[asterisk-users] Fwd: Legacy TDM400
Hello all, It's been quite some number of years since I played around with Asterisk and I'm just now getting back into it. I think the last version I worked with was 1.8. I have a legacy Digium TDM400 PCI card and am wondering if that will still work on newer versions of Asterisk. My initial attempts on Debian 10 and the Debian repository version of Asterisk didn't get me very far. Any pointers would be appreciated. -Roy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dial multiple extensions
Hello, I've got a problem with something I'm doing and can't seem to figure it out. I've tried different suggestions I've found on voip-info.org as well as other sites but nothing I do seems to work. I've got an older Digium TDM400P. The FXO daughter card is connected to my POTS line and the FXS daughter card is connected to a TDM phone. I also have multiple SIP extensions. My desire is to ring all the internal extensions (the TDM and SIP extensions) on an inbound call and send the call to whichever extension picks up first. This seems to be working just fine if the extension that picks up is one of the SIP phones. On the other hand, if the extension that picks up is the one off the FXS port, then the SIP phones continue to ring and the dial plan continues to execute even though the caller on the FXO port has been connected to the phone on the FXS port. Inbound calls are sent to extension 3100, which looks like this: exten = 3100,1,Dial(SIP/3105SIP/3106SIP/3108dahdi/1,20,tr) exten = 3100,n,Voicemail(3100) exten = 3100,n(end),Hangup() Like I said, if I pick up on one of the SIP extensions, it seems to do exactly as I expect. If I pick up on dahdi/1, however, the SIP phones continue to ring and the FXO and FXS ports are connected and passed into voicemail. I'm running on Debian Squeeze/6.0.1 and am running the stock Asterisk 1.6.2.9-2+squeeze2 package. If anyone has some suggestions, I'd be happy to hear them. Thanks! Roy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial out from AGI
I'm writing an AGI script and want it to dial a number on a channel
connected to the PSTN. It would look something like this (pseudo-code
follows):
if ($a){
dial(8005551212);
}else{
dial(866555);
}
The part I can't seem to get right is the dial function. I tried to
mimic the dial plan like so
sub dial($number){
print Dial(\Zap/1-1\, \Zap/g2/$number\)\n;
}
but I get the error
handle_exec: Could not find application (Dial(Zap/1-1,Zap/g2/866555)
Anyone have any suggestions?
Thanks,
Roy
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Re: [asterisk-users] extension launch into AGI
Time Bandit wrote: I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card connected to a POTS line and a phone set (physical extension). I've got all incoming calls launching directly into an AGI script. I'd like to do the same for the physical extension. In other words, when picking up the hand set, the AGI is launched without dialing any digits. check http://www.voip-info.org/wiki-Asterisk+config+zapata.conf keyword is : immediate Perfect. Thank you. ___ --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] extension launch into AGI
I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card connected to a POTS line and a phone set (physical extension). I've got all incoming calls launching directly into an AGI script. I'd like to do the same for the physical extension. In other words, when picking up the hand set, the AGI is launched without dialing any digits. Anyone have any ideas for me to try? Thanks in advance, Roy ___ --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] Announce caller-id
I did something along these lines, but I was playing the caller ID back to the caller, not after a transfer. In a perl AGI script. I split the caller ID number into an array, seperated by '//' so each number was an element. Then I played digits/$array[0]... digits/$array[1]...etc. coolbreeze wrote: I would like to transfer an incoming call and, when the call is answered, have the caller id of the call spoken when the call is answered on my cell phone. Any tips greatly appreciated AAH 2.7 using sip trunks exclusively. Regards ___ --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 ___ --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] AGI transfer question
I'm looking for some guidance on setting up an extension (associated with a physical port) properly and being able to transfer calls in an AGI script. I started out my home brew system with just an fxo port attached to the PSTN. Now, I've added a fxs port and I'd like to hang a phone off of it (TDM411). My current extensions.conf looks like this: exten = s,1,Wait,1 exten = s,2,Answer (and so on) and it works just fine picking up an incoming call and launching into my AGI script. But I'd like to expand the system to be able to transfer calls between the fxo and fxs ports in an AGI script. Basically, a very simply PBX. So, an incoming call on the fxo could be sent to the fxs. And an outbound call when the phone on the fxs port is picked up, would be able to dial 9 to get out. Thanks in advance, Roy ___ --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] Help with perl AGI script
Russell Bryant wrote: On Sat, 2006-08-05 at 20:43 -0400, Roy Kidder wrote: Is there some way I can better control the execution of playbacks so that they take place as I expect them to? Yes, your script needs to read a line of input from stdin to wait for Asterisk to send back the result code indicating that the application execution is complete. You'll get nack something like 200 result=0. I tried it again, reading a single line from stdin and got the 200 result=0 message. Is there potential for there to be other messages? i.d. 200 result=1 or 404 file not found? Also, is there always going to be a single line from stdin, or should I loop until I find the line I'm looking for? Thanks again, Roy ___ --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] Help with perl AGI script
I'm new to Asterisk and am trying to write an AGI script in perl and need some pointers. The script simply plays a few gsm files in succession before doing a database insert (using perl's DBI in a sub). In a nutshell, it looks like this: print EXEC Playback foo1\n; print EXEC Playback foo2\n; print EXEC Playback foo3\n; print HANGUP\n; dbinsert(); The problem I'm seeing is that the call to the sub dbinsert is taking place before the audio of foo2 even starts (not to mention foo3 or the HANGUP). I'm assuming that the EXEC Playback forks and plays the audio file when the channel is clear, allowing the script to continue. Is there some way I can better control the execution of playbacks so that they take place as I expect them to? Thanks in advance, Roy ___ --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] voicemail/privacy system
Moises Silva wrote: Yep, perfectly possible. I would do that with AGI and php, in your case, perl works as well. The only thing you need is read documentation regarding AGI, Voicemail and extensions. Its kind of difficult to helo you further if you dont tell us how much you know about contexts, extensions etc. But in general you will I've read The ASterisk Handbook Verson 2, so I have a very basic understanding of contexts, extensions and so forth. After reading more on AGI, it looks like I could do everything I want to with a rather simple Asterisk config and a perl script. The privacy manager function could be accomplished by looking for an empty (or missing) agi_callerid: value from STDIN. And the individual voicemail boxes could be accomplished by playing a prompt, waiting for a digit and using set_extension to send the call to the desired voicemail box. Does that sound like I'm on the right track? Thanks in advance, Roy ___ --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] voicemail/privacy system
Hello all. I'm relatively new to Asterisk, and before I get too involved in it, I want to find out if it will do what I'd like it to do (I'm relatively sure it can). In short, my goal is to set up a voicemail system and privacy manager for my home. For my proof of concept, I have a single port FXO card attached to a single POTS line. Currently, the FXO card sits in parallel to my phone. If I can get the answering/routing working the way I want, I'd upgrade that to a TDM11B and put my phone on the FXS side, completely hidden from the telco network. The privacy manager part is similar to the residental product many telcos are offering these days: when a call comes in with no caller-id, play a message stating that such calls aren't accepted and hang up. The voicemail part is also similar to what telcos are offering. When an incoming call (with caller-id, of course) comes in and isn't answered, it's routed to an attendant that allows the caller to press 1 to leave a message for Joe or 2 to leave a message for Jane, etc. So basically, the end product would be a single incoming line with a single physical extension and multiple virtual extensions (the voicemail boxes). Another feature I'd be interested in is being able to gathering up those voicemail messages, converting them to MP3 (if not already converted), and emailing them to the recipient (instead of leaving them in the voicemail system). I'm probably write that in perl with a mysql backend if there isn't already a tool out there for that. I currently have the kernel modules zaptel and wcfxo working and recognizing my card, as well as asterisk answering incoming calls and playing the demo, but I have no idea where to go from there. Any help would be appreciated. Thanks, Roy ___ --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
