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 costs atleast 2-3 mysql queries for every call untill the dialed code is matched out of 14000 dialcodes. I dont know how to calculate the exact delay due to execution of agi, but on the asterisk cli whenever that agi executes, there is a visual delay of about half a sec to move from the agi extension to the next extension (can anybody tell me how to calculate the delay). Now im planning to use the manager api for constant connectivity to mysql and to enhance the longest match technique. Can anybody help me with this? Is it a good idea to ue manager api for lookingup the rate of the live call? I'm a FirebirdSQL guy myself, but I find that if I implement connection pooling, apps like the one you've described are faster since connection setup and tear down are always expensive. -- Warm Regards, Lee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 be generic to Asterisk. No Welcome to so-and-so's business unless the business is fake and the prompt is funny. 2) The prompt may not be profane. Our professional speakers do have a sense of humor, but there are some things they just will not say. I'll open it to the floor now, with the caveat that since Digium is paying for the recording session, it maintains final editorial approval over which sounds are selected. How about some prepaid balance-related ones that aren't calling-card-specific. Things like: Your balance is too low to connect this call. Please add additional funds to your account. Your account balance is... and one for the permissions set: ...from the account... (to go along with the Calls to the number you have dialed are not permitted) Also, I'm not sure if there any as of yet, but maybe some kind of credit card processing statement? Please hold while we process your transaction..., Your transaction has been approved.. I'll be writing an AGI executable in the next few months to interop with PCCharge and Mercury Payments so it would be nice to see some credit card snippets added too. -- Warm Regards, Lee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 next number for me. I guess sort of like scanning an exchange but I want to be on the line and if possible complete / talk on certain calls. I think this is more of a power dialer rather than a predictive dialer, which uses a predictive algorithm to pace calling based on current and history data. I think vicidial is capable of both types of dialing I think. -- Warm Regards, Lee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. Plus its more modular. You can chose 2/4/6 ports to buy and if you need more just add remoras up to 24 ports. The Digium card is fixed to 4 ports, period. Having said that, make sure you stick with the version that has hardware echo cancel and not even try the other one. We made the mistake of buying the first time without echo cancel expecting to test the 'software echo cancel'. But there is no such thing as 'software echo cancel' on this card. I do not even understand why Sangoma would make a version without the hardware echo cancel. You get some degree of echo on practically every call. Andres. I have a couple of installs using the A200 analog card with FXO modules and the Octware echo cancel software works like a charm. These are 2-4 POTS line installs. -- Warm Regards, Lee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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, 0) in new stack -- Executing SetMusicOnHold(Zap/3-1, default) in new stack -- Executing Goto(Zap/3-1, check_time|s|1) in new stack -- Goto (check_time,s,1) -- Executing GotoIf(Zap/3-1, 0?set_no_callerid|s|1) in new stack -- Executing NoOp(Zap/3-1, CallerID: 443866 Cell Phone MD) in new stack -- Executing GotoIfTime(Zap/3-1, 08:30-17:00|mon-fri|*|*|?daytime_ivr|s|1) in new stack -- Executing Goto(Zap/3-1, after_hours|s|1) in new stack -- Goto (after_hours,s,1) This call came in at about 3:10 PM EDT today (Thursday). I did a date command at the linux prompt and the date and time of the computer is set correctly. Now, I have had problem with this particular computer in that the date/time gets changed somehow, although I'm not sure exactly how. I've changed it back several times using the commands (copied from command line history): # date -s 23 APR 2008 1:42:00 # hwclock --utc --systohc I'm still quite the linux noob, so it could be something I'm doing wrong although it seems doubtful since everything was working until recently. Maybe the computer's clock battery is screwed? Thank you, -- Warm Regards, Lee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 stack -- Executing Ringing(Zap/3-1, ) in new stack -- Executing Wait(Zap/3-1, 0) in new stack -- Executing SetMusicOnHold(Zap/3-1, default) in new stack -- Executing Goto(Zap/3-1, check_time|s|1) in new stack -- Goto (check_time,s,1) -- Executing GotoIf(Zap/3-1, 0?set_no_callerid|s|1) in new stack -- Executing NoOp(Zap/3-1, CallerID: 443866 Cell Phone MD) in new stack -- Executing GotoIfTime(Zap/3-1, 08:30-17:00|mon-fri|*|*|?daytime_ivr|s|1) in new stack -- Executing Goto(Zap/3-1, after_hours|s|1) in new stack -- Goto (after_hours,s,1) Never mind, the problem turned out to be between the back of the chair and the keyboard. Sorry for the false alarm. -- Warm Regards, Lee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 post, that is, not in response to a question??? There seems to be two standards here. The fact that you do not work for them is immaterial. If your argument is no commercial reference at all, then how do you explain your post? I felt like the cogoblue stuff is out of place and off topic, so I agree that it should not be arbitrarily posted as a solution to any question that it might seem to somehow solve. However, I do not take exception to the IBM server post, even though strict adherence to the rules would probably make it illegal as well. I think most people agree (or a least don't mind too much) if a commercial product is offered as a possible solution to an OP's query, assuming it is in fact, within context. I'll leave it to others on the list to decide if John Signorello's post was appropriate or not, given the context of the OP's original query, but if someone posts a query directly related to a product I have to offer, I fully intend to let them know about it in as least intrusive manner as I can. Assuming a person's product is directly within context, not offering it as possible solution could be a disservice to the OP and list in general. Just a thought... -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 based Flash Operator Panel http://www.datatrakpos.com/pos/datatalk/maestro.aspx There a couple of other ones out there too that I thought were nice, but can't remember the names. You should be able to find them by gooling for Asterisk Control Panel or such query. -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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) http://www.datatrakpos.com/pos/datatalk/maestro.aspx. There is also the nice flash based Flash Operator Panel http://www.datatrakpos.com/pos/datatalk/maestro.aspx Oops. Sorry, for FOP that is: http://www.asternic.org/ NoteToSelf note=Stop replaying to email while on the phone/ -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
Bob G wrote: Introducing Click-to-Call http://1ezphone.com/ 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 support etc have asked for a Click-to-Call service. I think you're going to get yelled at ;) -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 yelled at ;) 1) You hijacked the thread. 2) You top-posted. 3) It's a non-commercial list -- RTFMLIBP (Read the Mailing List Instructions Before Posting). 4) You just used a run-on sentence. ;) -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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] mailto:[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-compile to support. http://www.voip-info.org/wiki/view/DTSwift+Cepstral+AGI+Wrapper -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 recently released Maestro Control Panel (beta): http://www.datatrakpos.com/pos/datatalk/maestro.aspx Its mouse driven, but easy to use. We have a few clients using it for their secretaries with good success. You can minimize it to the system tray and it'll popup when flagged numbers come in or click on it to do things like get so and so on the phone for me type of functions. It uses the manager api for its functionality so its pretty flexible. We're also working on a cross platform (Win/Linux) sister product designed to run on small touch screens systems. The idea is that it will run on a small embedded linux box (maybe fastened to the underneath of the desk) using a small 8 touch screen. Nothing to show yet unfortunately. You can check back into the message board every so often for news of it when its released. -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX Console
Guilherme Loch Waltrick Góes wrote: What is the default username/password. In the Maestro forum's it only says it's hardcoded, but doesn't say the actual username/password. Guilherme, The username is leebo and the password is 123. You can 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 (firebirdsql) database file to the (\program files\Maestro Control Panel) directory which was a mistake on my part. Windows may be refusing to let you connect to the database because of that. I have changed the installer to save these files to CSIDL_COMMON_DOCUMENTS (Users\All Users) folder so there is no longer any problems accessing the database, assuming you have that problem. Either way, I recommend you download it again directly from here: http://www.datatrakpos.com/pos/datatalk/downloads/maestro_setup.zip That's the same link on the webpage, if you don't feel like navigating back. Please post any further support questions to the message board if its not too much trouble: http://www.leebo.dreamhosters.com/pbxbb/ Thanks for downloading and sorry about the inconvenience. -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and have made several calls to Verizon with no solution to the problem. I know the telco switch that is feeding my analog lines is an old switch and can't even do CallerID with 2 lines in a rollover configuration. Audio quality on the line is perfect during voice calls. No static or other noise. I've asked for disconnect supervision to be added to the line, but It doesn't look like it's there. The line still seems to keep the channel open long after the far end hangs up. Has anyone else ever seen this problem or have any ideas how to eliminate it? Brent, I had a similar problem and I feel for you, its frustrating. Are you using polycom phones by chance? Here is the problem that I had, not sure if your problem is related. Specs: - 6 Polycom 301 phones. - CentOS 4 Server with Asterisk 1.2.x - Sangoma A200 card with 3 FXO ports. Pretty simple settings for a small office where a group ring between all 6 polycoms was initiated once the call was received. After that it would go to a auto attendant and give the caller option to continue to hold, leave a message, etc. At any rate, once in a while, Caller ID would fail, either on the Sangoma card or somewhere in Asterisk, not sure, but CLI clearly showed an error on caller ID every once in a while when a POTS call came in. All six polycoms would ring, but when you picked up the handset or hit the Answer soft button, nothing would happen, you couldn't answer the call. The phones would just ring, and ring and ring for the duration of the group ring (about 60) and the customer was really annoyed since it was a small office. Continuing, the problem finally turned out to be the polycoms! When no caller ID information was present, the polycoms wigged out and while they did ring, you could not get the phones to pick up. I could readily replicate the behavior by initiating a Call File without specifying the caller ID information using the local channel. It would happen every time. Specifying the CID would allow the polycoms to work correctly. On the customer side, I did a quick GoToIf in their dialplan to see if the caller id info was set and if it wasn't I would set it manually to something like: CALLERID(num)=555-555- CALLERID(name)=CID FAILURE That cleared up the problem. HIH -- Warm Regards, Lee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. Not web based, but: http://www.datatrakpos.com/pos/datatalk/maestro.aspx -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 includes many built objects available from scripts such as Cepstral TTS Engine class, database access class for FirebirdSQL, MySQL 4.1-5.0 and SQLite3 databases, Call File generation and more. Because AsterPas is a TCP socket server, it can be installed on the local Asterisk PBX or on a different computer to offload processing. AsterPas is written in 100% ObjectPascal using the Lazarus IDE for the FreePascal Compiler. Yes, ObjectPascal, It's not your mom and dad's pascal ;) AsterPas has been compiled and tested on: CentOS 4/Linux Windows 2000/XP Windows Server 2000 More information on AsterPas can be found on its web page at: http://www.datatrakpos.com/pos/datatalk/asterpas.aspx Source Code can be downloaded via svn or viewed from here: http://leebo.dreamhosters.com/asterpas/ Note: AsterPas relies on several 3rd party libraries: - Synapse (http://synapse.ararat.cz) (Open Source) - Pascal Data Object (http://pdo.sourceforge.net/) (Open Source) - sqlite3ds (included with lazarus/Freepascal) (Open Source) - TPasAGI (included with AsterPas sources, written by me ;) ) (Open Source) - RemObjects Pascal Script (http://www.remobjects.com) (Free with Source) -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 copy it to the outgoing directory all at 1 time. Yes, Create them with a future time and date and they will only be acted upon when the proper time comes. Doug -- There was a similar post recently and I wonder if using future time stamps would work well although I guess it depends on what you're doing specifically. In other words, if all you're doing is calling a number and playing a pre-recorded message for instance, that may be doable as you could at least estimate the average time it takes to dial and complete one of those calls and create your originate files at x intervals in timestamps. Otherwise for something more complex where the amount of time each call can take varies much, I would see some kind of queuing system being useful where the calls are tracked and originate files are produced as (specified) resources allow. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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-background.agi my conf background is a symbolic link - then my permissions are : [EMAIL PROTECTED] src]# ls -l /home/silentm/bin/conf-background.agi -rwSr-Sr-- 1 root root 81 Mar 17 10:44 /home/silentm/bin/conf-background.agi I have tried with just 744 and also with 744 and chmod +s. Any ideas why I would get the permission denied? Jerry I seem to have gotten past the permissions error by putting #!/bin/sh # at the top of my script file... Now when I run permissions issue is gone, however, When I try to access by variables (as I have dont many other times) I am getting an error. In my call file I have SetVar: MEETME_PLAYFILE=/home/silentm/record/pc.610.wav And when I asking for the value of MEETME_PLAYFILE it is giving an error about the PIPE being broken. Is there something special about this option b and what an AGI can and cannot do??? The AGI actually calls a C program (this is the same C file I have been using for a long time) and I just ask the AGI for the values of the variables. Jerry Is that an Originate command that you're using? Shouldn't that be: Variable: MEETME_PLAYFILE=/home/silentm/record/pc.610.wav ?? http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Originate Or are using the SetVar AMI Command? Sorry, I wasn't clear on this. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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, that sounds pretty good, can you hold for a sec? Telemarketer gets transferred to a context that plays a cepstral voice saying You have just been added to our Do not call list. Please add us to yours. Further attempts to contact us from your number are being recorded. Then adds the CID to a SQLite database and simply hangs up. The number is stored in a database at that point and if they call again they get Ceptral William saying Sorry, you have been added to our do not call database. You have been asked previously to place us on your do not call list. Each attempt to contact us by your number are being recorded and may be used in legal proceedings. Hang up. I've only had a couple actually call back. One called back about 6 times and my guess was that he was showing co-workers/managers the implementation we put in place or just got a kick out of it. I just shot off a letter to my Attorney General's office with the log and never heard from them again. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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, I do not want to send messages to Manager to generate all the calls at once, I just want to monitor the status of the call placed by me, so that I do not place more than 2 or 3 calls at the same time, hence not consuming all the available line on the Asterisk Server, leaving some lines for other people too. Please help. Regards, Sanjay. I could have sworn I heard/read of a settings that setts the maximum number of calls for AMI. I could very well be wrong though. I think you'll have to track the number that you called in the Originate command against Events specific to dialing like NewState and Hangup events. I'm been doing quite a bit of work with the AMI lately, but I'm still learning so take this with a grain of salt. Anyway, you could track the events as they come in. For instance, Event: Newstate Privilege: call,all Channel: SIP/axVoice-08f9d168 State: Up CallerID: 302381 CallerIDName: unknown Uniqueid: 1205515775.1003 You can check this event to see if the CallerID matches the number that you called. If so, then store the Channel variable and/or Uniqueid variable to match against other events that do not provide the CallerID. But now you know the number and the channel/Uniqueid the call is being made on. Hangup Event: Event: Hangup Privilege: call,all Channel: SIP/axVoice-08f9d168 Uniqueid: 1205515775.1003 Cause: 16 Cause-txt: Normal Clearing Here's what you're looking for. From what I can tell, the Hangup event will always fire if there is a NewState event which makes sense since the call is Live or up. Here we can take the channel variable and match it against our application's internal cache of active or attempting calls and decrement the count or free object references to ongoing attempts and then fire off another originate command. I'm sure someone more knowledgeable will pipe in and correct me where I'm wrong. HIH though. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 receiving lot of Events back, not able to recognize which was the call generated by me as same time multiple users are dialing out. 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 or download it directly from our site: http://www.datatrakpos.com/pos/datatalk/dpdownload.aspx I use it all the time when write apps for AMI. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 or download it directly from our site: http://www.datatrakpos.com/pos/datatalk/dpdownload.aspx I use it all the time when write apps for AMI. Both links work. Our website page that I linked to was changed recently which means .net compiles the page the first time its visited after its changed. I just tried it, both work ;) -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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/downloads/astmantest.zip -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. So, I cannot make any calls in, nor out. I am located in the United States. Is there any other suggestions, besides adding busydetect= yes and busycount=8? Any other suggestions would be appreciated. Thanks in advance. Here is what my zapata.conf looks like: I had a big problem as a similar situation. Polycom phones would wig out if caller id info came in wrong or there was an error (which looking at a few CLI's with analog zap, happens fairly often). The polycoms would ring, but would not pick up the call in this case. Not if you picked up the handset and not if you hit the Answer soft key either. Took me about a month to track it down and finally was able to reproduce by performing an originate AMI command on local channel without setting caller ID. Anyway, I finally just added a conditional to the entry point of my dialplan (after property answer(), Wait(), etc) that looked at the caller id and if they were blank, I set to something like 555-555-555/UNKNOWN and the problem went away. In this case, sometimes the zap channels would get hung up as well and required a restart now. HIH -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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... could you please help me with this issue? ill keep reporting thanks. I think it is because analog lines to not provide call progress like sip does. Someone more knowledgeable can correct me here if I'm wrong, but that is my first guess. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 please? Alternatively, are there any other GUIs (free or commercial) that reliably support attended transfers? I'm trying to replicate as much of the Avaya Phone Manager application in asterisk as possible, and really finding it quite a struggle. I haven't played with FOP personally, but I wonder if you can just park the caller, call the party you want to transfer to and then drag/drop the caller over onto the extension's gui object to do the transfer. We just finished writing a control panel application for a client for a linux/gtk based solution: http://leebo.dreamhosters.com/images/guiApp.png We retained rights to the business objects used in the application and are almost finished writing a Windows based solution as well. http://leebo.dreamhosters.com/images/maestro.jpg I have to be honest in saying that I never thought of doing attended transfer through the GUI, but it makes a lot of sense. According to this recent post to asterisk-dev, native attended transfers are not possible through the AMI right now without the patch mentioned. I'm not a bleeding edge patch type of guy though :( http://lists.digium.com/pipermail/asterisk-dev/2007-August/029206.html I don't think its a problem for our application since its functionality is completely derived from user entered AMI commands with variable substitution at execution time so I would imagine that an originate or redirect to dialplan logic could achieve this if attended transfers can be initiated from the dialplan. Anyone know if that is possible? Maybe by issuing digit tones? -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 way to do this would be to take the two original parties on the call and stick them in a meetme room using Redirect with ExtraChannel, then dial the new party and also dump them into the meetme room. The problem I am having is this: I know the extension of the SIP phone that is on the call, but I don't know it's channel, or the channel of the other party. I need to figure both of these out to be able to use the Manager API and dump those callers into the meetme room. Can anybody tell me how to determine the channels on an active call? Kind Regards, You need to track those calls somehow, Mitch. Someone can correct me where I'm wrong, but I see you can do this in a couple of ways. 1. Track the status of peers. My application performs a sippeers manager (and zapshowchannels) command to get the status of each device I'm watching at start up. As events are sent from AMI, I match each device with that event, specifically, the LINK event (changed to Bridge event in AMI 1.1). This way, when the user goes to click on or drag and drop a device on screen, we already know its information such as its channel info and linked channel information. 2. Another way I can think of would be to use the CLI command show channels from AMI and parse the output for your device. After figuring out which one is the device you're interested in, you can use the Status manager AMI command to get the info (including linked channel on the device). As you probably figured out, the Status command requires the channel of the device and not just its name/ident such as sip/114 so you have to go through the Show Channels hoop first, I imagine. As you say, its the easiest to just redirect both parties to an extension already setup in your extensions.conf. I also push channel variables from my application to Asterisk channel vars for use in the dialplan. This way I can have a bit of dynamic operations. If my user want to create a new conference by dragging a live sip phone to the conference view of my application, I just prompt the user for conference number, send it as a var along with my redirect request to AMI and use dialplan logic from there. As I said, I'm still learning (although learning a lot!) about AMI operations as I build my own application for AMI so take my info with a minuscule portion of sodium. ;) -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 originates a call.
Thanks. -A
Anthony,
I may not understand your question, but setting variables from the AMI is easy
enough:
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Conference Waiting
Async: true
--
Warm Regards,
Lee
Everything I needed to learn in life, I learned selling encyclopedias door to
door.
___
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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)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 originates a
call.
Thanks. -A
Anthony,
I may not understand your question, but setting variables from the AMI is
easy enough:
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Conference Waiting
Async: true
That was exactly my question (even though I forgot the =sign). However, I
am
not able to get that to work for reason. I'm trying to set the
CDR(accountcode) on the first leg of the call and am using Channel: Local/...
I am able to get it to work if I use Variable: var1=12345 then, use
CDR(accountcode)=${var1} in the dialplan, but I was hoping to avoid this
hack.
Not sure what could be the reason, maybe something in the cdr stuff and call
origination maybe?
--
Warm Regards,
Lee
Everything I needed to learn in life, I learned selling encyclopedias door to
door.
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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 see the flow of status, etc., but I'm sure there is a better way to do this without using Telnet (maybe not?). Any suggestions? I want to write a presence monitor (a virtual sidecar if you will) Bill As Razza said, you can just use the winsock control included with VB. The protocol is very simple, basically just name/value pairs delimited by #13#10 (CRLF) with an extra CRLF at the end to denote termination of the packet. Action: Originate Channel: local/[EMAIL PROTECTED] Context: to_meetme Exten: s Priority: 1 Variable: CALLERID(num)=123432|CALLERID(name)=Automated Call Async: true extra CRLF == extra CRLF here. So, like this: 1. Send your properly formatted packet to AMI . 2. Read incoming response terminated by double #13#10. 3. Parse values as you are comfortable with. I am in the process of writing a similar product for one of our customers. Well, a re-write to add features and make it cross platform. Here's a screenshot running on Linux/GTK: http://leebo.dreamhosters.com/images/guiApp.png A couple of side notes from what I've learned myself and read on this mailing list or through the wiki: 1. Packet Volume The volume of messages that you can get from the AMI is impressive. I've tested on our Asterisk system which has only 2 pots lines and two sip trunks with 10 desktop phones and the amount of messages can be staggering! Use a proxy for AMI if you have any decent phone traffic. AstManProxy is VERY propular. I wrote one as well, but its still beta and I think there's another one out there somewhere. Usually with these proxy servers you can filter out unwanted/extraneous events to reduce the amount of messages your app has to contend with. 2. Make good use of Observer/Mediator pattern to distribute events to different parts of your GUI. Monolithic loops to write everything out on a timer's event or after a Sleep() for instance, is not a good way to go in my experience. 3. Check the source for manager interface for changes between Asterisk 1.2 and 1.4 (and 1.6?) if you're using 1.2 or plan to. I believe the latest version of AMI is 1.1 (someone can correct me here). A few label names for some of the AMI packets have been changed and a couple events (like LINK event) have been changed drastically. I originally wrote against the 1.2 Manager interface only to find that I had to refactor some code and write descendant classes to handle the slight differences between the two versions' events. I could have saved myself some work had I thought to look for the changes. I think this link is up to date: http://svn.digium.com/view/asterisk/trunk/doc/manager_1_1.txt?revision=98152view=markup Happy coding. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 and traffic associated with those from asterisk using a C Code. Thank you. Soumya, This may help: http://www.voip-info.org/wiki-Asterisk+manager+API Not sure what you mean by traffic though. For call history, you might look at: http://www.voip-info.org/wiki/view/CDR For current status of sip lines, etc. the Asterisk Manager Interface (AMI) is still your friend. AMI command SipPeers will force all event packets to be issued for sip peers which you can catch and analyze. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 is more of an class object wrapper around AGI functions that I wrote a while back because I'm lazy that way ;) Indeed and I think such a higher level API could be implemented in different languages. There is/was a port of the Asterisk-Java API to .Net at least. I think especially the live API of Asterisk-Java is worth having a look at. It provides an object view on top of AMI with rich objects like Channel and methods like hangup() and redirect(). So it makes the developer focus on his tasks rather than thinking in terms of actions and responses. 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 through FastAGI or AMI without knowing about the underlying details. Where is more information on this new feature for Asterisk 1.6? Any details? If someone is interested in defining a language-neutral general higher level API that can be implemented in a variety of languages I am happy to support this effort. This would be refreshing as the current AMI output is a little all over the place. Example: Conf Num PartiesMarked Activity Creation 1110001 0001 00:17:57 Dynamic Above is a line from MeetMe command issued from AMI. After the header line, each successive line denote information about a conference. No problems there, except there is an extraneous Tab (#9) character right after the Parties field which screws you up when parsing until you figure out that there is a Tab character there. There appears to be no reason to have a tab character there that I can see, well maybe to trip up unwary developers ;) I'm not sure what your point is, but I'll say that I'm a definite proponent of abstraction layers provided they don't bar access to lower level logic when I need it. I think you'll agree that good abstractions lend themselves to reuse and reduced development time (easy of use, less runtime logic errors, easier to extend, etc). And don't miss the additional benefit of supporting multiple versions of Asterisk that you get almost for free. Asterisk-Java will run with Asterisk from 1.0 to 1.6 without changing your code even if the Asterisk guys decide to rename properties and the like. Just have a look at doc/manager_1_1.txt in the betas of Asterisk 1.6 and decide what your efforts would be to support Asterisk 1.4 and 1.6 if you stick to low level APIs. Another great reason for abstraction/encapsulation IMO. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 through FastAGI or AMI without knowing about the underlying details. Where is more information on this new feature for Asterisk 1.6? Any details? I wrote this blog entry when I was writing the AsyncAGI feature: http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ This is the bug entry: http://bugs.digium.com/view.php?id=11282 I changed my mind regarding the behavior of this feature after opening the bug entry, so the initial description of the bug can be confusing and totally different from the final implementation and behavior, so you will have to read all the comments in the bug entry to understand what is this about. Moisés Silva Thanks. That's pretty slick. It could add some flexibility, but as you noted on your blog, you could just as easily redirect to a FastAGI server, etc. Being able to call AGI's on a channel through the AMI seems like it could have some possibilities as well. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 is more of an class object wrapper around AGI functions that I wrote a while back because I'm lazy that way ;) I always figured that was kind of the whole point of such bindings, so nothing about it strikes me as incomplete or lacking in a sufficient degree of reality. The only difference between this and Asterisk-java is simply that the latter encapsulates many of these actions in more high-level wrappers, which is likely to be a concession to the phenomenology of Java Thinking(TM) more than anything else. Your mileage may vary. I'm not sure what your point is, but I'll say that I'm a definite proponent of abstraction layers provided they don't bar access to lower level logic when I need it. I think you'll agree that good abstractions lend themselves to reuse and reduced development time (easy of use, less runtime logic errors, easier to extend, etc). -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 since Jan 1, 1970 UTC, as I recall. Thanks. Apparently Asterisk reports it in QueueMember* AMI events as GMT based as well, requiring that we apply our regional offsets (GMT -5 for instance). -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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? 2) Is there any chance of a real API for Perl? What is your criterion of real? That is to say, what do you need that it does not provide? I've used AGI and FastAGI in Perl extensively and it is yet to fail to serve my purposes. Maybe Ken is referring to a pre made framework like Asterisk-Java or PasAGI. I don't know perl so maybe there *is* a framework already. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. Thanks. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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.
The following doesn't work: The file never gets written:
===
//check_cid.c
#include stdio.h
#include stdlib.h
#include syslog.h
#include string.h
int main(int argc, char *argv[])
{
char line[80];
int i;
setlinebuf(stdout);
setlinebuf(stderr);
FILE *file;
file = fopen(file.txt,a+);
while (1) {
fgets(line,80,stdin);
fprintf(file,%s,line);
if (strlen(line) = 1) break;
}
fclose(file);
return(EXIT_SUCCESS);
}
===
This is how it's called in extensions.conf:
===
[inside]
exten = ,1,Verbose(Yes!)
exten = ,n,AGI(check_cid.exe|123)
===
And this is the output of agi debug in CLI:
===
*CLI -- Executing [EMAIL PROTECTED]:1] Verbose(SIP/2000-0904bee0,
Yes!) in new stack
Yes!
-- Executing [EMAIL PROTECTED]:2] AGI(SIP/2000-0904bee0,
check_cid.exe|123) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/check_cid.exe
AGI Tx agi_request: check_cid.exe
AGI Tx agi_channel: SIP/2000-0904bee0
AGI Tx agi_language: en
AGI Tx agi_type: SIP
AGI Tx agi_uniqueid: 1201412176.3
AGI Tx agi_callerid: 2000
AGI Tx agi_calleridname: Fred
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callington: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid:
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: inside
AGI Tx agi_extension:
AGI Tx agi_priority: 2
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx
-- AGI Script check_cid.exe completed, returning 0
== Auto fallthrough, channel 'SIP/2000-0904bee0' status is 'UNKNOWN'
===
If someone has a very basic example in C that shows how to read the
CID #, and rewrite the CID name, I'm interested.
Thank you.
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 the file to a fully qualified path?
I say that because in pascal, usually you can do this on Windows:
var
sFile: string
AStringList: TStringList;
begin
sFile := ExtractFilePath(ParamStr(0)) + 'myfile.txt';
AStringList := TStringList.create;
try
AStringList.LoadFromFile(sFile);
Write(AStringList.Text);
finally;
AStringList.free;
end;
end;
Normally ExtractFilePath would return the directory path that the executable is
locate in. My understanding is that that is not necessarily the case on linux
which takes into account the directory from which the call to the executable is
being made, which might not be the same directory that the executable is
located on.
Could be the same thing you are experiencing. Try using a fully qualified path
/usr/mydirect/myfile.txt
--
Warm Regards,
Lee
Everything I needed to learn in life, I learned selling encyclopedias door to
door.
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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 the file to a fully qualified path? My hero! :-) That did it. Both work: file = fopen(/tmp/file.txt,w); file = fopen(./file.txt,w); Also, and contrary to what I've read in some online pages, we don't have to put AGI scripts in /var/lib/asterisk/agi-bin: exten = ,n,AGI(/tmp/check_cid.exe|123) BTW, as I'm used to Delphi, I'm also interested in checking out FP. Is it easy to install on eg. CentOS? Can I come up with a totally self-dependent EXE that I can just drop into a limited host running AstLinux? I've used the Windows version for most development, but have lazarus running nicely on a CentOS4 VM. Installing from the setup program on windows and from rpm on linux was easy. Just as with delphi, freepascal creates native stand alone executables as well as libraries (.dll or .so) and with speed on par (but not as fast) as C . Lazarus is the IDE that I use to work with Freepascal. Lazarus is now about on par with Delphi 6, IMO and one of the best IDE's available for Linux. Not the prettiest, but one of the most useful ;) I also wrote an objectpascal based AGI/FastAGI wrapper a while back if you're interested: http://www.leebo.dreamhosters.com/asterisk/pasagi.pas -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 its compile it should run on Mono. Try using MoMA to test for compatibility: http://www.mono-project.com/MoMA -- Warm Regards, Lee The only thing that kept me out college...was high school. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 its compile it should run on Mono. Try using MoMA to test for compatibility: http://www.mono-project.com/MoMA Oops, looking a little closer, it appears that it is a standard win32 VB app and not vb.net. My mistake... -- Warm Regards, Lee The only thing that kept me out college...was high school. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 written about 70% of the core functionality so I decided to continue on. I'm not a C programmer so having something in my preferred language to use and extend later on is nice. I still most of my development on Windows so I haven't had a chance to build any Linux binaries other than for debugging, but should have some ready in the next week or so as Linux testing continues. In the meantime, there are Win32 binaries in the repository. Currently I'm working on xml and ini based decorators to customize the packets to/from clients. I'm using the proxy for a re-write of an existing operator panel I have in order to make it cross platform, but I've released the proxy itself released under GPL. It is written in ObjectPascal using Lazarus IDE (0.9.24) with Freepascal compiler (2.2.0); Sources are available here: http://www.leebo.dreamhosters.com/AMIProxy/ -- Warm Regards, Lee The only thing that kept me out college...was high school. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. One thing is that it requires (if I'm not mistaken) an mysql installation which is too heavy of a dependency for some applications that I have in mind to write. It does not require an mysql of any type at all. I thought for sure I read that it did require a mysql database. Thanks for correcting me on that. I've already put in quite a bit of work on it so I'll continue with mine, but had I known this, I wouldn't have written my own. -- Warm Regards, Lee The only thing that kept me out college...was high school. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 access to a high (or even medium volume) system as this office only has 4 extensions with maybe 50 calls a day. My particular needs for this software: * Real-time, event driven interface to Asterisk AMI, no polling. (done.) * Event Filtering. (somewhat done.) * Various Decorator plugins to customize packets/requests. (plain and xml done.) * Easy configuration of users that mimic existing Asterisk AMI permissions model. (almost done.) Any thoughts, suggestions or comments welcome. -- Warm Regards, Lee The only thing that kept me out college...was high school. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 that I have in mind to write. The proxy I'm writing allows real-time traffic between proxy clients and the Asterisk AMI and the traffic is cached maybe 50-100ms before being sent to either end so there is no need to save the packet traffic to a database. For me, I need something lean in terms of 3rd party software and lean on memory with a simple deployment of an executable and maybe a few config files. I figured since I was writing it anyway, I'd just release it to the community... I'll post when its ready for testing or usage along with link to sources. Thanks again, -- Warm Regards, Lee The only thing that kept me out college...was high school. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. sarcasm mode=SCNR class=ignore Some tutorials: http://www.google.com/search?q=learn+c+in+21+days When done ask for commit access to http://svn.digium.com/view/zaptel/ /sarcasm Regards, Philipp Kempgen That was an inappropriate answer, Phillip. The OP said he was new to zaptel programming, not necessarily C programming. You could have actually just ignored the post/query for real without wasting bandwidth and being uncivil just to act superior and show everyone you can write a simple xml document. Its not that clever and is far removed from the usual quality posts I've seen from you. Bhrugu, you should check out the dev mailing list for specific questions regarding coding and development. -- Warm Regards, Lee If I don't see you around here, I'll see you around, hear? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. And this is a 300E no-name box. If your PC is so unstable, you should investigate the hardware and/or the device drivers. Maybe. Its not that its unstable, the system just becomes progressively slower and less responsive if I don't reboot once in a while. I also run scandisk and defrag weekly. Of course, it may have just as much do with the type of apps that I have open and running all the time as well. As I said, I like Windows, but I don't see a Server 2000 box out performing a comparable linux box for larger pbx systems. A small office, sure. I wonder if the linux box was also running Gnome or some other desktop at the same time, would that make it a closer comparison? Maybe Windows would outperform the linux box then? -- Warm Regards, Lee If I don't see you around here, I'll see you around, hear? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 solution to installation http://www.voip-info.org/wiki/view/DTSwift+Cepstral+AGI+Wrapper Works fine for me. -- Warm Regards, Lee If I don't see you around here, I'll see you around, hear? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 follwing way 1)modified manager.conf to facilitate this API to talk to asterisk 2)used the command Originate to call a Registered user under asterisk and when the user answers the phone it plays whatever i put against the extension.. But my exact requirement is like this 1)Call to the user 2)if answers connect him to the extension provided in the extensions.conf 3)if the user didnt lift the phone within the deault timeout period(30 sec) 4)if the user cancels the phone (Congestion case) 5)if the user not registerd to the(unreachable case) to trace the cases of 3, 4, 5 how should i follow the API I got confused with originate action,orginate sucess event , originate failure event http://www.datatrakpos.com/pos/datatalk/dpdownload.aspx Download the Manager API Testing Utility. I wrote to help me with a software program that I was writing that used the Manager API heavily. Allows you to view the AMI activity, send commands, etc outside of your dev environment. Helped me a lot and its fee to use. You can also get it on: http://www.voip-info.org/wiki/view/Asterisk+GUI Wish I had more time to do Asterisk related development, its a lot of fun... -- Warm Regards, Lee If I don't see you around here, I'll see you around, hear? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 half-baked, dying OS that in essence is a 32 bit extension and graphical shell, for a 16 bit patch to an 8 bit operating system, originally coded for a 4 bit microprocessor, written by a 2 bit company, that can't stand 1 bit of competition. Nice. Do you really want to reboot your telephone system 3 times a day? I'm not a Windows basher as I make a good living from Windows based software, but I couldn't see it either. My asterisk box was rebooted about 3 months ago when I made some changes last. It's running Asterisk, FirebirdSQL, 1 FastAGI server and a lot of natively compiled AGI executables handling tech support, sales, caller id database lookups, nag calling, etc, etc. I have to reboot my desktop xp box daily for it to run well. -- Warm Regards, Lee If I don't see you around here, I'll see you around, hear? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 integration with Asterisk, but does that include say Skype IN and Skype OUT ? Or is that integration component really just for being able to contact skype users? Looking for the easiest and cheapest way to reach the PSTN, and well the options out there are plenty regarding SIP trunks, but most tend to be geared towards businesses for obvious reasons. Curious what others are using, and if anyone can make some recommendations? Not sure if this has been covered already on the list, and not sure if recommending companies are allowed, so maybe I need get replies off list? 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. --- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. I think the pay as you go plans make more sense for most people -- why do you think the vendors push the flat rate plans? At $25.00 per month, you'd have to be on the phone for about an hour a day for it to be cheaper than a $0.015 per minute plan. True, but I work from home, have a wife and 4 kids with friends and family all over the U.S. so it makes more sense for me. Good point though, Steve. --- Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. Even better question for me is if Asterisk can call libraries not written in C, but that export their routines under cdecl calling convention. This might be a better question for dev lists though. I'd really like to start writing some .so libraries for use within Asterisk without having to use AGI. -- Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 presented two arguments - the first is '-rx', the second is extensions reload (needs additional quoting to contain the space) which is actually a parameter to the '-x' switch just used. $output = shell_exec(asterisk -rx 'extensions reload') is right. Generally, the difference between single quotes and double quotes is that with double quotes, PHP is allowed to make $variable substitution while with single quotes, it is not. Mojo Nice. Thanks for the tip, Mojo. --- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. Why most people don't use it? Am I missing something? I just think its the default so probably many new people to Asterisk start there and then possibly move over to AEL or AGI scripts later on as needs become more complex... For those that have been in the Asterisk community for a longer period of time, the traditional flat line script was all that was available until AEL came along as far as I know. I wrote an automated dialplan generator so much of *our* systems had the traditional flat script because its much easier to produce that traditional asterisk script from a GUI that generates script for you. I prefer pascal syntax personally, so we use a pascal based AGI/FastAGI engine that I wrote for much of our more advanced logic. In the end, it probably comes down to preference and need, I would think. Nice to be proficient in writing it all; flat scripts, AEL, AGI/FastAGI/Manager API (using your programming/script language of prefernce)this way we can have more tools to solve more problems for our customers or company. --- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 command line, it runs just fine (works perfect, actually).
I think it is permissions related. Does anyone have any ideas?
php
$output = shell_exec('asterisk -rxextensions reload');
echo $output;
?
I'm not a PHP guy, but shouldn't the double quote be surrounding the
entire shell command like this?
$output = shell_exec('asterisk -rx extensions reload');
Lee
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Re: [asterisk-users] Stable-Stable Asterisk
Steve Totaro wrote: David Gomillion wrote: On 8/23/07, *Ed Pastore* [EMAIL PROTECTED] mailto:[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 bleeding-edge. They qualify as stable, but I wouldn't call them production stable since half the time a new one comes out, a fix for it comes out the next day. That's the niche that ABE is supposed to fill. I personally don't use it, though. I just test the features I plan to use, disable everything else, and seem to do OK. I stay with 1.2.12 or somewhere around there. End Of Life but seems to have a better ticker than 1.4. Thanks, Steve 1.2.12/14/17 all have seemed very stable to me so far. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 this administrator what Asterisk config files are but I'm wondering if any GUI exists for such task (editing a vocal menu tree). Maybe something not related to telephony could be used for that. Any idea ? Best regards Oliver, You can check out DialplanPro if you like. Its very easy and graphical to create dialplans for our users. http://www.datatrakpos.com/pos/datatalk/Default.aspx Unfortunately, roles/permission groups are not yet implemented (another reason why its still in beta) so the user would have the full gambit of functionality. However, if you feel comfortable with the admin, then you can simply create the initial menus and train her to modify only those menus. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 commercial PBXs to a more pure asterisk setup. The systems I have utilize a nice web GUI to make changes, but it really limits what I can do beyond what they have programmed into their GUI. Bill, If you like working from Windows, you can also check out DialplanPro. I've been using it for our few (so far) clients and our personal phone system. http://www.datatrakpos.com/pos/datatalk/Default.aspx I wrote it to be more of a swiss army knife for Asterisk. I like to use the GUI widgets and visual menu builder to build the basic dialplan menus then use the editor (basic syntax highlighting, parameter suggestions, etc) to write custom scripts using either traditional flat asterisk script or AEL2 and INCLUDE them in the final project scripts which can be automatically uploaded to the server. I also use it to parse my AEL2 scripts remotely from my windows computer using a hook into the aelparse executable written by murph. Its still beta, but mostly because it doesn't yet have all the features I want to eventually include in it. Also, its commercial software or will be someday. -- Warm Regards, Lee Keeewwwl...Delphi! However, all I can get it to do is generate errors: == Application... Start Date : 08/17/2007 20:20:27 Name/Description: astclient.exe Version Number : 0.9.6.75 Exception... Date : 08/17/2007 20:22:40 Address: 00409A5A Module : astclient.exe Type : EConvertError Message: '' is not a valid integer value. Doug, We get about 120 downloads of that product a day and this is the first time I've seen this error so I would be very interested in tracking it down as no one has reported it. It looks like you're trying to build your project but there is some data its looking for during build that is not there (trying to cast a string to an integer, but the string is empty it looks like). Probably just a setting that you did not set. Most settings (as you would think) that are critical to the application have mandatory values and I'd love to find out which one this is to that it can be required as well. Would you mind dropping me a line off list or posting the error to our message board? http://www.datatrakpos.com/community/ (Main Message Board) http://www.datatrakpos.com/community/Default.aspx?g=topicsf=20 (DialplanPro Forum on the board) Yes, we are a Delphi shop. Also Lazarus/Freepascal, C#/.Net, etc. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 what I can do beyond what they have programmed into their GUI. Bill, If you like working from Windows, you can also check out DialplanPro. I've been using it for our few (so far) clients and our personal phone system. http://www.datatrakpos.com/pos/datatalk/Default.aspx I wrote it to be more of a swiss army knife for Asterisk. I like to use the GUI widgets and visual menu builder to build the basic dialplan menus then use the editor (basic syntax highlighting, parameter suggestions, etc) to write custom scripts using either traditional flat asterisk script or AEL2 and INCLUDE them in the final project scripts which can be automatically uploaded to the server. I also use it to parse my AEL2 scripts remotely from my windows computer using a hook into the aelparse executable written by murph. Its still beta, but mostly because it doesn't yet have all the features I want to eventually include in it. Also, its commercial software or will be someday. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 happens, it appears that the call still goes through as I can see the caller still navigating through the systems menus and dialplan by watching the CLI. The problem however is manifested with polycom 301's that are setup with the system. When a call comes in after receiving that particular caller id error, the polycoms, which are on a group ring by the way, will all ring but you cannot pickup the call. The Answer|Reject soft buttons display, but only the reject button works. Pressing the Answer button or picking up the handset does nothing. Since only the Reject button works someone has to go to each phone and hit the reject button (4 polycoms in this department) so the ringing will at least stop. It's been about 3 months tracking this problem down (even drove the 2.5 hours back and forth to replace the sangoma card to try to fix the problem) and the customer is about ready to have me pull the system because of it. I can easily reproduce the problem with Polycom phones (but not the actual error). Just issue a .call file using the local channel calling one number and having the call bridged to a polycom phone (at least 301's here): Action: Originate Channel: local/[EMAIL PROTECTED] Context: to_meetme Exten: s Priority: 1 Async: true The above will cause the polycom to exhibit the behavior mentioned above. However, sending a .call file like the following causes the phone to work as it should: Action: Originate Channel: local/[EMAIL PROTECTED] Context: to_meetme Exten: s Priority: 1 Variable: CALLERID(num)=1234|CALLERID(name)=Homey D Clown Async: true I also have tried this with Aastra, Grandstream and XLite soft phones and they do not exhibit the same behavior. Instead these other phones simply show the default caller id info as set in sip.conf and allow you to answer them. Any help or suggestions would be greatly appreciated. OS: CentOS 4 Asterisk: 1.2.17 Sangoma A200 with 2 fxo ports. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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
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 happens, it appears that the call still goes through as I can
see the caller still navigating through the systems menus and dialplan
by watching the CLI.
The problem however is manifested with polycom 301's that are setup with
the system. When a call comes in after receiving that particular caller
id error, the polycoms, which are on a group ring by the way, will all
ring but you cannot pickup the call. The Answer|Reject soft buttons
display, but only the reject button works. Pressing the Answer button
or picking up the handset does nothing.
To me this looks like a firmware problem in your phones. Perhaps a
firmware update could fix this. However - as it looks to me - the
firmware chokes on some CALLERID strings, not on others. What is the
caller id that is displayed in the error case? Perhaps you could get
around by having a dialplan hook that rewrites the callerid to 000 if
that invalid callerid comes in. Maybe those phones just choke on
CALLERIDs with empty num or name With your test .call file that
reproduces the problem, if you insert a line in your dialplan before the
Dial() happens, that reads
Set(CALLERID(all)=000)
does that help? Does
Set(CALLERID(num)=000)
alone help, does
Set(CALLERID(name)=000)
?
BR
Anselm
Anselm,
Thanks for responding. My apologies as I should have mentioned that I
have tried several workarounds including the following test scripts I
placed on the server:
[check_time]
; -
; Called right after Answer() is called
; -
; check for default value in sip.conf
exten=s,1,GotoIf($[${CALLERID(name)} = UNKNOWN ]?set_no_callerid,s,1)
; check for null value
exten=s,2,GotoIf($[${CALLERID(name)} = ]?set_no_callerid,s,1)
exten=s,3,GotoIf($[${CALLERID(num)} = ]?set_no_callerid,s,1)
exten=s,4,Noop(CallerID: ${CALLERID(num)} ${CALLERID(name)})
exten=s,5,Set(FAIL_MENU=daytime|TIMEOUT_MENU=daytime)
exten=s,6,GotoIfTime(08:30-17:00,mon-fri,*,*,?daytime_ivr,s,1)
exten=s,7,Goto(after_hours,s,1)
[set_no_callerid]
exten=s,1,Set(CALLERID(num)=410555)
exten=s,2,Set(CALLERID(name)=UNKNOWN)
exten=s,3,Goto(check_time,s,1)
I'll update the firmware on the phones and see if that helps.
Thanks again,
--
Warm Regards,
Lee
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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] Set(TIMEOUT(digit)=3 Set(TIMEOUT(response)=60 These are missing closing brackets for one thing... -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 DialplanPro: http://www.datatrakpos.com/pos/datatalk We're still considering it beta, but we use it for our own pbx and those of the few clients we have using Asterisk and it works very well. It's also commercial (or will be someday...) Either way, its in beta and free to use if you like. --- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 to make sure I don't put two people
together !)
I was going to use a global variable ${NEXTMEETME}, and add one every
time I redirect.
Is the changing of a global variable atomic ? That is, if I have two or
more channels being redirected at the same time, and they all execute
exten = _X.,n,Set(NEXTMEETME=${MATH(${NEXTMEETME}+1,i)})
exten = _X.,n,Set(MYMEETME=${NEXTMEETME})
if NEXTMEETME is initially 0, would channel A get MYMEETME as 1, channel
B get 2 and channel C get 3, even if they execute the dialplan at the
same time ?
Someone more knowledgeable about Asterisk than I can correct me, but I
would look at it from the perspective any development environement:
Global variables are typically bad in a threaded environment without
some form of queuing/locking/critical section functionality to avoid
collisions.
If I needed a globally unique, sequential number, I'd push it out to
AGI/FastAGI so I could use a language with support for locking/queuing.
A DB like MySQL or FirebirdSQL would easily handle this need as you
know, but then is the overhead of establishing DB connections worth it
for simply getting a incremented int?
--
Warm Regards,
Lee
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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 error please notify us immediately
and then destroy it.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julian Lyndon-Smith
Sent: Saturday, July 28, 2007 5:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] global variables and updates
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 to make sure I don't put two people
together !)
I was going to use a global variable ${NEXTMEETME}, and add one every
time I redirect.
Is the changing of a global variable atomic ? That is, if I
have two or
more channels being redirected at the same time, and they all execute
exten = _X.,n,Set(NEXTMEETME=${MATH(${NEXTMEETME}+1,i)})
exten = _X.,n,Set(MYMEETME=${NEXTMEETME})
if NEXTMEETME is initially 0, would channel A get MYMEETME as
1, channel
B get 2 and channel C get 3, even if they execute the dialplan at the
same time ?
The changing of variables is not atomic as would hope, but there is a
solution for you. Look the application MacroExclusive. Put your Set to
increment the global variable inside of a macro and call it using this,
and you will get the behavior you desire. One caveat, however, is that
you will want as little logic as possible inside of this macro.
MacroExclusive will block all other calls to this macro until the first
one exits. But this is not an issue if all you are doing is a quick
var++ and then leaving.
That's a very nice feature. A quick Google search on the wiki didn't
turn up any topics. Does it queue subsequent calls or just block them
and then logic in the dialplan must be used against a return value?
---
Warm Regards,
Lee
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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 announcements into the mp3 file... -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 replace a few Micros systems a year. Same thing with some of the other brands out there. We've had a few switched from us over the years as well. Basic attrition, I guess. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. Has anybody connected Asterisk with Micros-Fidelio? As I understand this isn't some local developed application, it's something that is used world wide. Any informations are welcome. I wrote a middleware bridge (TCP = Serial) for Micros a 2 or 3 years back and it was relatively simple. This was the serial interface for the 8700 standard. If I remember correctly, it was a simple string that was broken up into fixed length fields like char 1 through 10 was a field and chars 11 through 15 was a field, etc. If you need help, email me off list and I'll look for that source code. Lucky for you it was written in pascal so its easy to read ;) Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 from the queue if there are no agents present, add the following line to your queues.conf file: leavewhenempty = yes http://www.voip-info.org/wiki/index.php?page=Asterisk+config+queues.conf -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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.php?page=Asterisk+config+queues.conf -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. Has anybody connected Asterisk with Micros-Fidelio? As I understand this isn't some local developed application, it's something that is used world wide. Any informations are welcome. I'd say that Micro is the MS of Restaurant POS. We replace their systems regularly ;) I have some contacts in the PMS field and I will ask them about it. They were very helpful when we integrated our POS software with Micros/Fidelio to post charges from the restaurant to guest folios. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: dropped_conf=111
The to_meetme context is very simple:
[to_meetme]
exten=s,1,MeetMe(${dropped_conf},id)
If I specify every other device I have to test:
* Grandstream 101
* XLite Client
* My Cell Phone
It works as expected. But with the Polycom, the phone will ring and the
usual ANSWER REJECT FORWARD soft buttons are painted on the display, but
hitting the answer button seems to fail to do anything other than
silence ringing.
SHOW CHANNELS shows the polycom as ringing still although the polycom
has stopped ringing (audibly at least).
Of course, all other calls originate through the dialplan are answered
with no problem.
It appears that it is something with my Polycom configuration. It seems
like the polycom is having a problem with calls that do not contain
correct CID info.
In the originate command, I added some lines to populate the CALLERID(x):
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Operator
Async: true
After explicitly setting the Caller ID info, the polycom then accepts
the call correctly.
Anyone know off hand what setting might be creating this behavior?
Thanks again,
--
Warm Regards,
Lee
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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 Location: SIP/112 Membership: dynamic Penalty: 0 CallsTaken: 2 LastCall: 1184016974 Status: 1 Paused: 0 -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 communication seems to be best through email when dealing with them. Emails however, are returned very promptly. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 this, but QueueMemberStatus returns the same codes for channel status as defined in devicestate.h. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 solution Regards Satish Patel And this: http://www.voip-info.org/wiki-Asterisk+config+features.conf Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. On 6/30/07, *Jordan Novak* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have three polycom 501 that are all hearing echo. The other party sounds fine but you can hear yourself rather well. The volume does help if lowered but that also makes the other party extremely quiet. Is there any way to control the gain of the mic or stop the microphone from picking up so much from the handset. It only happens while you are on the handset. If you're using FXO/analog lines, I'd recommend trying Octware's software echo canceler. I had the same problem on a recent installation and it fixed it. At $10.00 per channel, its a very good value, IMO. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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://www.voip-info.org/wiki-Asterisk+config+features.conf -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 shouldn't be seen as a 'solution', as callprogress has it's place (disconnection detection, etc). Don, have any changed been made to your zapata.conf immediately before this issue started occuring? Jason. I thought that callprogress was highly experiemental according to the wiki. Not sure how recent that information is though. -- Warm Regards, Lee ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 here them. The customer hangs up and calls back and the call goes through.. I rolled back to 1.2.14 and the problem is much better but is still there, Are there any ideas Don Briggs 573-614-5667 ext 4037 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. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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=s,1,MeetMe(${dropped_conf},id)
If I specify every other device I have to test:
* Grandstream 101
* XLite Client
* My Cell Phone
It works as expected. But with the Polycom, the phone will ring and the
usual ANSWER REJECT FORWARD soft buttons are painted on the display, but
hitting the answer button seems to fail to do anything other than
silence ringing.
SHOW CHANNELS shows the polycom as ringing still although the polycom
has stopped ringing (audibly at least).
Of course, all other calls originate through the dialplan are answered
with no problem.
Anyone have an idea what might be causing this? Its a polycom 301 with
lines 1 2 registered to separate sip accounts in sip.conf.
Thanks for any suggestions.
--
Warm Regards,
Lee
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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. There should be enough information to get some idea of why this is happening. Thanks, I have done this yesterday by setting up putty to log to a file, but the customer employees have inadvertently shut it down on a couple of a occasions :) Hopefully it will be running when this happens again so I can try to track down the problem. You should be able to tell it to log to a file in addition to the console in logger.conf. Something like: full = notice,warning,error,verbose Then it should show up in /var/log/asterisk/full and you wouldn't need to keep a session open to the console to see it, just go back and look at the file later. Nice tip, Dave. Thanks, -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phantom Calls
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. There should be enough information to get some idea of why this is happening. -Original Message- 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 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 when they answer the phone, there is only silence and then they hang back up and it sometimes rings again. I've been trying to track this down for a couple of weeks now with no success yet. If anyone can lend a suggestion or a pointer to look for, I would greatly appreciate it. I've tried using WaitForRing() in case it is bad signaling coming from the phone company. But that has not helped. Below is the CLI output when this issue happened. As you can see, I am using WaitForRing() to discourage phantom calls. Every time this has happened, there appears to be an error getting caller ID. I'm thinking that if I insert a Wait(1/2) before Answer, that may resolve the problems with Caller ID as it looks like Asterisk is not waiting long enough for the CID to come in. Whether or not that will fix the problem with phantom calls remains to be seen after I make the changes. Also notice, the line: localhost*CLI -- Got SIP response 400 Bad Request back from 192.168.1.216. What does 400 Bad Request usually mean for sip? Generic message or something that would provide a clue? localhost*CLI -- Starting simple switch on 'Zap/3-1' localhost*CLI Jun 21 10:44:48 NOTICE[11257]: callerid.c:325 callerid_feed: Caller*ID failed checksum localhost*CLI Jun 21 10:44:51 NOTICE[11257]: chan_zap.c:6233 ss_thread: Got event 18 (Ring Begin)... localhost*CLI Jun 21 10:44:53 NOTICE[11257]: chan_zap.c:6233 ss_thread: Got event 2 (Ring/Answered)... -- Executing WaitForRing(Zap/3-1, 1) in new stack localhost*CLI -- Got a ring after the timeout -- Executing Answer(Zap/3-1, ) in new stack -- Executing Ringing(Zap/3-1, ) in new stack -- Executing SetMusicOnHold(Zap/3-1, default) in new stack -- Executing Goto(Zap/3-1, check_time|s|1) in new stack -- Goto (check_time,s,1) -- Executing Set(Zap/3-1, FAIL_MENU=daytime|TIMEOUT_MENU=daytime) in new stack -- Executing GotoIfTime(Zap/3-1, 08:30-17:00|mon-fri|*|*|?daytime|s|1) in new stack -- Goto (daytime,s,1) -- Executing Set(Zap/3-1, TIMEOUT(response)=1) in new stack -- Response timeout set to 1 -- Executing Dial(Zap/3-1, SIP/114SIP/115SIP/116SIP/117|20|tr) in new stack -- Called 114 -- Called 115 -- Called 116 -- Called 117 localhost*CLI -- SIP/117-0a0718d8 is ringing localhost*CLI -- SIP/116-0a06c398 is ringing localhost*CLI -- SIP/115-0a057678 is ringing localhost*CLI -- SIP/114-0a066c58 is ringing localhost*CLI -- Nobody picked up in 2 ms -- Executing BackGround(Zap/3-1, custom/no-answer) in new stack -- Playing 'custom/no-answer' (language 'en') localhost*CLI -- Got SIP response 400 Bad Request back from 192.168.1.216 localhost*CLI -- Timeout on Zap/3-1 == CDR updated on Zap/3-1 -- Executing Goto(Zap/3-1, daytime|s|1) in new stack -- Goto (daytime,s,1) -- Executing Set(Zap/3-1, TIMEOUT(response)=1) in new stack -- Response timeout set to 1 -- Executing Dial(Zap/3-1, SIP/114SIP/115SIP/116SIP/117|20|tr) in new stack -- Called 114 -- Called 115 -- Called 116 -- Called 117 localhost*CLI -- SIP/115-0a076e18 is ringing localhost*CLI -- SIP/116-0a07c358 is ringing localhost*CLI -- SIP/114-0a06c398 is ringing localhost*CLI -- SIP/117-0a081898 is ringing localhost*CLI -- Nobody picked up in 2 ms -- Executing BackGround(Zap/3-1, custom/no-answer) in new stack -- Playing 'custom/no-answer' (language 'en') localhost*CLI -- Got SIP response 400 Bad Request back from 192.168.1.216 localhost*CLI -- Timeout on Zap/3-1 == CDR updated on Zap/3-1 -- Executing Goto(Zap/3-1, daytime|s|1) in new stack -- Goto (daytime,s,1) -- Executing Set(Zap/3-1, TIMEOUT(response)=1) in new stack -- Response timeout set to 1 -- Executing Dial(Zap/3-1, SIP/114SIP/115SIP/116SIP/117|20|tr) in new stack -- Called 114 -- Called 115 -- Called 116 -- Called 117 localhost*CLI -- SIP/116-0a07c358 is ringing localhost*CLI -- SIP/117-0a0718d8 is ringing localhost*CLI -- SIP/115-0a057678 is ringing localhost*CLI -- SIP/114-0a066c58 is ringing localhost*CLI == Spawn extension
Re: [asterisk-users] Phantom Calls
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. There should be enough information to get some idea of why this is happening. -Original Message- 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 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 when they answer the phone, there is only silence and then they hang back up and it sometimes rings again. I've been trying to track this down for a couple of weeks now with no success yet. If anyone can lend a suggestion or a pointer to look for, I would greatly appreciate it. I've tried using WaitForRing() in case it is bad signaling coming from the phone company. But that has not helped. Thanks, I have done this yesterday by setting up putty to log to a file, but the customer employees have inadvertently shut it down on a couple of a occasions :) Hopefully it will be running when this happens again so I can try to track down the problem. This one is a sticky situation. This particular installation is for a friend of mine and his company. He decided to get a system from me instead of another friend of his that sells Panasonic or Avaya systems. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 channel back to the originator unless they receive an answer indication from asterisk, so even if you could place audio on the channel without an answer, there is no guarantee still it would propagate back to the originator of the call. Can't he just setup an extension to Answer() the call, play message or Ringing() and then transfer the call? -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 that
you drop your called party off to will play the sounds and do the
transfer. So really you need to concentrate on creating that context,
the .call files are very easy to generate.
Nitesh Divecha wrote:
Finally, this is what I was looking for... to generate a call.
I have been working on my Time Clock application, where an employee will
call into the system using his cellphone to clock in and clock out his
hours. And it works perfect...
Now I was looking for an option where or if an employee is late to clock
in, the system has to generate a call and call the supervisor and inform
him that XYZ employee is late and give an option to supervisor Would
you like to call XYZ employee, Press 1 and the system will call the XYZ
employee and connect with the supervisor...
Is it something feasible to do using the .call files? Or I am way too
off...
Cheers,
Nitesh
Christopher Robinson wrote:
I've done this many times, also used the .call files. If you don't need
your application to initiate the call the .call files are the better way
to go, otherwise it's a bit too much file management overhead.
Here's some working code on our end. In this case the Channel is
actually a context which makes the actual call, but I've used it both ways.
?php
require('PHPAGI/phpagi-asmanager.php');
$callid = 'Somebody';
$asm = new AGI_AsteriskManager();
if($asm-connect())
{
$call = $asm-send_request('Originate',
array('Channel'=LOCAL/[EMAIL PROTECTED],
'Context'='called_party_context',
'Exten'='899',
'Timeout' = '1000',
'Async'='1',
'MaxRetries' = '5',
'RetryTime' = '5',
'Priority'=1,
'Callerid'=$callid));
$asm-disconnect();
}
?
nik600 wrote:
hi
i'd like to write a simply application in php with phpAgi that:
- connect to Asterisk
- call an external number using a Zap channel
- play a message
here is some code:
?php
$asm = new AGI_AsteriskManager();
if ($asm-connect()) {
$asm-Originate(Zap/g1/1,number,default,1);
/*
play message...
*/
} else {
die(error\n);
}
?
But it doesn't work.
Is it possible to create a program like this?
thanks
Sorry, I can't help you with PHP. All my stuff is in pascal. But here
is a link to call origination info:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
I did something a bit like what you're doing, but it was a script to
call into the system and generate a broadcast type message to a
different party. Again, a bit different, but the elements are all the
same; call control, origination, database access, etc. Its in pascal,
but the syntax is very easy to understand and may give you an idea of
how program flow might be.
http://www.leebo.dreamhosters.com/apscripts/msgcast/
--
Warm Regards,
Lee
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[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 and it sometimes rings again. I've been trying to track this down for a couple of weeks now with no success yet. If anyone can lend a suggestion or a pointer to look for, I would greatly appreciate it. I've tried using WaitForRing() in case it is bad signaling coming from the phone company. But that has not helped. This setup has: Asterisk 1.2.17 Zaptel (whatever was distributed same time as Asterisk 1.2.17) CentOS 4.4 Polycom 301's throughout Sangoma A200 with 2 ports connected to PSTN. Thanks for any help. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 when
they answer the phone, there is only silence and then they hang back up
and it sometimes rings again.
I've been trying to track this down for a couple of weeks now with no
success yet. If anyone can lend a suggestion or a pointer to look for,
I would greatly appreciate it.
I've tried using WaitForRing() in case it is bad signaling coming from
the phone company. But that has not helped.
I highly doubt this is the cause. I suspect a dialplan error.
Ask your client if they are doing anything else when this happens, like
making a fax call ;)
This setup has:
Asterisk 1.2.17
Zaptel (whatever was distributed same time as Asterisk 1.2.17)
CentOS 4.4
Polycom 301's throughout
Sangoma A200 with 2 ports connected to PSTN.
Thanks for any help.
We kinda need to see your dialplan to provide any useful help.
-Stephen-
Thanks for responding, Stephen.
The client has a fax line, but it is separate line from the Asterisk
box. They have 3 lines coming in. 1 goes directly to fax machine and 2
go to Asterisk box.
I've searched the archives and phantom ringing comes up a few times
without any real resolutions that I can see.
The only thing very different about this installation is that the
customer has no intermediate IVR. They want the phones to ring directly
to a group and if no answer, then go to a mini IVR that asks if they
would like to hold longer or leave a message.
The device calling on CallerID when this happens is the default
CallerID set in sip.conf Device callernum which was never changed.
That was changed to a correct value. It appears as though the system is
calling itself.
extensions.conf:
[incoming]
exten=s,1,WaitForRing(5)
exten=s,n,Answer()
exten=s,n,Ringing()
exten=s,n,SetMusicOnHold(default)
exten=s,n,Wait(1)
exten=s,n,Goto(check_time,s,1)
[check_time]
exten=s,1,Answer()
exten=s,2,GotoIfTime(08:30-17:00,mon-fri,*,*,?daytime,s,1)
exten=s,3,Goto(after_hours,s,1)
[daytime]
exten=s,1,Answer()
exten=s,2,Set(TIMEOUT(response)=1)
exten=s,3,Dial(${GRP_All},20,tr)
exten=s,4,Background(custom/no-answer)
exten=1,1,Set(loops=0)
exten=1,2,Wait(0.5)
exten=1,3,Goto(ring-all-with-moh,s,1)
exten=2,1,Voicemail(b116)
exten=2,2,Hangup()
exten=5,1,Voicemail(b116)
exten=5,2,Hangup()
exten=555,1,VoicemailMain()
exten=i,1,Background(invalid)
exten=i,2,Goto(ring-all-with-moh,s,1)
exten=t,1,Set(loops=0)
exten=t,2,Wait(0.5)
exten=t,3,Goto(ring-all-with-moh,s,1)
[ring-all-with-moh]
exten=s,1,Answer()
exten=s,2,Noop(Loops are: ${loops})
exten=s,3,Macro(DialExtenNoVM,116|60|tm)
exten=s,4,Set(loops=$[${loops}+1])
exten=s,5,GotoIf($[${loops} = 2 ]?timeout-from-loop,s,1)
exten=s,6,Background(custom/no-answer)
exten=1,1,Goto(ring-all-with-moh,s,1)
exten=2,1,Voicemail(b116)
exten=2,2,Hangup()
exten=i,1,Playback(invalid)
exten=i,2,Goto(ring-all-with-moh,s,1)
exten=t,1,Goto(ring-all-with-moh,s,1)
sip.conf:
[general]
allowexternalinvites=yes
allowguest=no
autocreatepeer=no
autodomain=no
bindaddr=0.0.0.0
callerid=device callernum
canreinvite=no
checkmwi=30
compactheaders=no
context=incoming
defaultexpirey=120
dtmfmode=rfc2833
dumphistory=no
externrefresh=30
ignoreregexpire=no
insecure=no
maxexpirey=3600
musicclass=default
nat=no
notifyringing=yes
pedantic=no
progressinband=never
promiscredir=no
qualify=no
recordhistory=no
registerattempts=30
registertimeout=30
relaxdtmf=no
rtautoclear=no
rtcachefriends=no
rtpholdtimeout=600
rtpkeepalive=0
rtptimeout=3600
rtupdate=yes
sendrpid=no
sipdebug=no
srvlookup=no
tos=none
trustrpid=no
useclientcode=no
usereqphone=no
callevents=no
disallow=all
allow=ulaw
[116]
context=super-user
type=friend
canreinvite=no
dtmfmode=rfc2833
callerid=Barbara 116
nat=no
port=5060
qualify=no
secret=xxx
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=ulaw
[117]
context=super-user
type=friend
canreinvite=no
dtmfmode=rfc2833
callerid=Bill 117
nat=no
port=5060
qualify=no
secret=xxx
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=ulaw
[115]
context=super-user
type=friend
canreinvite=no
dtmfmode=rfc2833
callerid=George 115
nat=no
port=5060
qualify=no
secret=xxx
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=ulaw
[114]
context=super-user
type=friend
canreinvite=no
dtmfmode=rfc2833
callerid=Jack 114
nat=no
port=5060
qualify=no
secret=xxx
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=ulaw
--
Warm Regards,
Lee
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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] mailto:[EMAIL PROTECTED] wrote: We were having phantom calls as well. In our case, we had 2 pots line running in our sangoma card, and when you dial out, would would wait for whomever to pickup. If you gave up waiting an hung the phone up (we also had 2 normal phones plugged into fxs ports), it wouldn't immediately receive the hangup signal. The call would connect, then asterisk would turn around and try to call us back. If the other side hungup because they just heard dead error, then when you'd repickup your call, it would also be dead air. Not sure if this is the same case as yours, but ours was odd as well. Rob Thanks for responding, Guys. I can't say if that is the behavior that causes it to happen, but I have asked the customer to take note of that. The symptoms that you both describe are exactly what they are experiencing so this is a welcome lead. Here is the zapata.conf below. [channels] usecallerid=yes cidsignalling=bell cidstart=ring usecallingpres=no echocancel=yes echocanclewhenbridged=yes echotraining=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 hanguponpolarityswitch=yes answeronpolarityswitch=no ringtimeout=8000 musiconhold=default busydetect=yes busycount=6 usecallerid=yes hidcallerid=no callwaiting=no threewaycalling=no transfer=no echocancel=yes echotraining=yes echocanelwhenbriged=yes context=incoming callprogress=no answeronpolarityswitch=no signalling=fxs_ks channel= 3 busydetect=yes busycount=6 usecallerid=yes hidcallerid=no callwaiting=no threewaycalling=no transfer=no echocancel=yes echotraining=yes echocanelwhenbriged=yes context=incoming callprogress=no answeronpolarityswitch=no signalling=fxs_ks channel= 4 -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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/gmane.comp.telephony.pbx.asterisk.user/185537)
I would like to do this, but I am having trouble figuring out how. I have
tried the following but it is not working for me:
***test.php***
#!/usr/bin/php -q
?php
require_once('phpagi.php');
$agi = new AGI();
$dialstr=IAX2/wayne/[EMAIL PROTECTED];
$agi-SetVar(JAMES,$dialstr);
exit(0);
?
***extensions.conf***
[from-sip]
exten = 111,1,DeadAGI(test.php)
exten = 111,2,Dial(${JAMES})
exten = 111,3,Hangup
Thanks in advance for any help.
James
Checkout the h hangup extension:
http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension
--
Warm Regards,
Lee
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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 having trouble understanding the following xml attribute: voIpProt.SIP.requestValidation.x.request.y.event I understand what it does (at least conceptually) but ss the x variable still referring to a server (1 or 2)? And the y var, what is it referring to? An event? Which one? Determines which events specified with the Event header should be validated; only applicable when voIp- Prot.SIP.requestValidation.x.request is set to “SUBSCRIBE” or “NOTIFY”. If set to Null, all events will be validated. Please excuse me if it's an obvious question. Just as a breadcrumb, here is what Polycom support says of the this portion of the XML config: The voIpProt.SIP.requestValidation parameter is used for validation. The validation is used for security purposes. To set it up properly, you have the following parameters involved which compose the overall validation feature. The parameters involved are: voIpProt.SIP.requestValidation.x.request voIpProt.SIP.requestValidation.x.method voIpProt.SIP.requestValidation.x.request.y.event voIpProt.SIP.requestValidation.digest.realm Explanation: voIpProt.SIP.requestValidation.x.request With this parameter, you can specify which methods you want the phone to validate. The list of methods allowed as values are listed in the Admin Guide. Ex: if you wanted to use validation against all INVITES, this parameter would look like this voIpProt.SIP.requestValidation.INVITE.request voIpProt.SIP.requestValidation.x.method This parameter defines the method of validation to be used. The list of methods allowed as values are listed in the Admin Guide. The three methods are source, digest or both. If you wanted to use source as the method, the parameter would like this voIpProt.SIP.requestValidation.source.method. This means that when the phone is using voIpProt.SIP.requestValidation.INVITE.request it will apply voIpProt.SIP.requestValidation.source.method and validate that the INVITE is coming from the IP address specified on its line registration. voIpProt.SIP.requestValidation.x.request.y.event This parameter is only used when you specify voIpProt.SIP.requestValidation.x.request to be voIpProt.SIP.requestValidation.SUBSCRIBE.request or voIpProt.SIP.requestValidation.NOTIFY.request. This parameter will also do validation based on the method used on voIpProt.SIP.requestValidation.x.method against the EVENTS within a “SUBSCRIBE” or “NOTIFY”. Since the RFC for SIP may have different events, the list is not provided in the admin guide. For an updated list of EVENTS please check the RFC. A less updated list of EVENTS used within a NOTIFY is as follows: conference dialog message-summary presence refer reg winfo voIpProt.SIP.requestValidation.digest.realm In this parameter you can specify a string which you have also specified on your server. The value can be any valid string. Once the phone advertises the string, the server will match it against is list of allowed users challenging the phone for an user name and password using authentication digest. Generally, this string contains the name of the host performing the authentication. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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
; **
exten = 734xxx,1,Gosub(check-blacklist,s,1)
exten = 734xxx,n,NoOP(Caller not blacklisted)
exten = 734xxx,n,Set(CALLERID(number)=91${CALLERIDNUM})
exten = 734xxx,n,Goto(auto_attend,s,1)
[check-blacklist]
exten = s,1,MYSQL(Connect connid localhost anonymous '' blacklisted)
exten = s,2,MYSQL(Query resultid ${connid} SELECT flag FROM
BlackNumbers WHERE phone = ${CALLERIDNUM})
exten = s,3,MYSQL(Fetch fetchid ${resultid} results)
exten = s,4,MYSQL(Disconnect ${connid})
exten = s,5,MYSQL(Clear ${resultid})
exten = s,6,Set(BLACKLISTED=${results})
exten = s,7,GotoIf($[${BLACKLISTED} = YES]?blacklisted,s,1)
exten = s,8,Return
[blacklisted]
exten = s,1,NoOP(Caller: ${CALLERIDNUM} is on the black list)
exten = s,n,SetCDRUserField(Blacklisted)
exten = s,n,Set(PRI_CAUSE=17)
exten = s,n,Hangup()
I also wrote a pascal based script for doing the same thing:
http://www.leebo.dreamhosters.com/apscripts/calldirect/
--
Warm Regards,
Lee
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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 program!! What language are you developing it in? If it's PHP, I could help test or develop... -Kenneth Its something I'm doing in my spare time. Sorry, I'm writing it in Freepascal/Lazarus (we are primarily Delphi/Freepascal shop here) but you're welcome to get yo pascal on if you like ;) At any rate, we'll host it initially on subversion. It will be released under LGPL, I think. I will post updates to its status to the .biz list which I think would be more appropriate. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] agi with java?
Lenz wrote: Hi Lee, we are a Java shop and our experience with Java has been much the one you say - it does scale pretty well and it is very solid. What I was trying to say is that Java is not very well suited to the classic, Unix-style, fire-up-process-and-let-it-die that goes for CGI/AGI 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 on the same machine as Asterisk yields better performance than launching separate exe processes through the dial plan. Completely anecdotal of course. This is careful research conducted over our entire 5 customer base... I get what you are saying, I was agreeing with you. :) We *were* writing all of our AGI as binary executables and even then, the FastAGI server that we eventually built still gets better performance vs. when we launched separate AGI per call from the dialplan. My guess is that it is easier on the system for an existing executable (FastAGI Server) to spawn threads of execution for short periods of time to handle (Fasg)AGI requests than it is to run separate executable AGI's instead. We're hoping that performance will be improved even more when we introduce pooling of common objects (db access for example). -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 http://mocker.org/papers/. I welcome any comments or corrections to improve upon it. As I said, it was mainly done to force myself to research the topic so there are probably errors! :) -- Kyle Sexton Kyle, I liked it. Maybe you could also cover how the initial vars are pushed to the application one right after another initially and to look for an empty line to indicate end of initial vars coming in. Have you considered putting it on the wiki? That would be an ideal place for a nice white paper like that, IMO. Once google indexes it, it should be fairly easy to find for new Asterlings... -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
