similar to: realtime and save ip server in database

Greetings. Attempting to connect Asterisk to LDAP database using res_config_ldap module. While trying to register sip client (Ekiga softphone), according to slapd.log, asterisk connects to LDAP server, asks for some attributes to modify (they do exist, and asterisk user has all permissions to do that, etc). And then asterisk application just crashes. Without ldap (using just static users'

Any one knows where to install chan_sccp for asterisk 1.4 ???????. Please guide me from where can I download the asterisk 1.4 sccp channel driver and how to install it because I tried to get chan_sccp-mayday05.tar.gz When I trying to install it ,error happened like this. Please help me how to solve this issue. [root@XPL t]# cd chan_sccp [root@XPL chan_sccp]# make clean rm -rf

Hello *, is the rtpip patch still valid for asterisk 1.6 (with some code changes, obviously)? https://issues.asterisk.org/view.php?id=8161 Or, in asterisk 1.6 there is an alternative to using it? This is the difffile I produced for chan_sip.c in asterisk 1.6.2.11 --- chan_sip.c 2010-10-12 13:47:49.000000000 +0200 +++ chan_sip.c.orig 2010-10-12 13:47:27.000000000 +0200 @@ -987,9 +987,6 @@

Some time ago someone posted a nice script to monitor HTB classes, classmon.pl. A friend of mine ported it to CBQ, and attached is the result. Suggestions are welcome. Rubens > #!/usr/bin/perl > > # Classy CBQ Operations Monitor...in Perl > # Based on classmon.pl by Toby Cantor > # By BLFC > > # The following short command line options are parsed as you might expect. >

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

plain text document attachment (xx-paravirt-time.patch) General time changes for paravirt_ops. Signed-off-by: Steven Rostedt srostedt@redhat.com Signed-off-by: Glauber de Oliveira Costa <gcosta@redhat.com> Index: clean-start/include/asm-x86_64/time.h =================================================================== --- /dev/null +++ clean-start/include/asm-x86_64/time.h @@ -0,0 +1,18 @@

plain text document attachment (xx-paravirt-time.patch) General time changes for paravirt_ops. Signed-off-by: Steven Rostedt srostedt@redhat.com Signed-off-by: Glauber de Oliveira Costa <gcosta@redhat.com> Index: clean-start/include/asm-x86_64/time.h =================================================================== --- /dev/null +++ clean-start/include/asm-x86_64/time.h @@ -0,0 +1,18 @@

I am using Realtime on Asterisk 11.5 with a SQLite3 backend. While everything seems to be working fine I keep getting this error on my log files: [2014-02-17 19:55:18] WARNING[20569] res_config_sqlite3.c: Could not execute 'UPDATE "sip_buddies" SET "ipaddr" = '192.168.2.23', "port" = '5060', "regseconds" = '1392692118',

I'm trying to get asterisk working w/ packet8 (incoming and outgoing) and a dta310 so I can have more control over voicemail. I've looked at the data stream coming from the dta310 and from packet8, but I haven't managed to get the phone to actually place invites through asterisk. On the asterisk end with chan_oss.so, I can make it dial and I hear ringing and the first second of

Hi All, Has anyone else noticed that when a sip phone sitting behind a nat registers to asterisk using realtime database, the private IP of the phone is put into the fullcontact field instead of the public contact IP. The database has the correct public IP in the ipaddr field and correct port number in the port field, which is actually what asterisk uses to to contact the device. This

I''m trying to get link_to_remote to assign a CSS class to the link it generates. My code is as follows: <%= link_to_remote "Show Full Info", :update => "fullcontact" + reparray.last.to_s, :url => "/cm/full_contact/" + reparray.last.to_s, :classname => "contactlink" %> Which produces this HTML: <a href="#"

I am getting messages on the console about Registered SIP ... expires 60 How do I increase that 60 to 3 minutes??? I have tried in [general] of sip.conf to set expirey=300 defaultexpirey=300 nothing seems to affect it. Thanks, Jerry

Hi, We are considering samba 3.0.1 as a replacement for our old-hag Windows NT4.0 Server PDC. However, we do want some policy control, and from the samba 3.0.0 documentation (23.3), it says the only current functional policy is the password expirey. Is this still true in samba 3.0.1? Can an NT4.0 server be a backup domain controller to samba 3.0.1 PDC? That way we can keep the NT4 machine

Hi All, I'm getting a strange warning message when I perform a MYSQL data lookup. The operation performs fine, I retrive the data I'm looking for and continue on through the dial sequence without an issue. I'm wondering if this warning message is something to be concerned about, can't find any info about it. warning message: Mar 28 15:55:40 WARNING[27481]:

Hi We have some users who are supplying very long, broken contact details (from Cisco 7912 phones): Apr 25 11:29:46 WARNING[1480] chan_sip.c: No closing bracket found in '1st Floor Scanner - 137 <sip:xxxxxxxx@yyy.yy.yyy.yyy:5060;user=phone;transport=' Apr 25 11:29:46 NOTICE[1480] chan_sip.c: '1st Floor Scanner - 137

Hi We're using asterisk 1.4.17 with RealTime so our port and fullcontact values in out DB get updated dynamically. We use snom handsets and always set the network identity (port) in each phone to something in the 10000 range, so that each phone in a single location has a different port. When we look in the DB the location always has the port we set but the port value is often something

HiI want to load balance SIP calls between two(or more) Asterisks with only DNS SRV. I used bidirectional sync Unison to synchronize configuration files and internal database file between two Asterisk boxes.The problem is when a calls come to Asterisk1 but SIPendpoint is registered on Asterisk2.How we can check a SIP endpoint is registered or not and what is Contact?information in Dialplan ?

I'd love to understand what the function of the peer list returned by 'sip show peers' is for, especially when Realtime is used. If I start Asterisk with realtime enabled, a 'sip show peers' yields none. As each peer (phone) registers, or a call is made to the peer, Asterisk adds them to the list returned by 'sip show peers'. Correct? Apparently Asterisk doesn't

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

If anyone's interested, I've just put together a sip realtime patch, figured anyone that uses realtime may want to have a look at it. The patch basically takes the stuff asterisk updates (fullcontact, ipaddr, port, regseconds, and username) out of the sippeers table and puts it in it's own table. For those that are using multiple tables, this allows you to create a view of those