From my experience prune does not take effect without reload.
And after reload ALL your phones are unreachable for 2 minutes!
Imagine you have several thousands devices unreachable for 2 minutes.
How much calls will fail during that time?
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing
Hi
The link you put in your email was the starting point that I used
myself. It should give you a good grounding of where to start and how to
proceed.
Ish
lore wrote:
Hi,
thanks a lot for the reply,
yes I would like to put data in a web interface (maybe php made better
if already done :)
Hi
We run production servers for various customers all using realtime with
web interfaces so they can change their own config whenever they want.
Prune works fine for us and we never do sip reloads (1.4.17)
Ish
Mindaugas Kezys wrote:
From my experience prune does not take effect without
On Wed, 2010-03-03 at 11:31 +0530, Chandrakant Solanki wrote:
Hi All,
I have followed below steps to enable echo cancellation.
# cd /usr/src
# wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2
# tar xjf linux-2.6.28.tar.bz2
# tar zxvf dahdi-linux-2.1.0.4.tar.gz
# ln -s
Hi,
For some reason I can't get Asterisk to produce debug or verbose tracing
output.
I connect to asterisk using 'asterisk -r'
Then issue the command:
Core set debug 10
And
Core set verbose 10
And it confirms that the correct level has been set.
I then attempt a connection from an x-lite
Hello,
I made 3 questions because they are linked and actually dealing with the same
need of releasing trunks after transferring 2 calls connected on trunks between
the same machines.
1) I have a machine with Asterisk 1.4 connected with a SIP trunk to a PBX.
A (on the PBX) calls B
- DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu:
Hi,
Carlos
I checked dmesg on my server and i found following message
what is meaning for this ? i cant understand
VPM400: Not Present
VPM450: echo cancellation for 128 channels
VPM450: hardware DTMF disabled.
VPM450:
On Tue, Mar 02, 2010 at 12:44:06PM +0100, Andreas Brodmann wrote:
Hi all,
We encountered a strange phenomenon when trying to upgrade from 1.6.2.0 to
any newer releases:
We use the following cli command to feed a wave/mp3 file into an existing
conference on an other serve:
On Tue, Mar 02, 2010 at 03:19:36PM +0100, Andreas Brodmann wrote:
Hi Tzafrir,
yes, I will have to 'anonymize' the dialplan,
Can you reproduce it with any other large dialplan?
is this list the right place
though?
That, or a bug report in http://issues.asterisk.org/ .
--
In logger.conf do you have verbose and debug on the console line? If not add
them and do logger reload.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 3, 2010, at 2:13 AM, Tim Culhane wrote:
Hi,
For some reason I can't get Asterisk to produce debug or
You might have to load the canceller with a modprobe (modprobe mg2 for
example)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of wins mallow
Sent: Wednesday, March 03, 2010 1:28 AM
To:
The problem is on your x-lite end. If you were speaking to Asterisk (even
incorrectly), it would at least indicate a bad connection. IMO, it is
better to use numbers for extensions as opposed to user1, but that is
irrelevant. Make sure the x-lite client has the IP of your asterisk box.
I am having a problem setting the caller ID that shows when I make an outbound
call over my PRI line. If I make a call from a SIP phone registered with the
Asterisk box the PRI is connected to the correct ID shows on my cell phone. If
I make a call from an IAX trunk connected asterisk box
On Wed, Mar 03, 2010 at 08:19:12AM -0600, Danny Nicholas wrote:
You might have to load the canceller with a modprobe (modprobe mg2 for
example)
It's 'dahdi_echocan_mg2' . And dahdi should modprobe it for you when
you run dahdi_cfg .
--
Tzafrir Cohen
icq#16849755
On Tue, 2010-03-02 at 14:42 -0500, Fred Posner wrote:
On Mar 2, 2010, at 2:37 PM, jonas kellens wrote:
Does Asterisk know when it hits a voicemailbox ?
When calling to a cell-phone or GSM, after some rings and no pickup you
arrive at a voicemailbox.
If Asterisk does not know it's a
Hi List,
I'm working on making one of my applications multi-lingual and
find that I have this problem. The SayDigits and SayNumber functions in
1.4.26.2 recognize but don't process the CHANNEL(language) function. Here's
a snippet to verify.
exten = 317,1,Answer
exten =
Hi,
Not sure what I changed, but when I open x-lite now, I get the following
verbose output on the CLI:
[Mar 3 15:31:22] NOTICE[2273]: chan_sip.c:21331 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: user2
Does this indicate a successful registration of user2?
Is
W/O mailbox should just be a warning. Sip show peers will tell you if the
registration was actually successful.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Culhane
Sent: Wednesday, March 03, 2010 9:36
Here is my output of 'sip show peers'
user1/user110.41.3.12 D N 10434Unmonitored
user2/user210.41.3.12 D N 65293Unmonitored
user3/user3(Unspecified)D N 5060 Unmonitored
user4/user4
Hello guys,
Finally I have done the upgrade.
Theres no more deadlock now ! Thanks.
Something still goes wrong and I dont find anything on that :
Most of users connected on Asterisk/MGCP cannot place calls because a hang
up ringback tone triggered while typing the phone number on
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tim Culhane wrote:
Here is my output of 'sip show peers'
user1/user110.41.3.12 D N 10434Unmonitored
user2/user210.41.3.12 D N 65293Unmonitored
user3/user3
Hello all,
Here my architecture :
Proxy1-asterisk1-proxy2-phone1
If a call arrived from proxy1 to phone1 AND phone1 always forward to proxy,
asterisk1 say:
-- Now forwarding SIP/phone1-001d to 'Local/969990...@proxy2' (thanks to
SIP/proxy2-001e)
Why it use Local ?
I just
Ok thanks everybody.
I seem to have verbose output to the CLI working with the addition of
'verbose' to the console line in logger.conf.
Also, I now seem to have a couple of users registered via x-lite ... Though
I don't really know why x-lite - asterisk connection suddenly decided to
work.
On Wed, 2010-03-03 at 10:14 +0530, DHAVAL INDRODIYA wrote:
Hi,
Carlos
I checked dmesg on my server and i found following message
what is meaning for this ? i cant understand
VPM400: Not Present
VPM450: echo cancellation for 128 channels
VPM450: hardware DTMF disabled.
VPM450:
Hi,
I am trying to implement 911 funtionality in my PBX. A call should
drop if all lines are busy. Here is my context nineoneone from
extensions.conf
[nineoneone]
exten = s,1,Set(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,Set(EMERGENCY=1,g)
exten =
Hi All,
I'm about to setup an Asterisk install to take over an old legacy PBX
system. At present, the legacy system has modules in it which provides 4
* data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on
site, these use the ISDN30 (uk) that the normal voice calls go over.
Is
Greetings:
I'm in the situation where I'm trying to splash information picked off
by an asterisk IVR into a Cisco call center environment. I'm under the
impression that the ONLY way to do this is to setup socket connections
with the Cisco voice processor, or CVP, and send packets
corresponding to
On 3 Mar 2010, at 17:21, mir shahnawaz wrote:
[nineoneone]
exten = s,1,Set(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,Set(EMERGENCY=1,g)
exten = s,n,Set(SET_EMERG_FLAG=1)
exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten =
- Mark Adams m...@campbell-lange.net escreveu:
Hi All,
I'm about to setup an Asterisk install to take over an old legacy PBX
system. At present, the legacy system has modules in it which provides
4
* data ISDN links to the video conferencing unit (Tandberg 3000 MXP)
on
site, these
Warren Selby wrote:
You need to set your firewall public ip to dhcp in order for Uverse
dmz to work.
Thanks,
--Warren Selby
On Mar 2, 2010, at 8:53 PM, sean darcy seandar...@gmail.com wrote:
Fred Posner wrote:
On Mar 2, 2010, at 6:27 PM, sean darcy wrote:
I've just got Uverse
On Wed, Mar 3, 2010 at 12:03 PM, sean darcy seandar...@gmail.com wrote:
Well at least my RG doesn't let you use DMZplus _unless_ you've chosen
dhcp. So I did. And the RG shows the router as DMZplus. And I can ssh
into my router from the internet.
Anybody else got this working?
sean
I
On Mar 3, 2010, at 1:03 PM, sean darcy wrote:
Well at least my RG doesn't let you use DMZplus _unless_ you've chosen
dhcp. So I did. And the RG shows the router as DMZplus. And I can ssh
into my router from the internet.
Anybody else got this working?
sean
What are the issues?
Thanks for your reply. This all I have, am I missing something? Please
help in this regard. Here is full output from CLI
-- Executing [...@default:1] Goto(SIP/501-0137,
nineoneone,s,1) in new stack
-- Goto (nineoneone,s,1)
-- Executing [...@nineoneone:1] Set(SIP/501-0137,
I work with various fixes in the Asterisk source tree...
cross-compilation to new platforms, adding new features (channels,
resources, etc), and adding new configuration samples that do useful things:
https://issues.asterisk.org/view.php?id=16090
https://issues.asterisk.org/view.php?id=15858
Hi All,
Asterisk 1.4.25.1 .22 .29 - pretty much every Asterisk install we have out
there exhibits this.
Just wondering how to free a channel that will stay eternally busy ala:
carl*CLI core show channels
Channel Location State Application(Data)
Hi, thanks for your response.
I'm not sure if I explained correctly. I need asterisk to provide an
ISDN data function, whilst also routing voice calls over the same PRI.
Is this possible?
Regards,
Mark
On 3 Mar 2010, at 17:58, Vinícius Fontes vinic...@canall.com.br
wrote:
- Mark
Is there any easy way to identify which script or service is
connecting to the Asterisk manager? Somewhere on my system a script or
service is trying to connect with a bad user name or password. I get
the following error: connect attempt from '127.0.0.1' unable to
authenticate
I thought maybe I
On Wed, Mar 3, 2010 at 8:30 AM, BERGANZ Francois
franc...@acropolistelecom.net wrote:
Hello all,
Here my architecture :
Proxy1—asterisk1—proxy2—phone1
If a call arrived from proxy1 to phone1 AND phone1 always forward to proxy,
asterisk1 say:
-- Now forwarding SIP/phone1-001d
Using distinctive ring detection with bell202 cid, is there any way to tell
DAHDI to sometimes expect the cid after the 2nd ring, other times after the
1st?
I just added RingMaster service (2nd DID w/ distinctive ring) to a TDM800P
FXO line. No problem setting dringcontext for the 2nd DID.
Search bugs.asterisk.org and enter 'digital' in the search field.
It probably will is my answer. I currently am not using it, so YMMV.
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Adams
Hi all,
What i'm going to do is that enable caller sing while playing a
background music (likes karaoke). My approach is using Monitor and
Meetme apps.Caller make a call to asterisk, asterisk join caller in to a
voice conference and create a dummy caller which will play music, then
Monitor app
Hi all,
It maybe not clear that what i'm going to do.
What i want to do is that enable user to call to a number then a
background music will be played and he/she sing to mobilephone, the
voice will be recorded and synchronized with the music.
Any idea?
There is an approach which using Monitor
This may help you:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
On Wed, Mar 3, 2010 at 11:20 PM, Pham Quy qu...@vega.com.vn wrote:
Hi all,
It maybe not clear that what i'm going to do.
What i want to do is that enable user to call to a number then a
background
Hello
I have successfully compiled OSLEC for echo cancellation for DAHDI channel.
Is there any way to do echo cancellation for SIP Channel.
Is any, please suggest me.??
Thanks in advance..
--
Regards,
Chandrakant Solanki
--
Hello,
I'm facing problem where as whenever there are incoming call from
pstn, there will be no audio coming in. User at the other end also
could not hear my voice. This happens few days back. Im using asterisk
1.6.1.2 with dahdi tool 2.2.0.
I thought it was time to upgrade, so upgraded to dahdi
additional info on the system
Linux home 2.6.30.3-SLACKWARE #1 Sun Feb 7 09:09:33 MYT 2010 i686
Intel(R) Pentium(R) 4 CPU 2.00GHz GenuineIntel GNU/Linux
Asterisk 1.6.2.5, Copyright (C) 1999 - 2009 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes with
On Wednesday 03 March 2010 22:20:40 Pham Quy wrote:
It maybe not clear that what i'm going to do.
What i want to do is that enable user to call to a number then a
background music will be played and he/she sing to mobilephone, the
voice will be recorded and synchronized with the music.
Any
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