Re: [Asterisk-Users] Digium list server and spam assassin
On Oct 20, 2005, at 13:00, Jason Pyeron wrote: But the etiquette for SMTP is the HELO is done with a reverse friendly name. Huh? And which RFC say that? This 'assists' in detection of spoofed dns spamming. No, the only thing that will help with that is to do lookups on the envelope sender and see if the source IP is an allowed sender for the domain used in the envelope. I mean, come on... Thorsten ___ --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] Macro exists if an application returned -1
On Sep 21, 2005, at 8:27 , Armin Schindler wrote: Is there any solution for using ChanIsAvail() in a macro? Yes. Fix app_chanisavail.c such that it says if (ast_goto_if_exists (... instead of if (!ast_goto_if_exists( Somone bungled the conversion to use ast_goto_if_exists -- this may or may not be incorrect in other places as well. Thorsten ___ --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] Re: Who is going to AstriCon (The Asterisk Conference)?
On Sep 17, 2005, at 7:08, Steven Sokol wrote: So far we've had at least one person indicate that they would not want to travel to the US at this time. All politics aside, how many out there feel the same way? I do NOT want to start a polical flame-war, but I am curious at the number of people who simply won't come to the States due to the current Administration, the Iraq war, or some other reason. I do! However, since I already live here it does not matter. So I'll be coming Thorsten ___ --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] Sangoma VS. Digium
On Apr 7, 2005, at 4:11, Matteo Brancaleoni wrote: Please find an hardware producer that sells directly to endusers, when they have also distributors/resellers. Apple. The way is: if you have resellers, sell through them. if not directly to end user. Ever been to an Apple store? Thorsten ___ 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] VM: Multilanguage and digits
On Monday, Fedbruary 23rd Olle wrote: Change no to se (who cares about norwegian :-) ) and you'll be ok. And remember to report this to bugs.digium.com - tack! Hey, now!! Thorsten ___ 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] cant load drivers for TE410P cards
The TE410P does not use the wct1xxp driver -- it uses wct4xxp... Thorsten -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:16 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cant load drivers for TE410P cards hello, I have been using the T1 card with my asterisk for a while now, but an attemp to upgrade the system to use a TE410P card ( using the T1 option) i have a 3.3V motherboard. but when i try to load the module it gives the following errors: #modprobe zaptelO.k # modprobe wct1xxp gives /lib/modules/2.4.18-4/misc/wct1xxp.0:init module:no such device Hint: insmod errors cab caused by incorrect module parameters, including invalid IO and IRQ parameters. /lib/modules/2.4.18-4/misc/wct1xxp:: insmod wct1xxp failed. is there a way of installing the TE410P module? regarsd -- Olaifa Augustine General Data Engineering Services Ltd 18b oshin road,kongi bodija p.o.box 29460, secretariate, ibadan. tel:- 234-2-8105156 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto Starting Asterisk
make config does both the copy and the neccecary linking... Thorsten -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott (7805) Sent: Tuesday, December 23, 2003 8:50 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Auto Starting Asterisk An even better way to get asterisk started is to use the init scripts provided with asterisk and the zaptel kernel modules. cp /usr/src/asterisk/init.asterisk /etc/init.d/asterisk cp /usr/src/zaptel/init.zaptel /etc/init.d/zaptel Then do the proper linking, etc to get asterisk to start in your current run level. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David J Carter Sent: Tuesday, December 23, 2003 7:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Auto Starting Asterisk Hi, In rc.local I added the line /etc/rc.d/run-asterisk I then created a small script of 2 lines called run-asterisk #!/bin/sh /usr/sbin/asterisk do a chmod 755 on the file and reboot. The Asterisk server then starts at every reboot. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adthrawn Sent: 23 December 2003 12:18 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Auto Starting Asterisk Hi, I'm a newbie to the list, but have been screwing around with Asterisk for the last 6 months or so (on a purely experimental basis so far). I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm unsure where the line is drawn in terms of Linux issues or Asterisk issues. At present, I have to manually start Asterisk from the command line, but I'd like to have it automatically start up (and in the correct mode) at startup. For now, the server is running as a workstation, so I only need it to run as a background daemon, but in the near future, we're going to run Asterisk of a dedicated racked server, which we would only want to run Asterisk, and there bare minimums required - as far as I'm aware, you could start Asterisk very early on in the boot-up process. Can anybody guide me in configuring the system to start Asterisk from bootup... Probably a highly remedial question - but you've got to start somewhere! Regards, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS repository changes
So, the CVS repository has been redone, like we said a couple of weeks ago. This means that some modules will have to be checked out fresh -- just doing cvs update (or make update) will *NOT* work. The modules in question are: asterisk gastman gnophone libpri zapata zaptel The only other notable difference for the time being is that you will be able to check out previous releases by symbolic tag, e.g. you can now get Asterisk 0.5.0 by doing cvs checkout -r v0-5-0 asterisk. We also have a bit more history for the modules listed above, from before the loss of the CVS repository. This history is reconstructed from the FTP releases, so the granularity is not the same as for more recent changes. Going forward, we also expect to start using branches for major releases, with bugfixes only on these branches and so forth. Thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: OT : For the SQL gurus..
SELECT *, LENGTH(code)
FROM a
WHERE code = left('00442085673456', LENGTH(code))
ORDER BY LENGTH(code) DESC
LIMIT 1;
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Monday, November 10, 2003 9:33
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: OT : For the SQL gurus..
Reinhard Max wrote:
On Mon, 10 Nov 2003 at 13:34, WipeOut wrote:
I guess I am going to have to look through the table multiple times
dropping the last digit on each select until I get a result..
You could also try this one to see which one is faster:
SELECT *, length(code)
FROM a
WHERE code = '00442085673456'
OR code = '0044208567345'
OR code = '004420856734'
OR code = '00442085673'
OR code = '0044208567'
OR code = '004420856'
OR code = '00442085'
OR code = '0044208'
OR code = '004420'
OR code = '00442'
OR code = '0044'
ORDER BY length(code) DESC;
Again, the first row in the result set is the one you are after.
cu
Reinhard
That works quite well.. result in 0.11 seconds..
I just have to work out now how to create the query in my code.. :)
Thanks a lot..
___
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[Asterisk-Users] RE: [Asterisk-Dev] Upcoming Major CVS Changes
As for *why* we're doing this change, and what advantages there will be and how to use them, Thornston will send that information in a separate e-mail. Back in February, the original CVS repository was lost. With that, all change history from before then was lost as well. I have reconstructed the parts of that previous history that I could, using the old FTP releases, and then merging in what is in the current CVS repository. I addition to this, all the releases that has been made to date (0.5.0 for Asterisk, for instance), have been tagged. This means you can check out Asterisk 0.5.0 by just doing cvs checkout -r v0-5-0 asterisk once the CVS respoitory has been modified. Going forward we also expect to have properly tagged and branched releases, with real bug-fix branches -- what many people would consider STABLE branches. Look for a bit more formalized release process in this regard. Note that for those people that have been using the -D option to CVS to get at specific versions that they feel comfortable with, this *will still work*. All the time stamps in the CVS repository will be *unchanged* after this happens. But all the internal release numbers will have changed, so you will have to get fresh check- outs. Note that only the modules asterisk, gastman, gnophone, libpri, zapata and zaptel are being replaced -- any other modules will not be affected at all. Thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music Onhold Configuration
MPG123 is not included in Asterisk... Download the package: http://www.mpg123.de/cgi-bin/sitexplorer.cgi?/mpg123/precompiled/ Install using: rpm -ivh mpg123-0.59q-1.i386.rpm Copy the file mpg123 from /usr/local/bin to /usr/bin You no longer need to copy it from /usr/local/bin to /usr/bin -- Asterisk will look for it in either place. Thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2 Trunking confirmation?)
No, you actually don't need to use a context in the peer. Asterisk will ? leave it up to the far end to decide what context to use. We use it to avoid any possibility of confusion in the process, but it is not necessary. In fact, I just verified this with the master himself and we will no longer tell our customers to use a context in their peer. And, given that a context= entry is not used for a peer, only for a user, it also goes to say that if you use a friend, context is still only used for the *inbound* portion of it. So I just don't really see why you so strongly recommend against friend entries as opposed to having both a user and a peer entry that are otherwise the same... Thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2 Trunking confirmation?)
and the peer simply has the required information: [NuFone] type=peer secret=his_secret context=NANPA host=switch-1.nufone.net Uh. Why would you want to specify a context for a peer at all...? Aren't those used only for inbound anyhow? Thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2 Trunking confirmation?)
On Thu, 2003-10-09 at 15:20, Thorsten Lockert wrote: and the peer simply has the required information: [NuFone] type=peer secret=his_secret context=NANPA host=switch-1.nufone.net Uh. Why would you want to specify a context for a peer at all...? Aren't those used only for inbound anyhow? What if you wanted that specific user to drop into a specific context so you could tailor what was accessable for that user. But that would be an *inbound* again -- the question was why specify a context for a *peer*... Not for a user or friend, where inbound is possible... Thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with GPL license of Asterisk
Uh. This is *not* how the GPL works. The GPL clearly states that anything you write that links with GPL will be covered by GPL, and thus if you release it in binary form you are also obliged to release it in source form. The same goes for modules for Asterisk (unless you have a non-GPL license for it). To achieve the kind of scenario you describe you'd need a BSDD license, or something close to that. Thorsten -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Troy Settle Sent: Monday, September 29, 2003 9:59 To: [EMAIL PROTECTED] A few answers: 1) if your application is not released to a 3rd party, you do not have to make the source available 2) if you build your application as a module that loads into a stock asterisk server, you do not have to disclose your source 3) if you need to make changes to the core in order for your application to work, you'll need to disclose source for your changes to the core, but not for your application. This sounds horrid, but it's not too bad, as your simply augmenting the core API and keeping your goodies in the binary only portion of the release. With that said, if you're writing an application that you would like to sell, your IP lawyer should be able to easily decipher the GPL and advise you as to which parts of your code need to be made public. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Monday, September 29, 2003 8:38 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help with GPL license of Asterisk I would appreciate some help with this. I read the GPL license and basically it says you can do whatever you want with the software (sell, modify) as long as you include the source code, the License and make any changes you make available in the same manner to all others. My questions is this: If I develop an external application (say a Call Center application or a GUI management application) that uses Asterisk data is that also GPLd? I understand if I add code to Asterisk, but what about external interfaces? Where is the seperation here of the Cathedral and the Bazaar? Thanks -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: SIP i.e. Is something broken?
To roll back only the affected stuff for SIP negotiation, I would recommend: make update cvs update -j 1.181 -j 1.179 channels/chan_sip.c Note that the second line should only be executed *once*. Once this is fixed in CVS, you should *remove* channels/chan_sip.c to make sure the changes done by the second line above are removed again. By using this recipe you will be able to run the latest and greatest version of Asterisk, and still have working codec negotiation for SIP. Note that this negotiation problem only happens with certain devices, including Grandstream. Thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival Problems
Did you Answer the call before calling Festival? Thorsten -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, September 24, 2003 11:13 To: [EMAIL PROTECTED] Run the command festival Give it the command (SayText Would you like to play a game?) Does it say anything? If not, then there's a problem Festival. Type (Quit) to quit the festival app. On Wed, 2003-09-24 at 08:22, Bryan Nolen wrote: I am trying to use festival (latest version 1.4.3) I have downloaded all the files needed and patched it with the provided diff. festival does work and does tts fine. but when I call Festival either from an extention or an AGI script, I get this in my asterisk messages log, but no sound on the channels (H323 or SIP) - they (the clients) just say trying and then hangup... Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 304 (festival_exec): Text passed to festival server : Hello 1010 Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 381 (festival_exec): Passing text to festival... Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 400 (festival_exec): Passing data to channel... Sep 24 23:05:33 WARNING[606231]: File app_festival.c, Line 410 (festival_exec): Festival WV command Any ideas? Bryan Nolen Lead Developer http://Arc.Net.AU http://cdonline.com.au ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cdr_mysql
Did you actually turn on debug output in /etc/asterisk/logger.conf? If not you won't see any debug output anywhere. Also, what does your cdr_mysql.conf look like? Does it have a [global] just before the configuration statements? Compare it to the (updated) cdr_mysql.conf.sample that you got when yuou did your CVS update... Thorsten -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tais M. Hansen Sent: Monday, August 18, 2003 12:34 To: [EMAIL PROTECTED] -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 18 August 2003 18:19, Tilghman Lesher wrote: On Monday 18 August 2003 11:08 am, Tais M. Hansen wrote: Is cdr_mysql broken in latest CVS? It builds and loads fine but it doesn't insert cdrs in the database and there's no debug output at all. No, it should be just fine (works for me!). Turn on debugging output and post what you get. There's nothing cdr_mysql related debug output. $ asterisk -vgcdf (loads of output, mostly pri) *CLI load cdr_mysql.so Loaded /usr/lib/asterisk/modules/cdr_mysql.so = (MySQL CDR Backend) ... That's it. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/QQAT2TEAILET3McRAinLAJ9dZiIzQ2bo3btaCuBTuO80btwI6wCeKK7D GyNgQch0t+LKnpHFRpqGCCk= =6nl2 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
