Displaying 20 results from an estimated 1000 matches similar to: "Asterisk with Realtime registration problem"
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 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
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
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
2005 Jun 30
2
Asterisk failover solution
If your phones are setup to connect to the asterisk box by name, then a
smart DNS server can just point phones to the backup box after failure.
However, since asterisk running on the backup box doesn't know about the
phones, this is only half the solution
________________________________
From: Mohamed A. Gombolaty [mailto:mgombolaty@noorgroup.net]
Sent: Thursday, June 30, 2005 8:30 AM
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 Jun 08
5
Xlite not communicating with Asterisk
Dear All,
I have downloaded the xlite version 2.0 for windows and I made the
following conf in the xlite itself as the document suggested in order to
make it work with Asterisk but still it doesn't work as a matter of fact
when I tried to make a tcp dump I can see no packets going between the
windows client and the Asterisk server at all, here is the my conf on
the xlite itself:
in the
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
2005 May 09
1
Asterisk + SER and NAT
Hi,
We are testing a SIP solution * + ser solution for a large implementation.
All the clients are nated.
When a client is dialing outside the domain (to a FWD sip account for
example) all is perfect ! ;-)
But ,when a call is done to a sip account, the client is ringing, then the
caller can hear the nated client very well, but the nated client does'nt
hear anything. RTP issue no ?
I've
2007 Feb 27
2
RES: asterisk-users Digest, Vol 31, Issue 115
Questions:
Does anyone have a really STABLE asterisk system running about one year
without need to restart the service or the SERVER ?
Does anyone have a production Call Centre saled that don't lockup and is
stable for 6 months ?
I'm asking this questions because we have choose Asterisk for our call
centre solution but, since the bugtracker only grows and people still want
to stuck more
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
2005 Jun 29
3
UK SIP Provider
Hi,
I'm looking for a reliable provider to use mainly for outgoing calls in the
UK, incoming isn't so much of a worry as I think I'm going to accept them
over ISDN.
Cheers!
Steve
--
Steve Foy
steve@narnian.org
2005 Jun 16
3
SER and Asterisk question
Dear All,
I am trying to make the phones always talk to each other (peer to peer)
using SER as a sip proxy, and incase the call is not answered we will
use the voicemail of asterisk and other feautures, I have done that
already, but in order to do so I found that I have to make the users
dial different exten numbers, here is an example:
user with exten 666 wants to call 999 .
666 dials 1999 and
2006 Oct 16
0
SV: How do you like TrixBox?
I love TrixBox, with the custom config files you can tweak pretty much with TrixBox too, I have at least done some. Plan to do a plain Asterisk install later, but for now I learn a lot about the config files just with TrixBox. Some things might be a bit harder with TrixBox due to some of the premade dial plans, but can get it to work :-)
_____
Fra: asterisk-users-bounces@lists.digium.com
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
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 May 14
0
Problem with Asterisk 1.4 and Linksys Spa941/962
Hello,
Yesterday night we have upgraded our Asterisk from 1.2.32 to 1.4.24.1
with lipbri 1.4.10, dahdi-linux-2.2.0-rc4 and dahdi-tools-2.2.0-rc2.
Libpri and dahdi is only for dahdi dummy cause of the meetme function.
After the upgrade we had the problem that some Linksys spa941 phone at
one location could not dial out. incoming calls to the phones works
without any problem, but outbound the
2007 Jan 08
0
SIP rt load from db
Anyone know the command that tells * to load a sipfriend
from the realtime db rather than saying no such host? I've tried various
combinations of the rt commands:
rtcachefriends=yes;
;rtcache=yes
;rtAutoClear=yes
;rtautoreg=yes
;rtIgnoreRegExpire=yes
;rtupdate=yes
rtfromcontact=yes
Basically I have a group of 4 * servers all routing calls, but only two
are hearing the phones
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
2005 Jun 02
1
Newbie :Call Forwarding problem
Dear All,
I was trying to enable call forwarding, following the steps of the link
on voip.org regarding this issue it doesn't work and the phone I am
trying to implement on is still ringing. below is my conf in
extensions.conf and the CLI output during the process.
My configuration is :
exten => _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2})
exten => _*5X.,2,Hangup
exten =>