similar to: Asterisk and OpenLDAP

Just a little questions, what's the difference between asterisk 1.8 and asterisk 11? Best regards.

Hello all, I am going to register asterisk sip users through active directory accounts LDAP (that is a separated server with ip : 192.168.11.17) So I have followed the below link as well: https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html

Hi, I'm trying to get Asterisk running with LDAP to be able to authenticate sip user registrations. I'm using Asterisk (1.8.13.1~dfsg1-3+deb7u3) on a Debian server. Unfortunately I wasn't successful so far. My res_ldap.conf looks like this (so pretty minimal): --- [_general] ;url=ldaps://ldap.chaotikum.org url=ldap://ldap.chaotikum.org protocol=3 basedn=dc=chaotikum,dc=org [sip]

Hi all, I've updated Asterisk trunk LDAP schema [0] [1] to include queues and other attributes needed for a working LDAP backend (I'll open a bug to include these changes on svn). SIP users and dialplan are perfectly working, but when I call a queue the whole Asterisk (1.6.2.0) crashes: on extconfig: [settings] sipusers => ldap,"dc=nodomain",sip sippeers =>

Hi, I have a question, let''s say I have a zvol named vol1 which is a clone of a snapshot of another zvol (its origin property is tank/myvol at mysnap). If I send this zvol to a different zpool through a zfs send does it send the origin too that is, does an automatic promotion happen or do I end up whith a broken zvol? Best regards. Maurilio. -- This message posted from

Greetings listers. I'm running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. For the most part I can make and receive calls fine except for these 3 issues: 1. When I call an external conference, the call never bridges and hangs up after 60-90 seconds. 2. When I call another number there is a

> Date: Wed, 28 Jan 2009 13:11:19 -0600 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1? > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <D32AD473FC574B41AE6A842E46549174 at db0005> >

Hi everyone, in ldap realtime sip peers i need "fullcontact" set to "sip:exten@userip" for asterisk to correctly match the peers (at least for the natted peers to reach them)... anyway, how do I populate fullcontact "on the fly" with information from exten and userip? of course, i could just do it staticaly on ldap but since the info is already there why not

Hi, I had a Trixbox 1.4.18 that I "yum update"d to 1.4.22.1. Now, I seem to have a huge problem with phones not staying registered (registrations worked perfectly at 1.4.18). I phone will register the first time I plug it in, and then once you make a call and hangup (or sometimes even during the call) all the lights will go orange meaning a lost registration. Every so often the lights

Hi. I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and for some reason it's simply not doing it. I've even resorted to reading the source code to try and work out what I'm doing wrong... In channels/chan_sip.c I find: * SIP Dial string syntax: * SIP/devicename * or SIP/username at domain (SIP uri) * or

Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet

Hello all, I try to use a agi script to get a variable from * und put them into a script which gives me another variablke and put this in *. My problem is now it seems the var ID is empty coz i always jump into the result 0 loop. The $MSN should be in the SetCIDNum. #!/usr/bin/php -q <?php include("/var/lib/asterisk/agi-bin/phpagi.php"); $agi = new AGI(); $ID =

Hi I want to authenticate the asterisk users from the LDAP directory server not from the sip.conf. I tried to use the astirectory-1.2<http://www.asterisk-ev.org/download/astirectory-1.2-0.3.tgz>. But i am not able to configure it properly. If somebody used it then please help. In the res_ldap.conf file i made the following entries. I am using my normal username and password to connect my

Hi all... Does anyone know if it is possible to override sip.conf settings in extensions.conf (for example: session-minse=90) without needing to create an overarching peer in sip.conf and selecting it specifically in the dial plan? I'm on the 1.4 stable code base and looking to implement session-timers on certain call flows in a modular dial plan. Thanks, Josh Fuller josh.fuller at

Hi guys, I'm trying to use Asterisk with LDAP integration. I created some schemas and it seems to work fine for sip.conf replacement. When I try to register a softphone to test the service, it seems ok from the softphone point of view (user registred) but when I do a "sip show peers", no one is registered (nor sip show subrscriptions, users...) I put my Asterisk on full debug and I

Hello, I have something like this: SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN After calling from SIP to PSTN (and from PSTN to SIP too) I can't hear anything only in my SIPUSER. At the PSTN side everything is OK. I have another network with another h323/sip (in the place of asterisk) and there everything is OK. In AUDIOCODES logs I see that everything goes

Hello, is it possible to rout ALL RTP Data over Asterisk, like SIP1 <---RTP---> Asterisk <---RTP---> SIP2 I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) Thanx, Kalle

I'm kind of at a loss to diagnose problems like this, yet we get them a lot. - The ATA (Thomson 784 in this particular case) is logged into the Asterisk server. 'sip show peer' shows their IP address, port, and useragent. - The ATA is connected directly to the internet (no NAT, but the sip configuration has nat=always) and logs in to our server, which is also directly connected to the

Thanks Denis, I was looking for the option 'dns:x' in the wiki but I didn't find it. Now it works. I used    log level = 3 auth:3  dns:0 auth_audit:3 gives me unknown class message But where I can find a complete list of classes for log level? I'll also give a try on the last version of samba with json. Thanks again Giuseppe On 1/18/2018 4:52 PM, Denis Cardon wrote:

On 18/06/2014 04:17, Charles Lepple wrote: > On Jun 13, 2014, at 2:53 AM, Giuseppe Corbelli <giuseppe.corbelli at copanitalia.com> wrote: > >> As said in previous mail, I just finished a first working version of a driver for the UPS found on ASEM PB1300 device >> (http://www.asem.it/prodotti/industrial-automation/box-pcs/performance/pb1300/) >> Linux only, accessed