular customer site >> for months now. The problem occurs when there is an error with caller >

[asterisk-users] CallerID Error causes problems for Polycom phones

Hi everyone, I have been dealing with a certain issue with a particular customer site for months now. The problem occurs when there is an error with caller id as shown in the following: WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error on channel 'Zap/3-1' When this happe

Re: [asterisk-users] asterisk 1.2.14 with GUI

satish patel wrote: > dear all > > is there any GUI application with support asterisk 1.2 > version i am useing 1.2 and i have fine more about GUI base > configuration but i didnt got any GUI package for asterisk 1.2 > > If you're a windows user, you can also check out DialplanP

Re: [asterisk-users] global variables and updates

Watkins, Bradley wrote: > The contents of this e-mail are intended for the named addressee only. It > contains information that may be confidential. Unless you are the named > addressee or an authorized designee, you may not copy or use it, or disclose > it to anyone else. If you received it in

Re: [asterisk-users] global variables and updates

Julian Lyndon-Smith wrote: > Sorry if this appears twice - I originally sent it nearly 18 hours ago > and never saw it .. > > I have a need to have a unique integer number that can be used by a > dynamic meetme room (I am wanting to redirect a call into a meeting > room, and need a unique number

Re: [asterisk-users] Music on Hold and Announcements

OCOSA ListAcct wrote: > Does anyone know how to have an ad or announcement playing but in the > background play a MP3 file? > > I think this would be done with the "s" extension and background > application but not sure how? Any help would be appreciated!! > We just used Audacity and blended a

Re: [asterisk-users] Micros-Fidelio - billing in hotel

Tomislav Parcina wrote: > There is hotel application weary popular in Croatia - Micros-Fidelio. > Now I need to connect Asterisk with this application for purpose of > billing. Thing is that hotel would like to give customer one bill for > every service that he used while he was in hotel. > > H

Re: [asterisk-users] Micros-Fidelio - billing in hotel

Chris Mason (Lists) wrote: > Lee Jenkins wrote: > >> I'd say that Micro is the "MS" of Restaurant POS. We replace their >> systems regularly ;) > I'm curious what with? > www.datatrakpos.com Notice that I didn't say en masse but yes, we do

Re: [asterisk-users] Micros-Fidelio - billing in hotel

Tomislav Parcina wrote: > There is hotel application weary popular in Croatia - Micros-Fidelio. > Now I need to connect Asterisk with this application for purpose of > billing. Thing is that hotel would like to give customer one bill for > every service that he used while he was in hotel. > > H

Re: [asterisk-users] Queue property

equis software wrote: > Hi! > I want to have this behabior in my queue. > When a call come in, if there are unavailable agents or and busy agents, > the queue reject the call. > > Thanks ! > > Also check out joinempty=strict ...it's in the same article: http://www.voip-info.org/wiki/index.p

Re: [asterisk-users] Queue property

equis software wrote: > Hi! > I want to have this behabior in my queue. > When a call come in, if there are unavailable agents or and busy agents, > the queue reject the call. > > Thanks ! > > I think you need checkout: "Introduced right after the v1.0 release If you wish to remove callers fr

Re: [asterisk-users] Queue Status

Arun Kumar wrote: > Hi > > I already tried asterisk manager but Im not able to get status for each > queue member. > > thanks > That must be a problem with your configuration. I get QueueMemberStatus on my AMI interface (1.2): Event: QueueMemberStatus Privilege: agent,all Queue: support L

Re: [asterisk-users] Polycom 301 - Problem with AMI Originated Calls

Lee Jenkins wrote: > Hi all, > > I'm having an odd problem with my polycom 301. I am initiating a call > to it with AMI Originate() function: > > Action: Originate > Channel: local/[EMAIL PROTECTED] > Context: to_meetme > Exten: s > Priority: 1 > Variable

Re: [asterisk-users] Sip Providers

Alex Roston wrote: > Hi Everyone, > > I'm planning my first asterisk box, and I'd like to know what SIP > providers everyone likes. Voipjet? Gizmo? Somebody else? > > Thanks, > > Alex > I've been using www.axVoice.com for about 9 months now with great results. Quality is good, but communica

Re: [asterisk-users] QueueMemberStatus

Lee Jenkins wrote: > I've been poking for the definition of QueueMemberStatus and all the > source file indicates is that it is a integer member of the "member" > structure. > > Anyone know where I can find the CONSTANTS definitions? > OK, I didn't know t

[asterisk-users] QueueMemberStatus

I've been poking for the definition of QueueMemberStatus and all the source file indicates is that it is a integer member of the "member" structure. Anyone know where I can find the CONSTANTS definitions? -- Warm Regards, Lee ___ --Bandwidth a

Re: [asterisk-users] Call transfer in asterisk

satish patel wrote: > dear all > > I am new in asterisk and i have now setup asterik for > 40 phone now i want to configure call transfer between phone so how it > is possible and what configuration part in asterisk will perfomed for > this task give me suggestion for my solut

Re: [asterisk-users] Polycom echo problem

Zeeshan Zakaria wrote: > I had the same situation and I had to replace my T1 card with the one > with hardware echo canceller. All other solutions were failed. May be > you need to do the same if you're on a PRI or using PSTN lines. If > you're on a pure VoIP network, then its the phones. > > O

Re: [asterisk-users] Call transfer feature

satish patel wrote: > Dear ALL > >I want to transfer call from one phone 2 another > phone so this is asterisk feature or SIP Phone feature or endpoint > feature how can i transfer phone call from to another phone > > > Rgd > > Satish patel > Check out this page: http:/

Re: [asterisk-users] problem with one way audio

Jason Backshall wrote: >> Do you have CallProgress=yes in your zapata.conf? This one just bit me >> in the arse this morning. I set it to no and one-way audio went away. > > Have heard of issues similar to this - and whilst disabling callprogress may > make that symptom disappear, it probably s

[asterisk-users] Polycom 301 - Problem with AMI Originated Calls

Hi all, I'm having an odd problem with my polycom 301. I am initiating a call to it with AMI Originate() function: Action: Originate Channel: local/[EMAIL PROTECTED] Context: to_meetme Exten: s Priority: 1 Variable: dropped_conf=111 The "to_meetme" context is very simple: [to_meetme] exten=>

Re: [asterisk-users] problem with one way audio

Don Briggs wrote: > I have a company with asterisk 1.2.19 and polycom 501 phones. I get one way > audio. A caller from the pstn world hits the tdm400 card, This rings two > phones in a ring group. My client answers the phone, the calling party is > told the customer here her but she can not h

Re: [asterisk-users] Phantom Calls

sage- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Lee > Jenkins > Sent: Monday, June 18, 2007 1:56 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Phantom Calls > > Stephen Bosch wrote: >> Lee Jenk

Re: [asterisk-users] Phantom Calls

Dave Miller wrote: > Lee Jenkins wrote on 6/19/07 9:56 AM: >> Vadim Berezniker wrote: >>> Enable verbose logging for the asterisk log >>> Set verbose level to 4 >>> >>> Review the log file for anything that looks like a phantom call. >>> Ther

Re: [asterisk-users] PhpAgi call generation

Nitesh Divecha wrote: > Is there any info on how to create .call files with some examples? And > where to place this file? And how to initiate it..? > > Thanks man... > > Cheers, > Nitesh > > > > Christopher Robinson wrote: >> That should be pretty easy to do with a .call file. The context t

Re: [asterisk-users] Play dial tone withou answer

David Boyd wrote: > Two points, > > > > first (I believe from many previous threads, and viewing source code > ) you must answer a call to place audio on the channel. > > second, depending on the type of access being used by the originator of > the call, the carrier will not pass audio on the c

Re: [asterisk-users] Phantom Calls

sage- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Lee > Jenkins > Sent: Monday, June 18, 2007 1:56 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Phantom Calls > > Stephen Bosch wrote: >> Lee Jenk

Re: [asterisk-users] Phantom Calls

Matt wrote: > I too have seen what Rob is saying.. on a Sangoma card. It was an easy > fix in the config, but I don't remember what it was.. but basically it > was stray voltage. > > On 6/18/07, * Rob Schall* <[EMAIL PROTECTED] > > wrote: > > We were having phan

Re: [asterisk-users] Phantom Calls

Stephen Bosch wrote: > Lee Jenkins wrote: >> I have a client that is having problems with phantom calls. I have not >> been able to see it happen myself, but they say when it happens, the >> display on the phone (polycom 301's) says "Device" is calling, but w

[asterisk-users] Phantom Calls

Hi all, I have a client that is having problems with phantom calls. I have not been able to see it happen myself, but they say when it happens, the display on the phone (polycom 301's) says "Device" is calling, but when they answer the phone, there is only silence and then they hang back up

Re: [asterisk-users] combining AGI with dialplans

[EMAIL PROTECTED] wrote: > On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out): > >> Can't comment on this one, as I never use AGI to dial. >> My AGIs just set the context, extension and priority, >> and exit to the dialplan to do any dialling. > > (http://article.gmane.org/g

Re: [asterisk-users] Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event

Lee Jenkins wrote: Hi all, My company has pretty much standardized on Polycom phones and I am in the beginning phase of writing a GUI for administering/managing polycom provisioning at multiple sites which we intend to release as OS. I've started studying the docs and I'm havi

Re: [asterisk-users] GotoIf Dialplan inquiry

Doug Lytle wrote: Steve Finkelstein wrote: Hi all, I have the following in my extensions.conf: I use the mysql addon and create a subroutine that checks for black listed numbers. I then call it at each entry point (For faxes as well): ; ** ; Auto attendant ; *

Re: [asterisk-users] Introduction to AGI programming

Kyle Sexton wrote: I wrote an introduction to AGI programming paper as an exercise to learn more about the process involved. You can find a copy of it here . I welcome any comments or corrections to improve upon it. As I said, it was mainly done to force myself to

Re: [asterisk-users] agi with java?

programming. On the other side, I have no doubt that with an application server and FastAGI you can get quite a lot of bang for the buck. :) l. On Fri, 08 Jun 2007 18:07:50 +0200, Lee Jenkins <[EMAIL PROTECTED]> wrote: We have found that generally speaking, running the FastAGI server o

Re: [asterisk-users] Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event

Kenneth Padgett wrote: My company has pretty much standardized on Polycom phones and I am in the beginning phase of writing a GUI for administering/managing polycom provisioning at multiple sites which we intend to release as OS. I've I'd love to be notified when you release the Polycom admin

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