Re: [asterisk-users] Searchable Archives of this list
On Thursday 14 December 2006 13:31, cb wrote: Is there a searchable archive of this list? Did I overlook something obvious? I can find the archives, but short of downloading all the monthly gzips and building my own searchable database, it seems my only other option is to go month by month looking at subjects and hope to stumble on what I'm after. Does anyone maintain a public searchable version of the archives? Google does :) http://www.google.com/search?q=something+site:lists.digium.com -- http://nicegear.co.nz New Zealand's VoIP supplier ___ --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] Recording outbound analog calls with X100P
On Thursday 16 November 2006 06:44, Conrad Wood wrote: On Thursday 16 November 2006 06:42, Matthew J. Roth wrote: As per ManxPower at #asterisk, it is not possible to record a call dialed from an analog phone connected to the Phone In port of an X100P because the two ports on the card are hard-wired together. A bit off-topic maybe, but does that then mean you can't make 2 simultaneous calls through the card? E.g. 1. Call: pstn-phone - asterisk - sip... 2. Call: sip-phone - asterisk - pstn... As he said above, the ports are wired together. There is no FXS device on that card. -- http://nicegear.co.nz New Zealand's VoIP supplier ___ --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] Fax Digium
On Wednesday 08 November 2006 13:15, Ken Williams wrote: I was planning on using a TDM400P with 3 FXO 1 FXS, with the 1 FXS being used for a fax machine. It now appears that Digium doesn't support this, are there other manufacturers anyone can recommend that will support it? Has anyone used a TDM400P in this setup and had it work without much issue? Yes, I have implemented this on a few occasions and it has worked fine for me. hads -- http://nicegear.co.nz New Zealand's VoIP supplier ___ --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] Unicall Installation
On Monday 23 October 2006 21:45, Angel Heart wrote: Hi, Could anyone knows what went wrong with the error below result of installation of libsupertone. [EMAIL PROTECTED] latest]# tar xvf libsupertone-0.0.2.tar [snip] libsupertone-0.0.2/aclocal.m4 [EMAIL PROTECTED] latest]# ./configure --prefix=/usr/local/lib -bash: ./configure: No such file or directory [EMAIL PROTECTED] latest]# Help, pleeeaaassseee... You probably shouldn't blindly follow instructions if you don't know what they do. ./configure should be running the script called configure in the current directory. Which, as the error message states, doesn't exist. You need to change into the correct directory (cd) before you execute the script. -- http://nicegear.co.nz New Zealand's VoIP Supplier ___ --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] Locking phones at night...
On Wednesday 18 October 2006 05:47, Conrad Wood wrote: To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. This is a good use for the AstDB hads -- http://nicegear.co.nz New Zealand's VoIP supplier ___ --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] 1.4 beta voicemail warning
On Tuesday 17 October 2006 17:07, Carla Schroder wrote:
hey all,
I'm getting this warning on the console when I leave a voicemail on my test
server:
[Oct 16 20:56:36] WARNING[3853]: app_voicemail.c:6552 vm_exec: Prefixing
the mailbox with an option is deprecated ('[EMAIL PROTECTED]').
[Oct 16 20:56:36] WARNING[3853]: app_voicemail.c:6553 vm_exec: Please move
all leading options to the second argument.
This is what voicemail.conf looks like:
[local-vm-users]
250 = 1234,User1 One
251 = 3456,User2 Two
252 = 4567,User3 Three
This is what extensions.conf says:
[local-users]
exten = 250,1,Dial(SIP/User1,10,rt)
exten = 250,2,VoiceMail([EMAIL PROTECTED])
It works fine despite the warnings. I've been trying to change the configs
to make the warnings go away, but nothing I've tried works. Anyone know
what to do?
Do what the warning message says ;)
Change;
exten = 250,2,VoiceMail([EMAIL PROTECTED])
to;
exten = 250,2,VoiceMail([EMAIL PROTECTED]|u)
hads
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Re: [asterisk-users] iaxyprov downloading problems
On Friday 22 September 2006 15:21, Sean Kennedy wrote: I just recently purchased some iaxy devices. Being new to this, I didn't have the iaxyprov tool, so I downloaded the instructions and attempted to follow them. Below is the problem I ran into. [EMAIL PROTECTED] src]# svn co http://svn.digium.com/svn/iaxyprov/trunk svn: 'trunk' is already a working copy for a different URL Looks like you already checked out Asterisk or something to the directory trunk in the current working directory. Try; mv trunk whatever then; svn co http://svn.digium.com/svn/iaxyprov/trunk iaxyprov to check out iaxyprov to the iaxyprov directory rather than trunk. hads -- http://nicegear.co.nz New Zealand's VoIP supplier ___ --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 server crashes after two years
On Friday 01 September 2006 16:32, Ronald Wiplinger wrote: 2 years Asterisk sounds strange, since I can remember there was a bug with the date a year ago. If you have not upgraded, than this bug is still in your code. Maybe you just meant no reboot for two years. That bug was only in one version IIRC hads -- http://nicegear.co.nz New Zealand's VoIP supplier ___ --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] voicemailmain
On Friday 25 August 2006 08:39, existx wrote: The error from the CLI is: Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241 socket_read: Rejected connect attempt from 192.168.0.23, request '[EMAIL PROTECTED]' does not exist It looks like you have created 2699 in a different context than your phones. You will need to include = the-context to be able to dial the extension. -- http://nicegear.co.nz New Zealand's VoIP supplier ___ --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] STRFTIME dialplan function not picking up system timezone
On Wednesday 16 August 2006 10:18, Hadley Rich wrote:
I've just been playing with the STRFTIME dialplan function and am having
trouble getting it to pickup my systems local timezone.
According to show function STRFTIME and voip-info.org all the arguments are
optional and according to voip-info.org if you leave them out they will
default to the current time, the current timezone and %c respectively.
My local timezone is Pacific/Auckland (GMT+12) which is setup correctly
AFAIK - date returns the correct time and timezone. I have also tried
setting TZ=Pacific/Auckland and running asterisk at that console which
didn't alter the behaviour.
If I call a test extension with this in the dialplan;
NoOp(${STRFTIME(,,)}))
NoOp(${STRFTIME(,Pacific/Auckland,)}))
then I get this output (shortened) ;
NoOp(SIP/800-081778a4, Tue Aug 15 22:11:36 2006))
NoOp(SIP/800-081778a4, Wed Aug 16 10:11:36 2006))
I have also tried reading asterisk/stdtime/localtime.c which is (I think)
where this stuff goes on but it's over my head.
Does anyone have any ideas as to why I can't get this to work or am I
expecting the wrong behaviour (using SVN trunk)?
After further playing it's not just STRFTIME. Voicemail and other things such
as SayUnixTime are showing GMT time although cdrs (using cdr-csv) and
Asterisk logs show the correct time.
From looking at the code it appears that logs and cdrs use localtime_r
directly whereas the dialplan functions use stdtime/localtime in the Asterisk
source. Although as I said, that's a bit above me in terms of C.
Anyone have any ideas?
hads
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[asterisk-users] STRFTIME dialplan function not picking up system timezone
Hi all,
I've just been playing with the STRFTIME dialplan function and am having
trouble getting it to pickup my systems local timezone.
According to show function STRFTIME and voip-info.org all the arguments are
optional and according to voip-info.org if you leave them out they will
default to the current time, the current timezone and %c respectively.
My local timezone is Pacific/Auckland (GMT+12) which is setup correctly
AFAIK - date returns the correct time and timezone. I have also tried setting
TZ=Pacific/Auckland and running asterisk at that console which didn't alter
the behaviour.
If I call a test extension with this in the dialplan;
NoOp(${STRFTIME(,,)}))
NoOp(${STRFTIME(,Pacific/Auckland,)}))
then I get this output (shortened) ;
NoOp(SIP/800-081778a4, Tue Aug 15 22:11:36 2006))
NoOp(SIP/800-081778a4, Wed Aug 16 10:11:36 2006))
I have also tried reading asterisk/stdtime/localtime.c which is (I think)
where this stuff goes on but it's over my head.
Does anyone have any ideas as to why I can't get this to work or am I
expecting the wrong behaviour (using SVN trunk)?
Cheers,
hads
--
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Re: [asterisk-users] No path to translate from SIP/30-09df8928(4) to SIP/3470075XXXX-09dfb518(256)
On Friday 11 August 2006 18:26, Wolfgang Paul Rauchholz wrote: Aug 11 08:00:24 WARNING[2612]: channel.c:2706 ast_channel_make_compatible: No path to translate from SIP/30-09dfbdb8(4) to SIP/3470075-09e01778(256) Aug 11 08:00:24 WARNING[2612]: app_dial.c:1595 dial_exec_full: Had to drop call because I couldn't make SIP/30-09dfbdb8 compatible with You don't have the g729 codec installed by the looks. hads -- http://nicegear.co.nz New Zealand's VoIP Supplier ___ --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] safe_asterisk to start latest version from SVN - trying asterisk with googletalk
On Saturday 12 August 2006 14:30, Marco Mouta wrote: [Aug 12 03:27:35] VERBOSE[26610] logger.c: [format_mp3.so][Aug 12 03:27:35] WARNING[26610] loader.c: missing mod_data for format_mp3.so What could be wrong? Looks like you have an old format_mp3 module in your module directory. Removing this (and any other old modules) or adding a noload to modules.conf should fix the problem. If you compiled trunk it should have given you a big fat warning about old modules on installation. hads -- http://nicegear.co.nz New Zealand's VoIP supplier ___ --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] Ring Groups
On Monday 07 August 2006 06:36, Chris Hembrow wrote: I am new to asterisk, and learning as I plod along. Currently, I am trying to work out how to create a ring group without using AMP. You should check out the book - 'Asterisk: The Future of Telephony' - published by O'Reilly it's available to buy or download. It will give you a good starting point. I set my inbound line to ring multiple lines by using Dial(SIP/101,SIP/102) but when I answered the call, the lines which didn't answer became locked with no dialtone, as though on a call. That dial line should be Dial(SIP/101SIP/102) - the comma (or a pipe, |) separates what to dial from the options to the dial command. typing 'show application dial' from the Asterisk CLI will get you all the gory details. I thought that setting up a ring group might help, but could only find references to creating them through AMP. 'Ring Group' is just an AMP term, you are going about it the right way above. HTH hads -- http://nicegear.co.nz New Zealand's VoIP supplier ___ --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] Information about Softphone support G729 ?
On Saturday 22 July 2006 11:49, Adrian wrote: Anybody know about (open source with java or C++ ) Softphone support G729 ? At a guess, none because it costs money. hads -- http://nicegear.co.nz New Zealand's VoIP supplier ___ --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] A very lost newbie.
On Friday 21 July 2006 10:39, David R. wrote: My question is this: Where can I find good starter documentation(s) for my purposes? O'Reilly have published a book 'Asterisk: The Future of Telephony' under a Creative Commons licence. This is usually a good place to start. You'll find it here; http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 hads -- http://nicegear.co.nz New Zealand's VoIP supplier ___ --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] RE: Is there a search feature?
On Wednesday 05 July 2006 15:10, David Beckerdite wrote: Is there an archive for this list that can be searched? If so, could someone tell me where it's located? http://www.google.co.nz/search?q=site%3Alists.digium.com/pipermail/asterisk-usersie=UTF-8oe=UTF-8 -- The person you rejected yesterday could make you happy, if you say yes. ___ --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] disabling modules - how?
On Wednesday 21 June 2006 18:36, Tyler Retzlaff wrote: Hello, I am altering an asterisk configuration and would like to eliminate the loading of modules I do not want or do not need at the moment. For example I am do not want to use chan_zap (I'm using chan_capi) and don't want to be bothered with music on hold at the moment. Is there a way to configure these things off so asterisk doesn't try to load them? Or do I have to just move/delete the chan_xxx.so from /usr/lib/ asterisk/modules? What's the right thing to do? modules.conf -- Psychoanalysis?? I thought this was a nude rap session!!! ___ --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] Fun with Echo -- Follow up
On Wednesday 21 June 2006 03:30, Brian Swan wrote: 2. Use fxotune in zaptel-trunk: Find a silent-termination test number from the phone company and use FXOTune. I couldn't get it to dial right in order to get silence on the line. You can verify if it's working correctly by running it with an analog handset connected to your phone line. Pickup the handset and then run the command. In my case, fxotune would never clear the line, or dial the silent termination number I was giving it, not sure if this is a bug or not. What I eventually had to do was pick up the phone, dial the silent-termination number manually, run ./fxotune -i -b 4 -e 4, and quickly hangup the phone. This was the only way I got good results from the program. I had that problem too, it's a bug, see 7264 on mantis. hads -- Two wrongs don't make a right, but three lefts do. ___ --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] Config Revision Control
On Saturday 03 June 2006 09:37, Douglas Garstang wrote: Aaron, I'm trying to check-in (is that the right term?) the files for the first time. There's nothing in the repository yet. http://svnbook.red-bean.com hads. ___ --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] Config Revision Control
On Saturday 03 June 2006 10:05, Douglas Garstang wrote: [stuff regarding subversion] http://subversion.tigris.org/servlets/ProjectMailingListList -- Never try to explain computers to a layman. It's easier to explain sex to a virgin. -- Robert Heinlein (Note, however, that virgins tend to know a lot about computers.) ___ --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] Callerid and trunk
On Wednesday 31 May 2006 09:08, Julian Lyndon-Smith wrote:
Ok, I must be really stupid here -
I'm playing with ael and svn trunk.
given the following in ael:
context isdn10 {
444601 = {
Answer();
NoOp(${CALLERIDNUM});
Hangup();
};
};
isdn10 is the incoming isdn context.
why do I get this on the console:
-- Accepting call from '01702xx' to 'yy' on channel 0/1, span 1
-- Executing [isdn10:1] Answer(Zap/1-1, ) in new stack
-- Executing [isdn10:2] NoOp(Zap/1-1, ) in new stack
-- Executing [isdn10:3] Hangup(Zap/1-1, ) in new stack
callerid must be working: get the from (01702xx) and to yy
but why is ${CALLERIDNUM} blank ?
Because it's deprecated and I assume dropped completely for 1.4. Use
${CALLERID(num)}
hads
--
Destiny is a good thing to accept when it's going your way. When it isn't,
don't call it destiny; call it injustice, treachery, or simple bad luck.
-- Joseph Heller, God Knows
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Re: [Asterisk-Users] Quad BRI card
On Thursday 18 May 2006 18:35, stoffell wrote: Aside from being available.. What driver does it use? Will it be needing bristuff ? (that wouldn't work I guess) Or will the near future integrate BRI ( and hfc?) drivers in asterisk? And thus, making bristuff obsolete? (wich means, BRI users will be able to use cvs easily..) Just to make clear I'm very curious on this card. And yes I'm in europe ;) I'm curious too, unfortunately I don't know anything more about it sorry. hads. -- The means-and-ends moralists, or non-doers, always end up on their ends without any means. -- Saul Alinsky ___ --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] Quad BRI card
On Thursday 18 May 2006 08:59, Wayne Gemmell wrote: Does Digium make a quad BRI card? I can't see anything of the sort on their page but I thought they might call it something else in the States. They do, but it isn't released yet. Put B410P into google and you will get a couple of hits. Digium's marketing page says it is available and the distributor I use had one on show the other day so they can't be too far away. Failing that, can anyone recommend a make/model that would handle 4 BRI ports? Many people seem to like the Eicon Diva cards. HTH hads -- A Vulcan can no sooner be disloyal than he can exist without breathing. -- Kirk, The Menagerie, stardate 3012.4 ___ --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] Linksys IP Device Bulk Provisioning Guide
On Friday 12 May 2006 16:50, Kerry Garrison wrote: I have written up an guide on how to do bulk provisioning of the Linksys phones and ATAs. http://voipspeak.net/index.php?option=com_content http://voipspeak.net/index.php?option=com_contenttask=viewid=73 task=viewid=73 Thanks for the good article. One thing - the config file link at the bottom doesn't seem to work. Cheers, hads -- The picture's pretty bleak, gentlemen... The world's climates are changing, the mammals are taking over, and we all have a brain about the size of a walnut. -- some dinosaurs from The Far Side, by Gary Larson ___ --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] SPA941 SPA942 BUG. auto answer does not work.
On Thursday 04 May 2006 20:53, Asterisk wrote: The handsets do not work with the SIP flag to make them AUTO-ANSWER. (As documented is should) Ie, you cannot use them with intercom or Page features. Works fine here; SIPAddHeader(Call-Info:\;answer-after=0) hads -- You buttered your bread, now lie in it. ___ --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] Excessive Asterisk delay to answer on ZAP inbound call
On Wednesday 26 April 2006 20:59, Giorgio Incantalupo wrote: Why does Asterisk wait for these two rings? What is it doing meanwhile? Is it possible to shorten this interval to have an immediate response? It's most likely waiting on callerid info. If you set usecallerid=no in your zapata.conf you should see it pick up faster, although without callerid. HTH hads -- CChheecckk yyoouurr dduupplleexx sswwiittcchh.. ___ --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 not disconnecting after PSTN line hangs up (getting empty voicemails)
On Tuesday 25 April 2006 05:50, Mike Garey wrote: When someone calls into our asterisk server over a PSTN line, dials an extension and then hangs up, the SIP phone related to the given extension will ring about 4 or 5 times before asterisk shows that the channel has been hung up in the console. This isn't such a big deal on its own, but what's happening now is that if a user calls in from a PSTN line, gets voicemail on the extension, and hangs up before the voicemail starts to record, an empty message will still be recorded and sent to the user. It sounds very much like you need disconnect supervision. http://www.voip-info.org/wiki-Asterisk+Disconnect+Supervision You'll need to see what your provider provides (if anything) and setup your zaptel.conf/zapata.conf accordingly. hads -- A fool and his money are soon using Windows. ___ --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] MultiBRI in Australia - found one - maybe
On Wednesday 12 April 2006 18:51, MBIT Technologies wrote: [regarding the Draytek Minivigor 128] Any idea where I can get some of these units in Melbourne? According to Draytek AU they have been discontinued :( hads -- Age is important only if you're cheese and wine. ___ --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] Distinctive Ring on SPA941
On Wednesday 05 April 2006 11:56, Cory Hawkless wrote: Does anyone know how to set the distinctive ring on the Linksys SPA941? Try; SET(_ALERT_INFO=Classic-1) hads -- bureaucrat, n: A politician who has tenure. ___ --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] Not Found in archive
On Friday 24 March 2006 05:37, Rich Adamson wrote: Michael Welter wrote: I'm seeing quite a few Not Found pages when I google lists.digium.com. Is anyone else getting this? Yes, and it apparently has something to do with changes made at the digium server. Don't have a clue whether anyone is working at correcting the issue. A workaround for is to note the subject of the message from either the Google search page or cache, visit the non-existant link, move up a directory (i.e. remove the filename from the end of the URL) and then search for the subject in the message list. HTH. hads -- Every photograph of an American atomic bomb detonation was taken by Harold Edgerton. ___ --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] How to create [new_context] in extensions.conf?
On Friday 24 March 2006 12:53, Larry Alkoff wrote: What do I have to do to dial an exten - with the dial command in it? Asterisk isn't recognizing commands in my newly created [context]. There is a really good book available here[1] that will answer this and a lot of other questions easily and quickly for you. [1]http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 hads -- TeX is potentially the most significant invention in typesetting in this century. It introduces a standard language for computer typography, and in terms of importance could rank near the introduction of the Gutenberg press. -- Gordon Bell ___ --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] TDM400 FXO module not answering or dialing out.
On Thursday 23 March 2006 02:08, Dr. Michael J. Chudobiak wrote: I have hit a wall configuring a TDM400, I have set these up before without issue but today I just can't seem to figure out what I am doing wrong. I couldn't make TDM400/FXO work on my 1.2.5 Asterisk either. It wouldn't answer calls, for unknown reasons. Is anyone else having this problem or am I just going mad? FWIW I just tried an old X100P on the line and it works correctly. I don't think I am doing anything wrong in my configuration. Cheers, hads -- Madison's Inquiry: If you have to travel on the Titanic, why not go first class? ___ --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] [SOLVED] TDM400 FXO module not answering or dialing out.
On Thursday 23 March 2006 09:38, Hadley Rich wrote: Is anyone else having this problem or am I just going mad? FWIW I just tried an old X100P on the line and it works correctly. I don't think I am doing anything wrong in my configuration. OK, self reply again. Apologies, yes I was going mad. It just goes to show that you should always check everything possible, even the simple things -- it was the cable. Odd since it worked with the X100 and another TDM card but there you go. Cheers, hads -- In those days he was wiser than he is now -- he used to frequently take my advice. -- Winston Churchill ___ --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] TDM400 FXO module not answering or dialing out.
Hi all, I have hit a wall configuring a TDM400, I have set these up before without issue but today I just can't seem to figure out what I am doing wrong. On an incoming call the following is produced in the Asterisk console with verbose 4 -- Starting simple switch on 'Zap/2-1' Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 2 (Ring/Answered)... Mar 22 16:12:37 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... -- Executing Answer(Zap/2-1, ) in new stack -- Executing Wait(Zap/2-1, 15) in new stack Mar 22 16:12:37 WARNING[2051]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 2 Mar 22 16:12:38 WARNING[2051]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 2 Asterisk doesn't answer the line. For debugging I have a POTS phone plugged into the line as well as the FXO and this phone continues to ring. The Answer and Wait are in the dialplan merely for debugging this problem. I cannot dialout via the FXO either, Asterisk appears to be doing everything correctly but never touches the line. I am running Asterisk 1.2.5 and Zaptel 1.2.4 on Kernel 2.6.15.6 with a TDM12B REV I The board has a FXS module installed in slot 1 and a FXO module installed in slots 2 and 3. The following is my /etc/zaptel.conf; loadzone=nz defaultzone=nz fxoks=1 fxsks=2-3 The following is my /etc/asterisk/zapata.conf; [channels] context=internal signalling=fxo_ks channel = 1 ;busydetect=no ;callprogress=no context=from-zap signalling=fxs_ks channel = 2-3 I am loading the wctdm module with opermode=NEWZEALAND but have tried omitting this with no result. There are some posts in the list archive from late 2003 about this same issue that were resolved by adding busydetect=no/callprogress=no to zapata.conf although this seems to have no effect for me. I can find no other solutions via Google. Any help would be gratefully appreciated. I have omitted some needed information then please let me know. Cheers, hads -- There's no secret to balance. You just have to feel the waves. -- Darwi Odrade ___ --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] TDM400 FXO module not answering or dialing out.
On Wednesday 22 March 2006 16:24, Hadley Rich wrote: I have hit a wall configuring a TDM400, I have set these up before without issue but today I just can't seem to figure out what I am doing wrong. On an incoming call the following is produced in the Asterisk console with verbose 4 -- Starting simple switch on 'Zap/2-1' Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 2 (Ring/Answered)... Mar 22 16:12:37 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... -- Executing Answer(Zap/2-1, ) in new stack -- Executing Wait(Zap/2-1, 15) in new stack Mar 22 16:12:37 WARNING[2051]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 2 Mar 22 16:12:38 WARNING[2051]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 2 Asterisk doesn't answer the line. For debugging I have a POTS phone plugged into the line as well as the FXO and this phone continues to ring. The Answer and Wait are in the dialplan merely for debugging this problem. I cannot dialout via the FXO either, Asterisk appears to be doing everything correctly but never touches the line. I am running Asterisk 1.2.5 and Zaptel 1.2.4 on Kernel 2.6.15.6 with a TDM12B REV I Please excuse the self reply but I missed some info from my original post. The FXS module in the board works as expected - I can dial to and from it and SIP phones. I have tried shifting the modules around the board and using the two different modules with the same result. The card has it's own interrupt and zttest consistently gives 99.987793% I have used another TDM card on this line with Kewlstart signalling and it worked correctly, this was with Asterisk 1.0.9 though. Thanks again. hads -- He who is known as an early riser need not get up until noon. ___ --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] How hard to create Asterisk for Compact Flash?
On Wednesday 01 March 2006 14:15, mustardman29 wrote: I am aware of Astlinux and the other embedded Asterisk solutions out there? Astlinux is nice but the problem is that when I hit a snag and need to incorporate a patch and what not I cannot do that with Astlinux because I cannot compile my own version. How hard is it to create my own version of Linux/Asterisk to run on Compact Flash. I have seen 1GB Sandisk CF for as low as $50 recently so small size is not too critical. I won't be using AMP or anything like that either. The most important thing is for it to be read only so the CF is not constantly being written to so it will last a long time. Voicemail and config files will be stored on a second CF that is read/write. You could use any distro you want really. Some good options would probably be; - Debian (check out flashybrid package for read only root) - Gentoo (I know there is a read only root tutorial around somewhere) - Slackware (read only root should be fairly easy too) - Arch (I have my own experimental read only root package for this --uses rsync, similar to flashybrid) I personally like Arch Linux because of the ease of integration of binary and source compiled packages. So incorporating a patch like you mentioned above is a simple edit of a PKGBUILD and `makepkg`. My standard (not optimised in any way) install with everything I need to run Asterisk comes in just under 500MB. The downside of Arch for a server is that it has a rolling release. Of course everyone has their favourite distro, that is just MHO. HTH hads -- BOFH excuse #34: (l)user error ___ --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] Sipura SPA 3000 logic
On Wednesday 08 June 2005 12:25, Richard Smith wrote: Would a call coming in on the pstn line be answered by the ATA or just get passed through to the * server (depending on dialplan) to handle? So basically, the caller does not get charged until the appropriate extension hanging of the * server answers. The ATA will answer the POTS line, therefore the caller will be charged as soon as the ATA has tried to grab caller id and picked up the line (usually around two rings). hads -- We're fighting against humanism, we're fighting against liberalism... we are fighting against all the systems of Satan that are destroying our nation today...our battle is with Satan himself. - Jerry Falwell ___ --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] Sipura SPA 3000 logic
On Wednesday 08 February 2006 14:46, Chris Bagnall wrote: This is incorrect. Whilst the SPA3000 *can* work this way if you wish, it doesn't have to. Apologies, you are correct, there is more than one mode of operation. hads -- timesharing, n: An access method whereby one computer abuses many people. ___ --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] How to find out if a new voicemail exists
On Wednesday 18 January 2006 07:21, Sean Cook wrote: Koopmann, Jan-Peter wrote: I would like to see if during a call a new voicemail was recorded. I want to send a SMS to mobile phones if someone recorded a message on our voicemail system. I can use VMCOUNT to see if there are new messages in the Inbox but this will result in new SMS being sent even if the caller hangs up during the Voicemailpromt, at least if there are still unread/unheard messages in the inbox. Is there some option or variable I missed or is this a feature request? I believe you can use the externnotify to accomplish this... Here[1] is a small, untested script that does something similar to what you are asking. Maybe you could modify it to you needs. HTH hads [1]http://astug.org.nz/pipermail/users/2005-November/22.html -- Everyone is more or less mad on one point. -- Rudyard Kipling ___ --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] uplink call quality issues
On Monday 16 January 2006 15:20, Esteban Guana-Jarrin wrote: We are using [EMAIL PROTECTED] 1.5 and SIP trunks to communicate to the PSTN network. We are having some problems with the call quality. Although we can hear the other person's voice quite clear when making or receiving a call, we get complaints from the people on the other end saying that our voices sound very unclear, low and that the voice drops, therefore people on the other end can not understand what we are saying. But as I said in our end their voices sound clear. I have checked network wise and found no latency problems within our small LAN, with our VoIP provider and reaching their SIP server's IP address, also the CPU load in the asterisk server has been graphed and does not exceed the normal CPU load levels Any assistance will be very much appreciated You could be saturating your upload traffic? What is the upload speed of you connection? hads -- Nap: Going back to sleep after taking a shower. -Jason ___ --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] Extensions.conf error - 'Maximum Include level(10) exceeded'
On Saturday 14 January 2006 13:07, Douglas Garstang wrote: This therefore means that is a maximum of 9 #include statements that can be put into extensions.conf. This is a SERIOUS SERIOUS (how many times can I say it?) limitation. I thought Digium said that Asterisk was supposed to be enterprise-grade? Quit your whinging already. Learn some interaction skills. hads -- You can observe a lot just by watching. -- Yogi Berra ___ --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] Re: Failover Device?
On Friday 13 January 2006 15:59, James Harper wrote: Can anyone recommend a PRI-to-TDMoE device? Does such a thing exist? Have you seen the Redfone foneBRIDGE? I have no experience of it but it seems to be what you are after. HTH hads -- I WILL TRY TO RAISE A BETTER CHILD I WILL TRY TO RAISE A BETTER CHILD I WILL TRY TO RAISE A BETTER CHILD I WILL TRY TO RAISE A BETTER CHILD Marge Simpson on chalkboard in episode 9F03 ___ --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 needed
On Tuesday 10 January 2006 11:40, Amir Aziz wrote: I have just installed [EMAIL PROTECTED] version 2.1. It keeps working fine for couple of days. But after couple of days I start getting the following error as the Asterisk does not start automatically so I try to start it with asterisk -vc. Any ideas? and how to fix this error. Thank you for your help. == Parsing '/etc/asterisk/zapata.conf': Found == Parsing '/etc/asterisk/zapata-auto.conf': Found == Parsing '/etc/asterisk/zapata_additional.conf': Found [EMAIL PROTECTED] ~]# Ouch ... error while writing audio data: : Broken pipe Try asking your questions on the [EMAIL PROTECTED] forums/mailing list. It sounds like it is related to music on hold. hads -- Flashlight, n. An instrument of imperception which obscures vision by producing a concentrated glare at one point which is sufficiently intense to prevent the user from seeing anything else. Environmentalists have brought the cleverness of this device one step further by producing the solar powered flashlight. -- Hayward's Unabridged Dictionary, http://JonathansCorner.com/writings/hud/ ___ --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] TDM400 (TDM11B) configuration
On Tuesday 10 January 2006 12:36, Dan Littlejohn wrote: I have fixed this before, but I cannot for the life of me remember how I did it. I have a TDM400P with one fxo module and one fxs module. I setup Asterisk @Home and everything works fine, except when I try and call out. If I call out with a SIP phone it calls the zap extension and not the pstn line? If I take the zap extension offhook and call with the SIP phone it dials out the pstn line fine. I am not sure why the zap extension is being included in the group, but I cannot find where to change it in AMP or the .conf files. Any help would be appreciated. Try asking on the [EMAIL PROTECTED] forum/mailing list, you will probably get more help there. hads -- When Jennifer Lopez stays in a hotel, she brings her own sheets because she can't sleep on anything with a thread count of less than 250. ___ --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 1.2 + DMZ + NAT clients
On Friday 30 December 2005 07:19, Blake Krone wrote: Hey everyone I have my Asterisk server setup as the DMZ on my Linksys router. If I use the internal IP as the domain in Xlite clients will register and work, however, if I use the FQDN for my asterisk server the clients will not register. I have all the extensions set to NAT=yes and have modified sip.conf to include externip=insert FQDN here, externhost=insert FQDN here, and localnet=192.168.1.0/255.255.255.0 On 12/29/05, Kerry Garrison [EMAIL PROTECTED] wrote: If the machines with X-Lite are on the local network, use the private ip, if they are outside the network, use the public ip. Anyway around that? It's a PITA to have to change that all the time with my PDA laptop. You could set up an internal DNS server that points the FQDN to your private IP. hads -- The world's great men have not commonly been great scholars, nor its great scholars great men. -- Oliver Wendell Holmes ___ --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 shell script for externnotify
On Friday 18 November 2005 15:32, Tom Rymes wrote: Basically, I have 14 after-hours mailboxes that all have different e- mail addresses. I want this script to parse the mailbox number from the original command ($2), and then somehow look that up mailbox's address and send an e-mail. It then checks every five minutes to see if the message has been retrieved, and escalates things as necessary. I don't mind the messy solution of defining all 14 addresses in the script itself, though it would be nice to look it up from voicemail.conf or something eventually. I'm not sure I understand what you are trying to do, but this may (or may not) help. You mentioned looking up the email field from voicemail.conf, this should do that: EXTEN=`echo $2 | cut -f 1 -d @` EMAIL=`cat voicemail.conf | grep '^$EXTEN' | cut -d ',' -f 3` The above ignores contexts so if you have more than one it will not work. HTH hads -- At a recent meeting in Snowmass, Colorado, a participant from Los Angeles fainted from hyperoxygenation, and we had to hold his head under the exhaust of a bus until he revived. ___ --Bandwidth and Colocation sponsored by Easynews.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] X100P troubles?
On Monday 14 November 2005 16:26, Rich Adamson wrote: As I recall, the driver for the x100p was called wcfxs (or something like that), and those driver functions were merged into wctdm about a year ago. Now, wcfxs is an alias for wctdm. I've noticed a lot of people referring to wctdm and from reading through the zaptel-1.0.9.2 source it appears that wcfxs is the actual module. From the README file; wcfxo X100P - Single port FXO interface X101P - Single port FXO interface wcfxs TDM400P - Modular FXS/FXO interface (1-4 ports) and from the Changelog; zaptel 1.0.7 -- Makefile -- An alias has been added so that you can load wcfxs with 'modprobe wctdm'. Cheers, hads -- I used to be disgusted, now I find I'm just amused. -- Elvis Costello ___ --Bandwidth and Colocation sponsored by Easynews.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] Not saving voicemail message
On Friday 28 October 2005 12:06, Richard Smith wrote: [EMAIL PROTECTED] 1.2.0 beta4 writes to the respective voicemail directory and when the call is hung-up the .wav file disappears. Sounds like voicemail.conf is setup to delete the message after it is emailed to the user. You may also want to refer here http://www.catb.org/~esr/faqs/smart-questions.html HTH hads -- You may already be a loser. -- Form letter received by Rodney Dangerfield. ___ --Bandwidth and Colocation sponsored by Easynews.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] Where does Asterisk put it's files
On Friday 28 October 2005 16:22, Eric Bishop wrote: Does anyone have a full list of places Asterisk puts all config files and binaries. I need this to be able to fully rollback if I have a failed upgrade of Asterisk/Zaptel/LibPRI. So far I have: /etc/zaptel.conf /etc/asterisk/ /usr/sbin/safe_asterisk /usr/sbin/asterisk /usr/lib/asterisk/modules/ /usr/include/asterisk/ /lib/modules/`uname -r`/misc /usr/lib/ /usr/include/ Anything I have missed? Check out /etc/asterisk/asterisk.conf if you haven't already. hads -- Euphemisms for calling someone stupid: Clock doesn't have all its numbers. Contents settled some during shipping. Couldn't count to 21 if he were barefoot and without pants. Couldn't pour water out of a boot with instructions on the heel. Couldn't write dialogue for a porno flick. Cranio-rectally inverted. Depriving a village somewhere of an idiot. ___ --Bandwidth and Colocation sponsored by Easynews.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
[Asterisk-Users] New Zealand Asterisk Users Group
Hi, Since we're doing this... There is now a New Zealand Asterisk Users Group set up. There is a wiki and mailing list at http://astug.org.nz both are sparse at the moment and could do with some input. If you're in New Zealand (or not) and interested in Asterisk then sign up and get contributing! Thanks, and please excuse the spam. hads -- I can't decide whether to commit suicide or go bowling. -- Florence Henderson ___ --Bandwidth and Colocation sponsored by Easynews.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
