Displaying 20 results from an estimated 3000 matches similar to: "rtcachefriends and REALTIME + MWI"
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 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
2004 Oct 05
0
H.323: Inbound calls, incorrect remoteIpAddress
Hello,
I'm running Asterisk 1.0.1 (the same was with 0.9, 1.0). When it
receives inbound H.323 call it makes connection and uses local
127.0.0.1 address to send audio stream:
remoteIpAddress: 127.0.0.1
When making outbound calls from Asterisk it makes correct connection
to send audio stream. Is it a bug in h.323? Is there some more
settings to make in .conf files?
See detailed debug below:
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
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
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
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
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 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.
2005 Aug 05
1
Asterisk MWI and Realtime
I'm testing my asterisk system and the realtime backend. My Asterisk
build is rather aged, 03/18/2005 CVS. I have successfully moved Sip
peers and Voicemail boxes to the realtime database backend and this
works very well except for MWI. I don't seem to be able to get MWI to
work when I store the voicemail information in a database backend, from
a flat file it does work fine. I'm using
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
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 Dec 04
1
mwi for voicemail not showing up for realtime config.
Hello ppl,
Am using realtime odbc storage for voicemail, sip users/peers, static
for extensions and so on.
My issue is I am not getting MWI for any fones, even tho I've got
rtcachefriends=yes in sip.conf
WIth tcpdump, I always see the NOTIFY going as
Messages-Waiting:.no
Voice-Message:.0/0.(0/0)
even tho there are legitimate voicemails in the INBOX path for that
particular users in the
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 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
2011 Nov 23
1
MWI for non-subscribed Realtime peers?
Hi,
I have an Asterisk behind an OpenSIPS proxy. The proxy handles registrations and also SIP SUBSCRIBE for MWI. The Asterisk are configured to send NOTIFY to the proxy even when the SUBSCRIBE haven't been received. I can configure a user in sip.conf that works:
[az5134939706]
type=friend
host=xxx.xxx.xxx.xxx (IP of proxy)
port=5060
nat=no
mailbox=1234 at customer
subscribemwi=no
2006 Mar 14
0
MWI & Asterisk Realtime Architecture
Hi Everyone,
I am using real time asterisk architecture and have placed the following
in sip.conf:
[general]
notifymimetype=text/plain
checkmwi=10
rtcachefriends=yes
but the MWI doesn't work?!
Can anyone give me any pointers as to what the problem could be?
Thanks
ramin
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2006 Dec 04
0
mwi for voicemail not showing up for realtimeconfig.
Here's a link to it:
http://forums.digium.com/viewtopic.php?t=4363&highlight=
Regards,
Scott
-----Original Message-----
From: Scott Keagy
Sent: Monday, December 04, 2006 5:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] mwi for voicemail not showing up for
realtimeconfig.
A while back I posted a fully functional though somewhat elaborate
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