[asterisk-users] Compiling Zaptel 1.2.10 on Ubuntu 6.10
Here's a weird problem that I'm not quite sure how to resolve. Zaptel 1.2.10 compiles just fine with make, but when make install is run, this happens: [ `id -u` = 0 ] /sbin/depmod -a 2.6.17-10-generic || : [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf build_tools/genmodconf linux26 tor2 torisa wcusb wcfxo wctdm wctdm24xxp ztdynamic ztd-eth wct1xxp wcte11xp pciradio ztd-loc ztdummy [: 66: ==: unexpected operator [: 66: ==: unexpected operator Unknown kernel build version requested... exiting. make: *** [install] Error 1 This worked just fine under ubuntu 6.06 with the same set of packages installed. Any help is appreciated. -- Strom Carlson http://www.stromcarlson.com/ ___ --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] 911 versus 9.911
I play a recording that starts as soon as the second 1 is pressed: If this is an emergency, please hang up and dial 9-911. Short, simple, and to the point. -- Strom Carlson http://www.stromcarlson.com/ ___ --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 records nonsense, but record() works (??)
-Original Message- From: Stefan-Michael. Guenther (in-put GbR) [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: 6/10/06 8:47 AM Subject: [Asterisk-Users] Voicemail records nonsense, but record() works (??) Hello, I have setup an Asterisk 1.2.7.1 system, with a working voicemail box: /etc/asterisk/extensions.conf exten = 83086921,1,Answer exten = 83086921,2,Dial(SIP/stefan,5,r) exten = 83086921,3,VoiceMail,u111 exten = 83086921,4,Hangup exten = 83086921,103,VoiceMail,b111 exten = 83086921,104,Hangup /etc/asterisk/voicemail.conf [default] language=de 111 = 111,Mailbox 111,[EMAIL PROTECTED] The mailbox starts, I hear the intro and speak my message. In the CLI I can see that the message has been recorded and I get the recorded message via mail. But when I listen to the recorded messages or call the mailbox, I either hear nothing or just a short cracking sound. At least the length of the message is correct. If have tried to record the message with gsm, wav or wav49, the result is always the same. When I use the record() application to record a gsm file, everything is okay. I obviously made something wrong when configuring the voicemail system. Can someone give me a hint what's going wrong? Thanks for your help, stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --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
Re: [Asterisk-Users] Zap channels ringing too loudly
On 5/31/06, BJ Weschke [EMAIL PROTECTED] wrote: I could be way off here, but I thought FXS ringing was signaled only by a change in voltage on the pair, so I'm not sure how zaptel could instruct the hardware device to send a different voltage? I think its only capability with FXS is to fluctuate the voltage to support distinctive rings. Ring voltage in North America is supposed to be 90vAC at 20Hz. Assuming these are Western Electric 2500 sets or similar, then a less-wimpy ring voltage generator could very well make the phones ring louder. Fortunately, if these are Western Electric sets, then there should be a dial marked LOUD or HI on the underside of the phone. Turn that to the right to make the bells softer. -- Strom Carlson http://www.stromcarlson.com/ ___ --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] End of migration: adding support for some analog phones
On 5/26/06, Mimmus [EMAIL PROTECTED] wrote: Hi, during gradual migration to Asterisk, I put Asterisk in front of a legacy Alcatel PBX: PRI PSTN -- Asterisk -- E1 cable -- Alcatel PBX After successful deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I use a TDM2400P board together with the actual TE410P? I've had very good results at one of my clients' locations using the following setup: PRI -- Asterisk w/Digium TE406P -- Adtran channel bank -- Fax machine Assuming you have a spare span on your E1 card, it's probably worth it to get a channel bank with 24 FXS ports and put it in your wiring closet. Even if you don't use all the ports now, you'll be a simple cross-connect away from adding an analog station if and when it's needed in the future. -- Strom Carlson http://www.stromcarlson.com/ ___ --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] No rings before auto attendant
On 5/26/06, Dan Elder [EMAIL PROTECTED] wrote: Thanks, will try this... I actually don't really want to delay incoming calls before the attendant, but it seems to take about 7-10 seconds from the time I dial until the AA picks up, without a ring, it just sounds odd, like the call didn't go through...so I wanted to experiment with trying to add some kind of ringing sound...we'll see if this is actually a good idea or not when I mod this tonight. Out of random curiosity, is it a channelized T1, or is it a PRI? -- Strom Carlson http://www.stromcarlson.com/ ___ --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] large duration calls
On 5/26/06, Francisco Seratti [EMAIL PROTECTED] wrote: Hello mates, im having calls of about 120 o 130 minutes in my accounting DB but these are calls not made by users. I guess my asterisk is not catching some BYE requests and after some timeout it hangs up the call. Is this issue known? Is there a way to trace this problem? Im using Asterisk 1.0.10 on a i686 linux and mysql for the accounting. I have this problem with different protocols, like SIP and Zap. If you need another specs or something, askme. Cheers, Francisco. Well, first off, I'd suggest that you upgrade to the latest stable version...1.0.10 is comparatively ancient. -- Strom Carlson http://www.stromcarlson.com/ ___ --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] No rings before auto attendant
On 5/25/06, Dan Elder [EMAIL PROTECTED] wrote: Hi all, been searching not finding an answer to this, although I'm guessing it's absurdly simple... I just hooked up a T1 to our * box (1.2.0), which had been using POTS lines via a channel bank.. Now when I call the new T1 circuit, there are no rings, the Autoattendant just picks up right away.. Any clue on how to make it ring twice before getting picked up? I tried immedate=no and some other zapata.conf tweaks, but nothing seems to work. I also tinkered with adding some wait statements before the 'answer' but only heard silence then the attendant..sorry for such a basic question.. I guess I'm just not punching in the right search terms in my queries. Thx in advance You need to provide audible ring: exten = 2368,1,Ringing exten = 2368,2,Wait(11) exten = 2368,3,Answer and so on. Of course, if you're on a T1, why would you want to artificially delay the calling party's access to the auto-attendant? -- Strom Carlson http://www.stromcarlson.com/ ___ --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] PRI dialing IVR with inband DTMF
On 5/19/06, Anthony Cennami [EMAIL PROTECTED] wrote: Two situations: Shoretel Phone -- ShoreTel -- PRI -- Asterisk -- Softswitch -- VoIP/PSTN -- This situation does NOT work. User hears audio, but Asterisk does not appear to process DTMF. Note, this is ONLY on IVR applications where we are not getting 200/connect passed through even though we're hearing IVR audio (early media) Is the PRI connected directly from the ShoreTel to the Asterisk box, or is it connected through the PSTN? The PSTN is not supposed to set up the forward audio path until after the call supervises, so this is where your issue may lie. -- Strom Carlson http://www.stromcarlson.com/ ___ --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] Plan to free myself from AAH
On 5/17/06, Mimmus [EMAIL PROTECTED] wrote: I was thinking to this plan: - install another server with Red Hat 4 U3 - install PHP, MySQL and other usefuls stuffs - download latest version of Asterisk and third parts applications I use - compile all - copy /etc/asterisk from old server to new, change only what is needed - start and try Do you think is it OK? I doubt it. The problem I have with AAH / AMP / FreePBX is that the configuration files are absolutely full of useless garbage and are really not at all suitable for moving to a standard asterisk install. Set up a new server from scratch and start learning how to configure asterisk manually. Rebuild everything one step at a time so that the functionality remains as you'd like it to be, but that the actual configs aren't full of that FreePBX garbage :) -- Strom Carlson http://www.stromcarlson.com/ ___ --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] Plan to free myself from AAH
On 5/17/06, David K Parker [EMAIL PROTECTED] wrote: I wouldn't knock the third party friendly interfaces to Asterisk too hard. They will evolve and improve over time. The adoption of Asterisk as a mainstream PBX is dependent upon a user friendly interface. Well, as soon as a GUI shows up that doesn't make configuring Asterisk like trying to sew with boxing gloves on, I'll give it a good, hard, unbiased look. For now, though, the available interfaces are really just not there yet - they don't allow enough flexibility and they are very easy to outgrow. -- Strom Carlson http://www.stromcarlson.com/ ___ --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] Asterisk X100P - Interrupt a call?
On 5/15/06, Corey Frang [EMAIL PROTECTED] wrote: So, We want to be able to put a fax machine on the line port of the X100P in our asterisk server. We however also want to use this card for 911 calling. We need some sort of mechanisim to disable the line out port on the x100p by software to interrupt a call on the line. Put the fax machine on an FXS port, have faxes route through Asterisk's switching matrix, then have your dialplan tear that channel down if 911 is dialed. And also, pray that your echo cancellers work really well. -- Strom Carlson http://www.stromcarlson.com/ ___ --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] Text Messaging or AIM
On Sun, 13 Mar 2005 16:13:04 -0600, Anton Krall [EMAIL PROTECTED] wrote: I already have this workling for remote linux admin. For example, each linux box has it MSN user and I have them on ly MSN list. So if I need to reset a server, I just send an IM via MSN to the user with the keyword reboot and the server runs the command. If you need any help on setting this up, let me know. Anton Krall WOW! I have never seen a finer example of security in all my years of working with computers. -- Strom Carlson http://www.stromcarlson.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: * compatible with Pulse dialing phones ?
On Tue, 18 Jan 2005 10:01:30 -0700, Steve Murphy [EMAIL PROTECTED] wrote: On Tue, 2005-01-18 at 09:49 -0600, [EMAIL PROTECTED] wrote: I'd like to know if is is possible to connect a very old phone to asterisk and dial pulses with it? Arnaud-- I'm answering this via the users' mailing list instead of the developer's. I know they'd appreciate that, and it's a better forum for these sorts of questions. I'm using an old rotary dial phone connected to one of the 4 ports of the 4-port FXS card; it works great. I can dial, check voicemail, etc by spinning the dial and letting go. I was amazed, but pulse dialing works fine with the FXS card. The only trouble might be with # and * key functionality, I haven't investigated workarounds. Eleven pulses on a zaptel card will give you an asterisk, and twelve will generate an octothorp. Get your hookswitch-flashing finger ready... -- Strom Carlson http://www.stromcarlson.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] only allow long distance calls to countries x, y, and z
On Mon, 13 Dec 2004 07:22:27 -0800 (PST), Thomas Miller [EMAIL PROTECTED] wrote: Can somebody suggest the easiest way to only allow outgoing long distance calls to countries x, y, and z? Set it up that way in your dialplan...instead of matching _011., match _01144. for UK, _01130. for Greece, and so on. -- Strom Carlson http://www.stromcarlson.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P FXS polarity reversal?
Hello all, Is it possible to do a talk battery polarity reversal on a TDM400P FXS interface? Everything I can find seems to be referring to the procedure for detecting a battery reversal on a telephone company POTS line using the FXO interface, but not for actually generating one back to a station upon answer supervision. I would assume that VoicePulse and VoipJet provide a way of signaling far-end supervision back to the originating Asterisk PBX... Basically, my two questions are: (1) Is the hardware capable of even performing a reversal? (2) If the above is true, how would you make it happen in Asterisk? -- Strom Carlson http://www.stromcarlson.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
