Re: [Asterisk-Users] multiple days on a GotoIfTime command?
The problem is, there is no pattern. It´s not an open/close scenario. This month I need to call NUMBER1, NUMBER2 and NUMBER3 on those days. Next month, who knows? I´ll receive another schedule to implement on asterisk. I see no way to avoid changing those lines each month. What I´m trying to do is reduce the number os files involved. Gelson brian wrote: I see the pattern.. let me think for a second.. and I'm sure I can get you something that's simpler than 31 gotoif's bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of brian Sent: Tuesday, July 06, 2004 5:24 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] multiple days on a GotoIfTime command? You're making this WAY too complicated its simpler than you can even imagine. Mind answering my original question first? WHAT THE HECK is the pattern your logic? What times are you open.. what times are you closed? What? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roger Gulbranson Sent: Tuesday, July 06, 2004 4:20 PM To: [EMAIL PROTECTED] Cc: Roger Gulbranson Subject: Re: [Asterisk-Users] multiple days on a GotoIfTime command? On Tue, 2004-07-06 at 17:03, Gelson Dias Santos wrote: brian wrote: What are you trying to do? What is the end result and what hours are you open? Exactly what I said. Need to call a number if time and day matches what is on the rule. This month I have to: call NUMBER1 if day = 1,2,3,4,5,8,14,17,18,20,23,26,29 call NUMBER2 if day = 6,9,10,11,12,15,21,27,30,31 call NUMBER3 if day = 7,13,16,19,22,24,25,28 I have it working now using 31 GotoIfTime lines, one for each day of month but I would like to optimize it. If I could group all days related to a number somehow, I would end up with just three GotoIfTime lines. You are making this way too complicated. Use DBget to retrieve a number which is the extension you want and then dial that extension. Have a cron job (or something similar) set the extension you want via DBset. You can put all of your time logic into the cron job. There may be even simpler solutions. ___ 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 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 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 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
Re: [Asterisk-Users] Unreliable dtmf digit generation from tdm400p
I have two TMD400P with 4 FXO modules each working fine for about one week. Never experienced this problem. Are you using the power connector on each board? When testing I discovered it works without the power cord, but I assume that if it´s there then I need to connect it. :-) My only problem with these boards is the allways green status I reported on another email. If a line stops working, asterisk can´t senses it and still tries to dial outside through that line. Gelson Jean-Yves Avenard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I've had this issue too with a TDM4 cards with 3 FXS modules Sometimes I dial a number, and it gets somewhere else. Apologies to the person I called, press redial, hope we go. Quite annoying, it doesn't seem to happen very often though Jean-Yves On 07/07/2004, at 9:39 PM, Andrew Yager wrote: I'm not having this problem on either of my TDM400 cards with a mix of FXO and FXS modules - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA6/+8XeDVKqIr3GURApo6AJ45/sHGSxsu7WZv9kU6B+rXbHhCcgCffRiZ /5bRz3sRlNTdjKFAINbUxQE= =WSH3 -END PGP SIGNATURE- ___ 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 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
Re: [Asterisk-Users] Hangup's not detected correctly
Steven Critchfield wrote: On Tue, 2004-07-06 at 17:52, Ruben Fagundo wrote: I have an easy question. I setup Asterisk with a TDM400 w/ 4FXO ports and I have the following problem. Yep, so easy it seems to be covered almost weekly here because no one looks up any of the information already provided to them. Not quite easy. I agree its asked about once a week, but they get no solution. Callprogress does not work at all outside US, because it´s just a hack. Busydetect sometimes work, sometimes doesn´t and sometimes drops calls in the middle. I have busydetect=yes and busycount=15 and I still have dropping calls and no hangup detections on a daily basis. I also played with BUSYDETECT_MARTIN and/or BUSYDETECT_TONEONLY and it makes no difference. I also tried editing dsp.c and adjusting BUSY_MIN and BUSY_MAX, but nothing fixes these problems. Gelson A call comes in correctly. The callers dials extension 100 (grandstream SIP phone). The caller then hangup, before the call goes to voice mail, however, the phone continues to ring, then goes to voicemail, and leaves an empty vmail message, long after the caller has hung up. Is there a way I can correct for this, ie, have the system detect hangups correctly ? On analog... callprogress and/or busydetect. Better yet, get disconnect supervision if offered, ___ 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] multiple days on a GotoIfTime command?
I´m trying to setup a dial rule where I need to evaluate the day of month. Here is an example: exten = 4,1,GotoIfTime(16:01-07:59|*|14,17,18,20,23,26,29|jul?6) I found it doesn´t work. Is it possible to specify more than one day on the same line, or do I need to include one line for each day? I known I can use ranges but even then I´ll end up with around 25 lines for each month. I´m trying to simplify maintenance of this rules, because I´ll have to change it each month. Thanks for any tip/suggestion. Gelson ___ 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
Re: [Asterisk-Users] multiple days on a GotoIfTime command?
brian wrote: What are you trying to do? What is the end result and what hours are you open? Exactly what I said. Need to call a number if time and day matches what is on the rule. This month I have to: call NUMBER1 if day = 1,2,3,4,5,8,14,17,18,20,23,26,29 call NUMBER2 if day = 6,9,10,11,12,15,21,27,30,31 call NUMBER3 if day = 7,13,16,19,22,24,25,28 I have it working now using 31 GotoIfTime lines, one for each day of month but I would like to optimize it. If I could group all days related to a number somehow, I would end up with just three GotoIfTime lines. Gelson bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Gelson Dias Santos Sent: Tuesday, July 06, 2004 3:19 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] multiple days on a GotoIfTime command? I´m trying to setup a dial rule where I need to evaluate the day of month. Here is an example: exten = 4,1,GotoIfTime(16:01-07:59|*|14,17,18,20,23,26,29|jul?6) I found it doesn´t work. Is it possible to specify more than one day on the same line, or do I need to include one line for each day? I known I can use ranges but even then I´ll end up with around 25 lines for each month. I´m trying to simplify maintenance of this rules, because I´ll have to change it each month. Thanks for any tip/suggestion. Gelson ___ 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 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 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] No RED/GREEN alerts on TDM400P?
I replaced my X100P cards with two TDM04B fully populated (8 FXO modules). They are working fine, I can make and receive calls, but I noticed all modules are always in GREEN state, even if I disconnect the line. Both zttools and a cat /proc/zaptel/device shows no RED alarm. Is there a workaround for this? Gelson ___ 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
Re: [Asterisk-Users] NetworkWorld article on Open Source Telephony
Brent Franks wrote: Aside from echo issues that seem to be apparent with everyone occasionally (by everyone, those not running hardware T1 echo cans) I believe * is ready for the prime time. Integrators however should have a better I add to the list: hangup detection on FXO interfaces is terrible. The busy tone detection routines does not work right/reliably, at least not for those outside USA. Same thing about callprogress, the lack of support for DTMF CallerID and no R2 signaling. Without these features * will never be a serious option outside US. Gelson ___ 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
Re: [Asterisk-Users] Zapata FXO always answers call?
Reid A. Forrest wrote:
Ummm X100P IS an analog adaptor.
Well, I thought he was talking about some of the ATA´s available on the
market. If an X100P is one analog adapter and has this problem, then
everything else has this problem too (except E1/T1/PRI boards that have
it´s own signaling).
Ok, so how you guys set up voicemail or anything else that needs more
than one priority level? I can´t believe nobody has voicemail on analog
lines!! That would be a major flaw in asterisk.
As I and others have said, callprogress is experimental and of not much
use. For me it does nothing.
Gelson
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gelson Dias
Santos
Sent: Monday, June 07, 2004 4:32 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Zapata FXO always answers call?
Steven Critchfield wrote:
your using an analog adapter. We cover this regularly. You need
callprogress detection since you don't have a reliable way of doing it
via the analog adapter.
No, I´m not using an analog adapter. As I said, my X100P connects
directly to my PBX.
Gelson
On Wed, 2004-06-02 at 14:09, Gelson Dias Santos wrote:
I have some X100P connected to my analog PBX. When I want to call an
analog extension on that PBX I use the following rule:
exten = _21XX,1,Dial(Zap/g1/${EXTEN:2},20)
where 21 is just a prefix to indicate it´s an analog extension and XX
matches the real two digit extension number. (this is why I strip of two
digits when dialing Zap/g1. Well, everything works fine, except that *
says on the log that the call was answered, even if it´s still ringing.
The problem is that now I want to set up voicemail to those analog
extensions, but since * says it answered on first ring it never goes
to the next priority, where voicemail is called.
I tried callprogress=yes on zapata.conf but it has no effect. Here is
a
tipical log from a call I have _not_ answered:
-- Executing Dial(SIP/2000-a638, Zap/g1/32|20) in new stack
-- Called g1/32
-- Zap/1-1 answered SIP/2000-a638
I´m runnign 0.9.0 from the tar archive. Is this a known bug? Is there
a
workaround for it?
Gelson
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Re: [Asterisk-Users] Zapata FXO always answers call?
Steven Critchfield wrote:
your using an analog adapter. We cover this regularly. You need
callprogress detection since you don't have a reliable way of doing it
via the analog adapter.
No, I´m not using an analog adapter. As I said, my X100P connects
directly to my PBX.
Gelson
On Wed, 2004-06-02 at 14:09, Gelson Dias Santos wrote:
I have some X100P connected to my analog PBX. When I want to call an
analog extension on that PBX I use the following rule:
exten = _21XX,1,Dial(Zap/g1/${EXTEN:2},20)
where 21 is just a prefix to indicate it´s an analog extension and XX
matches the real two digit extension number. (this is why I strip of two
digits when dialing Zap/g1. Well, everything works fine, except that *
says on the log that the call was answered, even if it´s still ringing.
The problem is that now I want to set up voicemail to those analog
extensions, but since * says it answered on first ring it never goes
to the next priority, where voicemail is called.
I tried callprogress=yes on zapata.conf but it has no effect. Here is a
tipical log from a call I have _not_ answered:
-- Executing Dial(SIP/2000-a638, Zap/g1/32|20) in new stack
-- Called g1/32
-- Zap/1-1 answered SIP/2000-a638
I´m runnign 0.9.0 from the tar archive. Is this a known bug? Is there a
workaround for it?
Gelson
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[Asterisk-Users] Zapata FXO always answers call?
I have some X100P connected to my analog PBX. When I want to call an
analog extension on that PBX I use the following rule:
exten = _21XX,1,Dial(Zap/g1/${EXTEN:2},20)
where 21 is just a prefix to indicate it´s an analog extension and XX
matches the real two digit extension number. (this is why I strip of two
digits when dialing Zap/g1. Well, everything works fine, except that *
says on the log that the call was answered, even if it´s still ringing.
The problem is that now I want to set up voicemail to those analog
extensions, but since * says it answered on first ring it never goes
to the next priority, where voicemail is called.
I tried callprogress=yes on zapata.conf but it has no effect. Here is a
tipical log from a call I have _not_ answered:
-- Executing Dial(SIP/2000-a638, Zap/g1/32|20) in new stack
-- Called g1/32
-- Zap/1-1 answered SIP/2000-a638
I´m runnign 0.9.0 from the tar archive. Is this a known bug? Is there a
workaround for it?
Gelson
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Re: [Asterisk-Users] Wiki down
Michael George wrote: Apparently there is no mirror or anything for it? I've been in the groove for a couple days making great progress, but I need the application documentation... You can always use the Google cached pages. I survived :-) today searching at google and reading their cache instead of following the link. Gelson On May 27, 2004, at 8:44 AM, Gregory Junker wrote: http://www.voip-info.org gives: Warning: mysql error: No Database Selected in query: select `name` ,`value` from `tiki_preferences` in /var/www/html/tikiwiki-1.8.2/lib/tikidblib.php on line 133 Values: Array ( ) $result is false $result is empty Was going to grab a link to give to Florent regarding his CTI thread and question about how to program against the Asterisk API... ___ 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 -Michael ___ 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 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
Re: [Asterisk-Users] Re: FireFly doesn't work with 3rd party anymore
Hermann Wecke wrote: On Thu, 27 May 2004, Tony Mountifield wrote: No more SIP, No more IAX. It was a damn good IAX client... too bad its crap now. Are you sure? http://www.virbiage.com/firefly/download/ still says the following: [...] I just download the latest version (1.7 Build 3532) and they are no settings for IAX/SIP, only their own network. Also, it will only crash on my computer after a few seconds, not miliseconds as before... Hi Hermann I also downloaded latest version and found no IAX/SIP settings. Then I followed their devel link and it still has the settings. However, I tried to setup SIP and it not even try to register at Asterisk. Gelson ___ 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
Re: [Asterisk-Users] Asterisk on Compact PCI platform
David H Hickman wrote: I have it working on an industrial single board pc. :) Could you post some more info about your setup? Like board brand/model, what kind of interfaces are you using and even some photos :-) Seems a very interesting project... is there anybody else running a small/compact asterisk system? I would love to have such a small system that I could send to parents, instruct them to turn it on and plug their pstn line and broadband connection and have a pstn x sip intelligent call router that requires no user intervention. Gelson David Hickman TSG Computer Consulting - Auctions 314-865-4752 x2 On May 18, 2004, at 8:42 PM, Jacques Leisy wrote: Anybody running * on a compact PCI platform? I got a few CPCI boards on eBay including a T1 Natural Microsystems AG4000? Any hope to ever get * running on that platform? Linux Suse 9.0 is running fine Thanks Jacques ___ 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] Areski CDR graph incorrect
Is anyboby using the areski CDR reporting tool? I have installed asterisk-stats v1.2 three days ago, but I found a possible bug in it. My calls compare graphic shows most of the on the calls at first hours past midnight, and it never logs anything after lunch time. This is wrong, my calls are made on business hours. The call log lists those calls at the right time. Is there something I should set on the graphic engine, like a timezone or something? I´m on Brazil, timezone GMT -3. Tried to email the author directly but got no answer. Thanks, Gelson ___ 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] Help with busydetect (no hangups)
I´m using * 0.9.0 and have a X100P connected to my analog PBX. I can´t detect hangups on this line, so I turned on busydetect=yes in zapata.conf. I also have busycount=6. While the line is connected to the PBX, I can never detect busy and the line hangs at the end of every call. If I connect the same X100P to the telco line, without the PBX, then it can detect busy and hangs up the line after 6 busy tones, as expected. I have recorded the busy tones and found that telco uses a standar one (250ms tone, 250 ms silence). My PBX, however, is using a 120ms tone, 80ms silence sequence. How can I adjust the detection routines to the tone I´m getting? I have tried to mess with busy_min, busy_max etc on dsp.c with no luck. I´m sure I doesnt really understand the meaning of those parameters. A also tryed to compile using TONE_ONLY but it gives a compilations error. Can someone suggest what times should I been using? Thanks a lot. Gelson ___ 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
Re: [Asterisk-Users] X100P Answer
On Fri, 23 Apr 2004 14:55:52 +0100, Mark Olliver wrote Hi, I seam to have a problem working out how to get my X100P to answer after 1 ring. Currently it is working fine and connects to the switchboard menu correctly but just does it after 4 rings, which I would prefer if we could reduce. I found recently that my X100P was getting two rings before answer. That´s because the way Caller ID works in US; it sends the info after the first ring and my board was waiting for it. I disabled caller ID on zapata.conf using usecallerid=no and now it aswers on first ring. Gelson ___ 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] Supported USB adapters ?
Hello all, Reading the varios sources of documentation I can only find reference to one FXS USB adapter being supported by asterisk: the Digium S100U. Is there other alternatives? I´ve found that http://www.tjnet.com/ makes a voip chipset that has been used on a lot of internet phones and usb adapters. Can we use these also? I´m asking this because I´m trying to find a cheap alternative to FXS interfaces. I need one or two for a hone PBX and ATA´s are very expensive hard to find here in Brazil. Thanks, Gelson ___ 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
Re: [Asterisk-Users] X100P Tone-based Supervisory Disconnect ?
Steven Critchfield wrote: On Mon, 2004-03-22 at 15:26, [EMAIL PROTECTED] wrote: The X100P hangup problem is indeed pervasive. My current testbed has the X100P connecting to an FXS breakout of a dual ISDN channel box. Indeed, remote hangup is NOT detected. When I switched it to a POTS line, all the sudden it seemed to work OK. This is a serious limitation in some scenarios however. There was a response a few days ago about recompiling [*] with some different options in the driver (had to do with the same problem on a UK line). When this gets resolved, this would make for another fine wiki addition. Take responsibility for your ISDN equipment not providing what you need. Either it needs to provide you with disconnect supervision or it needs to play a set of tones that are identifiable and specified for the audio disconnect to work. Does it mean * supports tome based disconnect? How can I turn it ok? That what my original question (i´m the original poster). Thanks, Gelson ___ 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] X100P Tone-based Supervisory Disconnect ?
Hello all, Through the previous two weeks I have been working on my first asterisk installation. So far all my doubts and problems were aswered by the list history or on-line documentation. I have a two SIP softphones + external analog line working fine. However, I just came across a documented problem that as far as I can tell there is no good answer: the X100P FXOdoes not hung up when the remote end disconnects. I have a X100P connected to an extension of my company PBX. When I get an incoming call, it stays up even when the caller disconnects and I have to use a soft hungup. I have already tried all signalling methods (ks, gs, ls) with no luck. Searching on the web I found a good description of the problem on the Cisco site: http://www.cisco.com/warp/public/788/signalling/fxo_disconnect.html There is a method of hang up detection there called Tone-based Supervisory Disconnect. It seems to me that it would fix my problems, because my company PBX does send some fast tones when the line gets disconnected by the remote end. Does asterisk have this feature? How can I turn it on? Thanks a lot, Gelson ___ 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
