[asterisk-users] What is WARNING: Got 200 OK on REGISTER that isn't a register?
Hi Last couple of days I received the subject WARNING message on a home-based asterisk pbx. Is someone spoofing a register method on port 5060? Or, is this warning something random (sort of like sporadic alarms on an analog port)? (This warning message is from chan_sip.c). Am running asterisk V1.4.18; (the hardware is an AMD 64 X2 and a Digium 400P with 4 ports - just a home-based pbx) - and using Ubuntu Intrepid (alternate desktop version). Thanks for any reply, Gerald Harshany g...@jerryh.us ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] astcanary not exiting in asterisk V1.6.1
Hi, I only run a home-based asterisk (v1.4.18), and have never patched it, so I'm a unfamiliar with what time frame to expect for patches being implimented. I just downloaded (April 14) svn asterisk V1.6.1 r188415, on a play machine and noticed that when I stop asterisk, the astcanary module does not exit - when I restart asterisk, a new copy of astcanary also starts. In browsing through the bugs/lists, I see where a patch to trunk V1.6.0.7 was made (recently) due to astcanary not exiting when asterisk dies. Is this a different scenario than just stopping and restarting asterisk (as opposed to asterisk 'dying')? So, I'm just wondering if it is too soon to expect the patch for astcanary to have been applied in svn V1.6.1? Do I need to do something? Thanks for any reply, Gerald Harshany g...@jerryh.us ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] astcanary not exiting in asterisk V1.6.1
- Original Message - From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 15, 2009 9:28 AM Subject: Re: [asterisk-users] astcanary not exiting in asterisk V1.6.1 On Wednesday 15 April 2009 04:25:15 Gerald Harshany wrote: Hi, I only run a home-based asterisk (v1.4.18), and have never patched it, so I'm a unfamiliar with what time frame to expect for patches being implimented. I just downloaded (April 14) svn asterisk V1.6.1 r188415, on a play machine and noticed that when I stop asterisk, the astcanary module does not exit - when I restart asterisk, a new copy of astcanary also starts. In browsing through the bugs/lists, I see where a patch to trunk V1.6.0.7 was made (recently) due to astcanary not exiting when asterisk dies. Is this a different scenario than just stopping and restarting asterisk (as opposed to asterisk 'dying')? So, I'm just wondering if it is too soon to expect the patch for astcanary to have been applied in svn V1.6.1? Do I need to do something? If there's another release candidate, it will show up there. Otherwise, the fix will wait until 1.6.1.1. -- Tilghman OK. Thanks for the reply - will just wait then. Gerald ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is WARNING: Got 200 OK on REGISTER thatisn't a register?
I understand - Thanks for the reply. Yes, I have been registering with the sip provider Voicepulse for about 2 years, but never saw the message before. In the last 2 days or so it has popped up about 5 times each of these days, which started me wondering what the messages really meant. Gerald - Original Message - From: Martin asteriskl...@callthem.info To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 15, 2009 9:33 AM Subject: Re: [asterisk-users] What is WARNING: Got 200 OK on REGISTER thatisn't a register? Your box receives a 200 OK message as though it would have sent the REGISTER sip message - trying to register with a sip provider as a sip device. Asterisk doesn't recognize it because: 1) the REGISTER was not sent from Asterisk 2) the 200 OK was sent too late 3) there's some other issue like NAT or so Martin On Wed, Apr 15, 2009 at 4:30 AM, Gerald Harshany g...@jerryh.us wrote: Hi Last couple of days I received the subject WARNING message on a home-based asterisk pbx. Is someone spoofing a register method on port 5060? Or, is this warning something random (sort of like sporadic alarms on an analog port)? (This warning message is from chan_sip.c). Am running asterisk V1.4.18; (the hardware is an AMD 64 X2 and a Digium 400P with 4 ports - just a home-based pbx) - and using Ubuntu Intrepid (alternate desktop version). Thanks for any reply, Gerald Harshany g...@jerryh.us ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] a simple Asterisk AMI interface with Delphi (or Lazarus+FreePascal)
Hi Everyone, Those of you who have a simple home-based Asterisk box might be interested in a simple Win32 (Win2K or WinXP) interface to the AMI manager. The quick-start versions merely require unzipping with NO Installation - hence, NO Uninstall (i.e., no registry writes at any time by the install nor by the program). (Unfortunately) the INSTALL version does write to the registry due to the database licensing requirements. Would suggest that you download the PDF and, if interested, (or if you hate to read manuals, just ), download the quick-start version which only requires 3 settings in Asterisk's manager.conf file (the user name, the password, and the read/write privileges - program defaults to the 5038 port). The program was really written as a nostalgic cruise down the old Pascal OOP thruway, and not as a contender to the likes of FOP, etc. Pascal has nice features such as declaring any of your functions inline; or for that matter writing inline Assembler code which was the language in the '70s (that is the 1900's, by the way). The Win2K version was compiled on an old Delphi 5 compiler (and for you young'uns, that was circa 1999 when Win2K was unveiled). However, fear not, the WinXP version was compiled with the latest Delphi 2007 R2. However, I did NOT insert some required Vista enabling statements (such as for the glass effect), since I have no interest in testing it (yet) in Vista; so, the XP version may or may not function well in XP compatibility mode within Vista. As for my Subject - Is anyone in this Asterisk group doing anything using Lazarus and FreePascal for the Asterisk box? The FreePascal compiler is a total (and, yes, an open source work in progress) cross-platform compiler. What I mean is, it can compile for Win, Mac, and Linux, but also for about half a dozen CPU's. The documentation for the compiler is an outstanding example for open-source projects. Downloads and info at: http://www.jerryh.us/Downloads/amifiles.htm Gerald Harshany, Ph.D. Professor Emeritus of Mathematics And again, for you young'uns, Emeritus simply means ancient :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stupid Timeout Question
Hi, It may have to do with the version of Asterisk. I have (basically) the same coding on an Asterisk V1.4.18 box, and a V1.6 SVN test box - in both boxes the Asterisk does execute the = t,1,Playback(connection-timed-out) when nothing is entered. The only differences I can see between your coding and mine, is that a) I simply use the default timeout (i.e., WaitExten() ); but don't see why this matters, and b) I use the m option in the Background command, since I have a one-key extension. You could try using, exten = s,n,Set(TIMEOUT(absolute)=5) before the Background command, and see if this works. Gerald H. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk V1.6.0 SVN debug WARNING(6830) a bug or deliberate?
Hi lists,
Does anyone know if the following error message (from a debug screen) was
a
deliberate change from the behavior in asterisk V1.4.18 or just an
overlooked
parsing error in progressing to V1.6.0? Since, in this case, the string (Hi
there)
is quoted, it doesn't seem as though the parser should take notice about
about the
interior of a 'word'. However, if it is deliberate, then so be it. (a yellow
NOTICE
would be more soothing than a red WARNING) :-)
Gerald Harshany
WARNING(6830): pbx.c:7557 pbx_builtin_setvar: Please avoid unnecessary
spaces on variables as it may lead to unexpected results
('DB(Knowselgreat/Hi there)' set to ' myfile ').
Using current Asterisk version: SVN-branch-1.6.0-r114304 (on Ubuntu) and
Zaptel current SVN 1.4
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Re: [asterisk-users] Roaming callback?
${FromPathFile} ${DestPathFile} 1/dev/null 21 );one long line
exten = s,n,Return()
[doringback] ; in a NEW CHANNEL NOW-the call-file created channel Zap/3
exten = s,1,Verbose(== in context doringback ready to dial ring back
caller)
; THE Zap/2 CALL WILL OCCUR (USUALLY) BEFORE THE CALL-FILE CALL
;
exten = s,n,Dial(Zap/2,20,r)
; check DIALSTATUS etc
exten = s,n,Hangup()
Hope this helps,
Gerald Harshany
Original Message
Subject: [SPAM] Re: [asterisk-users] Roaming callback?
From: SIP [EMAIL PROTECTED]
Date: Mon, April 28, 2008 1:25 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: Jerry Harshany [EMAIL PROTECTED]
Jaap Winius wrote:
Quoting Jerry Harshany [EMAIL PROTECTED]:
There is an additional alternative for a ringback to a caller, which
is to use the Call File capability as noted in Van Meggelen's
Future of Telephone; 2nd ed, p306.
As it says in the book, call files allow calls to be created through
the Linux shell. If you've used this to create a roaming callback
service, then you must have created something that allows users to
submit a phone number to be called back on, after which a .call file
is created and moved to the /var/spool/asterisk/outgoing/ directory.
sleep 8s
mv $1 $2
exit 0
This looks like the step that moves the newly created call file to the
aforementioned directory.
In my case, when the caller calls in to 'asterisk', he is prompted
for the number he wishes to call. The caller can be at a US or
international number, and he can call any US or international
number, WITH or WITHOUT ringback. In other words the caller
designates whether this is a direct connect call, or a ringback (and
then bridge the called number). I have the complete flexibility of
my dial plan extensions to do as I wish with the phone numbers.
This is what I'm really interested in! How did you manage this? Would
you be willing to share how you did this?
I would imagine if it's a callback, it creates a callfile. If it's not,
it just connects the call as it would normally. We have a similar thing
for our business customers built using a reasonably simple agi script to
do verification of the caller/account and creation of the call files. A
rather simple Dial command can handle the direct connection after
verification, and a rather simple call file can handle the callback. The
hardest part was getting the DTMF reading to work well. ;)
N.
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