similar to: SQLite3 astdb back-end

That is the general idea. But how do i make it work? is there somewhere ready? On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkiniston at gmail.com> wrote: > Define your *72 and *73 extensions in your internal context, Have them set > a value in the ASTDB that you then check when dialing your handsets. > > The same can be done for call forwarding, store a number in the

Can I dial directly from the asterisk console with the Dial() application? or, is channel originate preferred: channel originate SIP/thufir extension 18003569377 at outbound thanks, Thufir

I would be very interested in using sipsak for something like this. What have you tried so far? -Thufir On Mon, 16 Jan 2017, Olivier wrote: > Thinking over my previous, I wonder if sipsak could be used to send > outgoing SIP NOTIFY messages. > Would both Asterisk and sipsak be able to share networks resources ? > > Thoughts ? > > 2017-01-16 14:10 GMT+01:00 Olivier

No idea on how to write it in my system. On Thu, Jan 11, 2018 at 12:17 AM, John Kiniston <johnkiniston at gmail.com> wrote: > There's some example code in the Dial-Users context of the basic-pbx > samples that might be of use in implementing it. > > They are checking a DEVICE_STATE to see if a phone is BUSY, You could > change it to be a database call or implement custom

when running asterisk -r, is there a way to turn off the messages? I didn't find the answer in the man page. thanks, Thufir

My understanding is that, given tables Alpha and Beta that the table which holds the 1-to-1 relation will be called Alpha_Beta (or is it alpha_beta?). So far so good. alpha_beta will have, lets say, two fields: alpha_id and beta_id. Seems that it''d be a good idea for those two fields to form a compound primary key, ensuring there are no duplicates. However, RoR doesn''t

I'm debating between a cloud PBX or, perhaps, rasberry pi. For a SOHO, maybe three hardphones, rasberry pi would suffice? I would be amazed, but, if so, great. thanks, Thufir -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160706/84a0ac7c/attachment.html>

I want to iterate ?nodes? and ?leafs? for a yaml document: thufir@ARRAKIS:~/projects/rss$ thufir@ARRAKIS:~/projects/rss$ ruby user.rb user.rb:6: undefined method `[]'' for nil:NilClass (NoMethodError) from user.rb:5:in `each_key'' from user.rb:5 thufir@ARRAKIS:~/projects/rss$ thufir@ARRAKIS:~/projects/rss$ ruby user2.rb user2.rb:5: undefined method `[]'' for

To use dovecot-postfix stack with thunderbird, do I put the configuration into /usr/share/dovecot/protocols.d/impad.protocol? That would seem to be how the stack is configured. "Thunderbird To use with Thunderbird, edit the file /etc/dovecot/dovecot.conf: protocol imap { ... login_greeting_capability = yes imap_client_workarounds = tb-extra-mailbox-sep }"

Why is dovecot using Maildir and not IMAP. Or is it using even using Maildir at all? Currently I'm using mailman, postfix and dovecot to manage a mailing list. Mail is sent to thufir at dur.bounceme.net which the "mail server delivery agent stack provided by Ubuntu server team" of dovecot-postfix handles fine, keeping it locally, so far as it goes. The mail ends up in

Looking at: http://appdb.winehq.org/objectManager.php?sClass=application&iId=10591 What are the steps to install this application? Yes, it's a garbage application, but I'd like to at least give it a go. Looks like msiexec apparently isn't the right approach. Should that be through wcmd instead? thufir at ARRAKIS:~/.wine/drive_c$ thufir at ARRAKIS:~/.wine/drive_c$ msiexec

ok, that's really all I need to know. Of course, if anyone else wants to throw in their two cents, don't let me stop you :) -Thufir On Wed, Jul 6, 2016 at 1:36 AM, Frank Vanoni <mailinglist at linuxista.com> wrote: > I'm currently using Asterisk 11.7.0 on a Raspberry Pi 2 Model B with > Ubuntu Server 14.04. > > Works fine! :-) > > Frank > > On Wed,

I'm looking at the <http://support.a-link.com/phonemate/IPU1.htm> phone because it works with Skype (from Linux), but can do SIP, too. Not necessarily asterisk related, but possibly. My networking situation might require IAX if I'm running Linux and want to use SIP, I'm not certain (Skype works fine). Putting that unknown aside for the moment, how does this phone work under

On Sun, 22 Feb 2015 08:32:26 +0530, Mitul Limbani wrote: > READ READ READ .... I know, I have the 4th edition and I've been reading it. Personally, I find it more general than specific, but I'll go back through that chapter, absolutely. thanks, Thufir

with func_odbc, in the definitive asterisk guide, they were suggesting the possibility that part, or perhaps all of, the dialplan could be written as SQL statement!? First off, that sounds like a good idea to me, but the tone of the authors was suggesting not so much, but that it was a personal preference. >From a naive perspective, why SQL statements at all? Why not just database config

I have the Asterisk book, it's enormous, the 4th edition as per http://www.asteriskdocs.org/. I'd like to do something like: http://www.voip-info.org/wiki/view/Asterisk+quickstart just to have two hardphones act as extensions and call each other. Is that a reasonable first task? I'm looking for the simplest litmus test for functionality possible. thanks, Thufir

So I found out my astdb database is boink (asterisk 1.8.something): [Jul 15 06:42:28] ERROR[18769] res_config_sqlite.c: database disk image is malformed # first stop asterisk before doing this, and do it on a copy: sqlite3 ./sqlite.db .dump |less PRAGMA foreign_keys=OFF; BEGIN TRANSACTION; /**** ERROR: (26) file is encrypted or is not a database *****/ ROLLBACK; -- due to errors (END) Anything

I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. linux-k7qk*CLI> linux-k7qk*CLI> sip show peer testcarrier * Name :

I'm looking into the dialplan specifics: tleilax:~ # tleilax:~ # cat /etc/asterisk/extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=DAHDI/r1 ; Trunk interface TRUNKX=DAHDI/r2 ; 2nd trunk interface TRUNKIAX=IAX2/ASTtest1:test at 10.10.10.16:4569 ; IAX trunk

On 15-04-09 12:06 PM, Chad Wallace wrote: >> but don't know where to put those lines. I have BABY defined as >> >channel variable: >> > >> >BABY = SIP/babytel_out >> > >> >but that seems circular, somehow. > You put them in the context for your clients... From what you show > below, I'd say they go in the "local_200"