similar to: Asterisk 1.6.2.11 Now Available

Greetings, I am in the process of testing Samba 3.0.23a with our own passdb plugin. As part of mytesting I am trying to join the domin so here are the steps I take... 1 - get local sid /usr/local/samba/bin/net getlocalsid SID for domain JAZZY is: S-1-5-21-1016995387-3159270912-1426853295 2 - create group mappings [paulg@jazzy ~]$ /usr/local/samba/bin/net groupmap list Domain Users

Greetings, Is there any tips/docs on trouble shooting a custom passdb module? The module compiles and can be loaded without problems I can even access shares with my username and password. When I try to assign privilages so our tech members can join computer to the domain, I get a error message. Even with level 10 dumps it is hard to see where the problem is. Our user database contains all the

It like we may have a bug in tar that ships with Centos. It looks like -o is mapped to --same-owner. Some basic info.... uname -a Linux jazzy 2.6.9-34.0.1.ELsmp #1 SMP Wed May 24 08:14:29 CDT 2006 i686 i686 i386 GNU/Linux cat /etc/redhat-release CentOS release 4.3 (Final) % /bin/tar --version tar (GNU tar) 1.14 --same-owner try extracting files with the same ownership --no-same-owner

Hai Paul, Look at this: user=paulg,uid=2381 (from mount -t cifs //xxxx.xxxx.yorku.ca/homes /tmp/1 -o user=paulg,uid=2381,gid=1000,domain=AD.ONE.EXAMPLE.CA) Now, look at this : > idmap config * : backend = tdb > idmap config * : range = 3000-7999 > # - You must set a DOMAIN backend configuration > # idmap config for the ONEEXAMPLECA domain > idmap config ONEEXAMPLECA : backend =

Hello, I am testing Samba 4.9.7 before we upgrade our 4.8 domain member server. I am running into a weird permission error with our test server. My home directory is NFS mounted. The problem comes from a mail directory in my home directory.  I can't access it over SMB/CIFS, it gives me a permission error. From another Linux host that has our home directories NFS mounted, I can access it

Hello list, I need to record all calls. So I'm using application Monitor. Works good until someone transfers a callee to another internal extension. Example: A calls B A set B on hold A calls C A transfers B to C with SIP transfer (SIP REFER - with phone funktions and not Asterisk attended transfer). I found http://bugs.digium.com/view.php?id=0013538 . "corruptor" asked about this

The Asterisk Development Team has announced the release of Asterisk 1.6.2.9. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.9 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.6.2.9. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.9 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.6.0.21. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.0.21 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix to Monitor which previously assumed the file to

The Asterisk Development Team has announced the release of Asterisk 1.6.0.21. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.0.21 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix to Monitor which previously assumed the file to

Here''s what I''m trying to do. I have two divs, one holds about 20 small images of products and the other holds a single large image of one of the products. I''m trying to use the jazzy "new Effect." way of changing images but I can''t seem to get it too work. My script is replacing the image and then making the element disappear. What

Hi folks, I just cvsupped the latest -STABLE source code, and noticed that it doesn't seem to build. Here's what happens: --- root@jasmina> make buildworld -------------------------------------------------------------- >>> Rebuilding the temporary build tree -------------------------------------------------------------- rm -rf /usr/obj/usr/src/i386 mkdir -p

Hello, everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I get the following : [Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username mismatch, have <329909006666>, digest has <3291119600> [Sep 12 18:59:29] NOTICE[2066]: chan_sip.c:20082 handle_request_invite: Failed to authenticate device "0473990000" <sip:0473990000 at

Hi, I have a problem that the machine running asterisk 1.6.2.11 freezes unexpectly time to time. Sometimes it runs for 4 weeks without any problem, sometimes after a free it freezes again in 24 hours. But usually it runs normally for 1 month or so before it freezes again. I could not find any additional info in any file located in /var/log/ including asterisk's messages file. Verbosity

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Hello list, I'm using asterisk 1.6.2.11 with realtime SIP (mysql DB). I notice that when the SIP peer registers, the fields 'fullcontact', 'ipaddr', 'port', 'regserver', 'regseconds', 'lastms' are filled with values. But after a while, these fields become empty. Asterisk CLI shows : asterisk*CLI> sip show peers Name/username

Hello, I have a problem of DTMF duplication. I receive call from my provider with SIP protocol. These calls pass through an interactive voice menu, using the application Waitexten to enter a client code. The menu works fine, but sometimes I have DTMF duplication that prevent proper code entry. All DTMF come twice. my sip.conf ----------- [general] context=default allowguest=no

List, Over the last few months we have managed to bring the total number of issue on the tracker from 610+ to 537 (as of writing). While this is good news, we still have a number of open issues that require testers to help move them along. Below, I have posted the oldest 50 issues that are in the 'Ready for Testing' state. Basically, we are looking for more people to step-up and test

This is by no means a complete fix for RHBZ#836710, because I haven't completely got to the bottom of this bug yet, but it is an improvement. Rich. -- Richard Jones, Virtualization Group, Red Hat http://people.redhat.com/~rjones New in Fedora 11: Fedora Windows cross-compiler. Compile Windows programs, test, and build Windows installers. Over 70 libraries supprt'd

Hi all, I'm having trouble with H.323 outbound calls, * connects but there is no sound in both ways. I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729 licenses installed and this is onli one call. I tested my * with another ITSP over SIP and G.729 codec and there was all ok Here is my configs