Displaying 20 results from an estimated 900 matches similar to: "rtcachefriends & qualify"
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
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 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
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
2006 Apr 12
1
Where is the difference sip.conf - Real-time ?
I have two phones (111 and 112) on a LAN, and I have on a users site a
phone 333.
phone 111 uses sip.conf, while 112 uses real-time set-up.
111 can call 333 AND the audio is working
112 can call 333 but audio is just white noise.
333 can call 111 or 112 and audio is working.
The phones are identically set-up (just user name = phone number and
password are different)
sip.conf (for 111 - all
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.
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
2010 Sep 28
2
NAT issue (i think?)
Hi All.
got this problem that IP phones could not re-register to my server. even
if device is power cycled it still would not register. the solution i
found was to change the SIP port settings on the phone and it will
register. but after registration expires and its time to re-register the
same thing will happen, so i have to update the port settings again just
to make it work which is
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
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
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
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
2003 May 20
3
Startup problem
What is this?
chan_iax2.c line 4695 (build_peer): Unable to support trunking on peer 'lamas-tigris' without zaptel timing
codec_g729b.c Line 413 (load_module): Unable to initialize va stuff: -1
This is why I can't start asterisk in the background
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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 Dec 05
1
SIP-Realtime and sip reload
Hi,
I use SIP-Realtime to store my SIP-users and I keep the informations
about the SIP-Providers my Asterisk registers to in sip.conf.
I'm running into the following problem. If I set rtcachefriends="yes"
because I want to use MWI and run a "sip reload" because I changed
something in sip.conf, Asterisk forgets about all registrations of the
users which are all unavailable
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:
2005 Feb 17
4
can't enable trunking :(
I have successfully installed and configured the asterisk, the incoming and the outgoing calls are working fine, its a tremendous solution :)
Now i want to enable trunking between two asterisk boxes, in the iax.conf i have put:
[karachi]
...
...
...
trunk=yes
...
...
...
everything seems to work fine but when i load asterisk it says:
--------------
Feb 17 10:59:14 WARNING[18726]:
2004 Dec 21
2
IAXTEL Configuration
I signed up for an IAXTEL account and have been trying, unsuccessfully,
to get it working. In IAX.CONF I have:
[iaxtel_out]
type=peer
host=iaxtel.com
username=USERNAME
secret=SECRET
auth=rsa
inkeys=iaxtel
[iaxtel]
type=friend
context=incoming
host=iaxtel.com
auth=rsa
inkeys=iaxtel
However, when I start Asterisk, I get the following warning:
[chan_iax2.so] => (Inter Asterisk eXchange
2009 Feb 12
1
1.6.1-rc1 errors
I am getting the following warnings on the CLI when loading Asterisk
1.6.1-rc1:
[Feb 12 12:32:34] NOTICE[22261]: timing.c:59
ast_install_timing_functions: Multiple timing modules are loaded. You
should only load one.
[Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders:
Failed to open /dev/dahdi/transcode: No such file or directory
[Feb 12 12:32:33] WARNING[22261]: