Displaying 20 results from an estimated 30000 matches similar to: "Asterisk Realtime SIP rtcachefriends"
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
2008 Mar 13
1
sip.conf help, inbound calls fall to last specified context
First of all, if Asterisk is the client and it must register to the other side, does the peer\user entry have to be in sip.conf, or can it be in ARA?
Second, why do all calls fall through to the last context specified, whether in that peer\user definition or not? I'm assuming it's a typo somewhere, but I can't find it. I had a full sip.conf, but axed a lot of the fluff trying to
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
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
2007 Jan 26
0
realtime sipusers and rtcachefriends... bigheadache!!
----- Original Message -----
From: "kjcsb" <kjcsb@orcon.net.nz>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: Wednesday, January 24, 2007 8:24 AM
Subject: Re: [asterisk-users] realtime sipusers and rtcachefriends...
bigheadache!!
>
>> hi folks,
>>
>> I am using asterisk 1.2.13 (debian
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
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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2009 Sep 18
3
DUNDi + SIP Realtime
Good afternoon gentlemen (and ladies).
A costumer of mine has many servers and each one maps their SIP extensions to the others via DUNDi. It works like a charm. SIP extensions can only register at one server, the one they "belong" to. In case one extension wants to call other that is registered in another server, DUNDi takes care of that by calling the other server using IAX2 and G.729
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
2005 Aug 16
6
realtime caching
Can anyone shed some light on realtime caching?
My desired behavior is that MWI works with realtime
voicemail/sip/extensions AND updates to the database take place on the
next call to the extensions.
Right now I have rtcachefriends=yes, and MWI works, but updates to the
database for a cached user seem to still require a reload.
It is my understating that removing rtcachefriends will
2010 Mar 01
1
rtcachefriends & qualify
[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify
is incompatible with dynamic uncached realtime. Please either turn
rtcachefriends on or turn qualify off on peer 'gerrie'
Am I correct that when I turn on rtcachefriends in sip.conf,
database-changes in my MySQL-DB will not be reflected untill a reload ??
Am I correct that when I turn off qualify in my realtime
2009 Aug 04
0
SIP server behind NAT
Hello.
I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage
to make outbound calls, but the communication drops off after 30 seconds
or so.
I'd really appreciate having some assistance from the mailing list on
this issue.
So, I'm having an Asterisk server behind a firewall and Zoiper
softphones on SIP connecting to Asterisk on the same local area network.
The
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
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
2005 Oct 10
1
Realtime regseconds update
Hi guys, im using realtime and I want to show registered users or
online users on a webpage and offline users. Im taking regseconds
field to make this happend
If regseconds value is 0 then user appers offline, it regseconds is
something else then its online, but sometimes this works and
sometimes it does not. Im using the following options
rtcachefriends=yes
rtnoupdate=yes
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
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
2005 Jul 19
0
Asterisk with Realtime registration problem
Dear All,
I am currently working on asterisk cvs-head version in order to use
realtime with mysql, 2 asterisk servers with duplicate mysql databases,
one asterisk server is serving the sip phones and the data is logged to
the database and replicated to the other asterisk database, when the
first server fails though it has the sip phones data in it's database
the sip phones need to re-register
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:
2010 Dec 25
1
asterisk realtime & calling sip users
Hello
We have recently upgraded to Realtime engine (sip buddies and
extensions) and now have problems with calling local SIP users.
I have rtcachefriends=yes but tried with 'no' and it's even worse.
(asterisk 1.8.1.1 + realtime mysql)
Here's an example:
User 1000 registers successfully and can then be called with
Dial(SIP/1000,30) successfully
After some time when I try to call