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Hi Does anyone have any rough idea how far away 1.8.3 is? We can't deploy 1.8 yet because of this issue https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and

The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and

I know in v1.6 its part of logger.c but I noticed this: https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=11625 However, it doesn't seem to ever been applied to any version of 1.4.x branch.. Nor can I figure out what it was applied to? This is over 3 years old, you would of figured it would have been applied to 1.4 at some point in time.. Any ideas? --------------...

...when i send a call via a sip-channel, i would like to know the network-status of the foreign host immediately(at least within 5 seconds) so i can reroute the call without having to wait for a host that is probably dead... this seems to be possible with iax and CHANUNAVAIL, (http://bugs.digium.com/bug_view_advanced_page.php?bug_id=3360&history=1), though i haven't tried it. also i _need_ to use sip, iax (currently) is not an option. is there any mechanism in asterisk that allows to get the vital sip-status of a foreign host?! thanks for your input!!! ;-) regards christian

...he GL application remotely and display on nvidia driver enabled screen. It's not clear from the OP if that is the case here. The solution would be to add +iglx to Xorg command line, but gdm is hardcoding the Xorg parameters, so if you use gdm/gnome you are out of luck. https://elrepo.org/bugs/bug_view_advanced_page.php?bug_id=610 https://bugzilla.redhat.com/show_bug.cgi?id=1336014 -Thomas

In the list I found some messages that *8 doesn't work so well. Is there any possibility to create a extention that you can call, and if you are fast enough, pick up a number? (Also if you are outside your callgroup) like pseudo code: exten => 888, 1, EnterPhoneNumber() exten => 888, 2, EnterPass() exten => 888, 3, TransferCallToThisPhone() exten => 888, 103, Invalid()

Hi ! I just wrote cdr_sqlite.c, see http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001986 This little app creates automatically the sqlite database file in /var/log/asterisk/cdr.db, creates a table 'cdr' inside it and inserts all CDR records into this table. Please comment. I'll use this in my project DESTAR (http://www.holgerschurig.de/destar.htm...

...ot really a 'phone' but it seemed like the most appropriate way to list it.) Working and tested is the ability to call into the automated attendant and receive/process digits, as well as call another IAD channel and hold a conversation. A patch has been submitted at http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002982 (Asterisk bug ID 0002982) which improves the reliability of the IAD by hanging up calls when the IAD and Asterisk get out of sync. More details are in the bug report. My IAD2421 provides 16 analog pots channels via an Amphenol connector. It speaks MGCP with Asterisk and does...

I'm testing a TDM400P with FXO module to receive incoming calls from an analogue line and send it to a SIP device. To recieve callerid, I need to use cidsignalling=dtmf and cidstart=polarity. The problem is that when a call is finished, the TDM400P seems to require about 20 seconds to prepare for the next incoming call. If a new call comes in within 20 seconds after the previous call was

Hi, Can anyone give me any information at all to get app_intercept working? I've found these pages, but there is just not enough for me to get it going. http://www.pbxfreeware.org/archives/2005/06/new_download_--.html and http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002692 Thanks, G

...overed that the problem is not in chan_iax.c as I originally thought, it's actually app_mixmonitor.c. Basically when I use 1.4.26.2 with an ilbc codec between two asterisk servers trunked via IAX, with mixmonitor Asterisk crashes on me. Here's a link to the post: https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=16070. Can someone possibly assign it to the right application. I started looking at 1.4.24 which didn't crash, and upped my revisions until I found the problem. The bug was introduced in revision 204012 of 1.4, here's the info from the changelog: 2009-06-29 15:04 +0000 [r2040...

I've set up asterisk with my X100P as a home answering machine. Works great so far - answers the phone after 20 seconds, runs the phone tree, emails voicemail, etc. However, the one feature traditional answering machines have that I haven't been able to figure out is how to listen in on the call. Ideally I could just route through Console/dsp and hear it on my speakers. I've tried

I have Asterisk running on a Linux 2.4.x box with ztdummy. Once I did a modprobe on ztdummy I was able to enter into a conference room using my softphone clients. I'm using SJphone and Firefly. I have noticed a significant delay (1 to 3 seconds) while talking within the conference room. I have tried both clients, SIP and IAX protocols and various codecs. I have also tried it from different

Hi to all! I'm trying to create a context for integration with extensions.lua and libsql.mysql, but I'm not getting to run. When I reload the module pbx_lua.so the following error appears: [Feb 24 16:59:29] ERROR[30749]: pbx_lua.c:1249 exec: Error executing lua extension: error loading module 'luasql.mysql' from file '/usr/lib/lua/5.1/luasql/mysql.so':

Hi, I recently bought a 7910. I found out too late that it would not do SIP as I initially thought. Anyway before ditchingit for a 7960 I wanted to try it out, I read that the guys at http://chan-sccp.sourceforge.net/ had done some improvements to the original chan_sccp driver and having 80% functionality with this model. I have not been able to compile their driver and keep getting the

Hi Everybody Any significant changes to CVS HEAD over the last couple of days. I've got two asterisk boxes - both on public IP but one is dynamic. The one on dynamic IP registers at the other one - that part is fine. Calls going from the one with dynamic to the static one goes fine. Call the other way results now in: Failed to authenticate user "1101"

Hi everyone, Does someone know why the solution for bug 13115 (https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=13115) was made only for trunk? Having that this bug went solved more than a year ago, it means that all the 1.6.X.X branches have it applied already? Can this be backported to the 1.4 branch? This could be another good reason to upgrade to 1.6.0.16 after I do some good testing......

...could not be loaded. [Aug 11 22:00:01] WARNING[20173]: translate.c:654 __ast_register_translator: plc_samples 160 format f ------------------------------- I searched and the closest thing I could find was this closed bug report (which referred the person to this list): https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=15608 Any suggestions for possible solutions would be appreciated. -Seth -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090811/3bf4c778/attachment.htm

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