similar to: SugarAsterisk vs. ________

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 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

Hi, I am new user on asterisk (for that matter linux) and i have lot of embedded programming experience. We have a new project from our client, to design a box that takes the telelphone line as input and route the line to the respective user with different ring tones. The box should be programmed by the users with buttons. Features. 1. I should be able to store some .wav files for different

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

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

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

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 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'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

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 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

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123

I ran dovecot -a and the blizzard of data seemed ok to my limited knowledge. Is there another log I should look into to trace this error down? Dovecot and system info: thufir at dur:~$ thufir at dur:~$ dovecot --version 2.0.19 thufir at dur:~$ thufir at dur:~$ cat /etc/lsb-release DISTRIB_ID=Ubuntu DISTRIB_RELEASE=12.04 DISTRIB_CODENAME=precise DISTRIB_DESCRIPTION="Ubuntu 12.04.1

How do you load the contact list? It's a database? Sqlite3? https://wiki.asterisk.org/wiki/display/AST/SQLite3+astdb+back-end I'm not clear on what this specific database does. If it's not this specific database which has contact information, which database does? thanks, Thufir

On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: > A "sip set debug on" will give you more info on why you are getting the > 404. It probably has to do something with your context/dialplan. on tleilax: tleilax*CLI> tleilax*CLI> sip set debug on SIP Debugging enabled tleilax*CLI> on doge: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:devries at

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 :