Displaying 20 results from an estimated 2000 matches similar to: "realtime sipusers and rtcachefriends... bigheadache!!"
2007 Jan 11
1
realtime sipusers and rtcachefriends... big headache!!
hi folks,
I am using asterisk 1.2.13 (debian etch).
My customer's sip accounts are stored in realtime sipusers.
I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes
Each account has nat=yes
Now, I have lot of problems.
for example, when I change the 'secret' field of a user in the database, it
doesn't
get reflected in Asterisk, who is still expecting the old
2007 Jan 15
1
ANY ADVICE ON THIS????
Hello List,
I am stuck with this problem for several days... anybody can give me a hint
on this??
I know many of you dealt with problems similar to this, how did you address
this??
Thanks in advance!!!
-lars
---------- Forwarded message ----------
From: Lars Knopf <lars.knopf@gmail.com>
Date: Jan 11, 2007 1:12 PM
Subject: realtime sipusers and rtcachefriends... big headache!!
To:
2007 Dec 19
0
Asterisk Realtime SIP rtcachefriends
I haven't been able to find this on the wiki:
If rtcachefriends=yes. When will a change to a realtime user/peer take effect? Next registration? Never?
It's also not clear to me what the purposes of rtautoclear and ignoreregexpire are. The only info I have found is the comments in the sample config file. Sounds like rtautoclear will save memory if I have lots of peers. Is there any
2005 Jul 11
2
Enabling rtcachefriends prevents phones from calling each other
With rtcachefriends = yes in sip.conf, my SIP phone registered to Asterisk Server A cannot dial another SIP phone registered to Asterisk Server B. The error message is: "Cannot create channel of type SIP (Cause 3 - no route to destination)".
The two phones _can_ call each other if I set rtcachefriends = no. The common extensions.conf simply uses Dial(SIP/extension) to dial extensions.
2010 Aug 03
1
sip.conf register in realtime DB
Hello list,
scrambling different pieces of info together I've come with the following :
I want to have my "register =>" statements in a MySQL-database, so I've
made the following table.
table ast_config :
id 1
cat_metric 0
var_metric 0
commented 0
filename sip.conf
category general
var_name register
var_val username:password at sip.provider.net
In ext_config
2008 Nov 12
1
What are the minimum realtime fields for sipusers?
I'm trying to get sipusers working with a realtime odbc database on
Asterisk 1.6. We have sippeers working from the database, but need
sipusers to be in a separate table for other implementation reasons.
sip show user test load returns results from the database.
CLI> sip show user test load
* Name : test
Secret : <Set>
MD5Secret : <Not set>
2009 Aug 25
1
Realtime with "rtcachefriends=no" problems...
Hello there!
I was testing Asterisk for the last two weeks using the Realtime driver
for MySQL, and leaving "rtcachefriends=yes" configured to enable MWI.
Today I started making additional tests with "rtcachefriends=no" because
we will probably need to use Asterisk without this cache.
For some strange reason, calls stop to get routed between the SIP clients.
I've
2006 Jan 18
0
rtcachefriends and REALTIME + MWI
Hi,
Is there something wrong with REALTIME (ARA) when used with
rtcachefriends parameter?
In my sip.conf (Asterisk 1.2.0):
rtcachefriends=yes
rtupdate=yes
rtautoclear=yes
Desired configuration is realtime configuration (via odbc) for SIP
phones + MWI. Realtime means the following: when I make changes to db
they should apply with no extra commands executed in CLI.
In order to use MWI with
2009 Jul 17
1
Realtime difference sipusers sippeers
Hi,
I would have expected that peers of type friend ( for example an
SIP-phone) registring at Asterisk will be searched in sipusers.
But the peers will be searched in sippeers.
May be sombody can explain the difference?
Asterisk 1.4
thanks
Thomas
2010 Apr 17
1
Realtime changes not reflected realtime
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<font size="-1"><font face="Helvetica, Arial, sans-serif">Hello list,<br>
<br>
Using Asterisk 1.4.25.1<br>
Using realtime sip_buddies<br>
<br>
I notice
2008 Jan 01
1
With rtcachefriends=yes, when do realtime changes take effect?
I asked this question last week and never got an answer. I also didn't find the answer in the wiki.
I think it would be nice if asterisk would check the database again if the user re-registers, but it doesn't seem to do that. A periodic update would be ok too, but it doesn't seem to do that either.
It seems like changes never happen until a reload.....if that is the case then
2011 Jan 02
1
Realtime SIP, multiple AX servers question
We have several Asterisk servers (1.6.2.15) all configured for Realtime, all backed by the same database. The Asterisk servers are all listed under DNS SRV records, and SIP ATAs find us this way.
Normally, no matter which Asterisk server an ATA connects to, we get our database fields filled out correctly, such as "regseconds", "lastms", "ipadr", etc. However, with
2014 Jun 10
1
Asterisk realtime peer registration
Hello there
I'd like to use sip users and peers realtime.
I think I done all I need to get asterisk works fine in realtime:
res_odbc.conf configuration.
extconfig.conf
sippeers => odbc,asterisk,sipclient
sipusers => odbc,asterisk,sipclient
sip.conf
[general]
rtcachefriends=yes
The sipclient table as suggest in this article: SIP Realtime, MySQL table
structure (
2006 Dec 18
0
Re: Best way to access MySQL data from dial plan
I'm not sure that any solution with the MySQL dialplan command is going to be ideal. You also can't nest your queries, ie the connectid/result id seems to only be good for one resultset at a time... try doing something like findme/followme with that!
Doug.
> -----Original Message-----
> From: kjcsb [mailto:kjcsb@orcon.net.nz]
> Sent: Monday, December 18, 2006 11:33 AM
> To:
2013 Mar 05
1
multiple sipusers tables
Hi,
I have 2 databases DBNAME1 and DBNAME2. In each database I have a table called sipusers (so DBNAME1.sipusers and DBNAME2.sipusers).
Can I use both sipusers tables in Asterisk RealTime?
Something like this:
/etc/asterisk/extconfig.conf:
[settings]
sipusers => odbc,DBNAME1,sipusers
sippeers => odbc,DBNAME1,sipusers
sipusers => odbc,DBNAME2,sipusers
sippeers =>
2007 Oct 09
2
Asterisk Realtime woes
I have configured asterisk realtime to work with two servers and a seperate MySQL DB.
Each sip client registers which server it is connected to in the MySQL DB. This works great as long as the clients are
1. On the same network
2. Behind a NAT and connected to the same asterisk server as the caller.
However I need this configuration to work for "NAT-ed" clients on different asterisk
2006 May 05
1
Realtime, 2 server setup problem?
All, we're running realtime and ast 1.2.7.1 stable. The problem I'm
having is when you register a SIP device on ServerA, works fine, sip
show peer works fine. When you dial the SIP device from ServerB however,
it tries to dial, even does a MySQL query like it should but comes back
saying no route to dest.
When I do a 'sip show peer ..' on ServerB, the peer comes back fine, it
even
2014 Apr 24
1
Realtime integration: Unregistered clients showing as registered?
Hello all,
I've been testing a Kamailio Asterisk Realtime integration, and found a
strange situation.
My problem is that when using the integration, everything seems ok but
Asterisk does not see the clients as registered. Kamailio and the clients
report registered clients. Also calls fail.
In Asterisk cli sip show peers shows nothing but for example realtime load
sipusers name 660 shows the
2013 Feb 17
0
Can Cisco 5XX phones share asterisk phone directory?
Hi!
Please is it possible for Cisco 5XX phones to use asterisk/FreePBX phone
directories, and if so, how?
Thanks in advance!
On Feb 17, 2013 6:40 PM, <asterisk-users-request at lists.digium.com> wrote:
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2015 Feb 16
1
Asterisk 11.6. SIP realtime lost peers after 'sip reload'
Hi, list.
We have a problem with loss peers after 'sip reload', our configuration:
Asterisk 11.6-cert1, SIP realtime peers, sip.conf:
- rtcachefriends=yes
- rtsavesysname=yes
- rtupdate=yes
- rtautoclear=yes
When we do 'sip reload' , peers are removing from available.
Before `sip reload` :
srv-pbx2*CLI> sip show peers
Name/username Host