Thanks for your replies.
1.. Our connection mainly for voip, occasionally used for surfing websites.
2.. We are using codec g711u for local calls through TE120P, and g729 only if
making international calls through our sip provider, which only allow g723 and
g729. How can we get the license
Peter Pauly wrote:
I currently have the following setup:
exten = 2000,1,Playback(/var/lib/asterisk/sounds/Greeting)
exten = 2000,2,Queue(Qabcdef|t)
exten = 2000,3,Playback(/var/lib/asterisk/sounds/EveryonesBusy)
exten = 2000,4,Hangup
exten = 2000,103,Hangup
What happens is, that the
On 12/7/07, Zaheer K. Master [EMAIL PROTECTED] wrote:
Hi all,
I'm not getting any sidetone on my Snom 370. I searched the web and the snom
wiki, but I don't see any place to enable/adjust it. Callers say I sound
great on the other end, but I don't hear myself so it is a little
off-putting.
Hi! Can anybody help with integration of asterisk with cisco call manager
express? I tried to connect them via H323 and Sip, but wasn't successful in it.
So can you share any advices or instructions?
___
--Bandwidth and Colocation Provided by
Hello to all.
would like to know if somebody already had problems of transmission of
FAX through interconnection two asterisk with protocol IAX2 and codec
Ulaw with siwtches Cisco Systems 2950.
I explain: If I place this family of switches(Catalyst 2950) passes
through voice, normally, but the FAX
Steve Davies wrote:
On 12/7/07, Zaheer K. Master [EMAIL PROTECTED] wrote:
Hi all,
I'm not getting any sidetone on my Snom 370. I searched the web and the snom
wiki, but I don't see any place to enable/adjust it. Callers say I sound
great on the other end, but I don't hear myself so it is
Anciso, Roy wrote:
Is there a way to tell asterisk to beep every few seconds rather than
play MOH.
I suppose you mean the beeps you hear when dialing any phone number -
the ringing indication? I'm not sure that this would work with queues,
however it works great with Dial(). I'm also not
Hi guys,
First of all, I know that this server must be upgraded asap, I'm just
wondering if anyone of you has already faced this problem and , if so, would
the upgrade solve my problems...
CAPI version 0.6
Asterisk 1.2.5
AGI scripts are being used
Main problems:
-Dropped Calls
- ps aux | grep
Benny Amorsen wrote:
The multiple registrations issue is currently our largest problem with
Asterisk. The workarounds are horrible and complicated. Say we want 3
phones ringing when a new call comes in, but if just one of them is
busy outbound or inbound, they should all return busy for
Hello.
I am going through the documentation and trying to find if asterisk can help me
in my case. It is quite difficult to find answer because I do not know the
exact question.
I have two location. Each in different country. Both locations have Siemens
HiPath - different type and software. I
My Gurus!
I'm still playing with asterisk in the lab here. There
is a feature that I need in a production asterisk
system. I was wondering if it already exists in
asterisk.
When we want to shutdown a production asterisk system,
we would like the shutdown to happen after there are
no
more calls
Hello all,
i have a problem on incoming call's from SIP Provider that ist going through
the Asterisk to a Grandstream HT502. The first ring is executed on the HT502
propperly, but no more ring will follow. But the call can nevertheless be
answered by a phone on the gateway.
If i call the same
On 12/10/07, Jeng Yu [EMAIL PROTECTED] wrote:
My Gurus!
I'm still playing with asterisk in the lab here. There
is a feature that I need in a production asterisk
system. I was wondering if it already exists in
asterisk.
When we want to shutdown a production asterisk system,
we would like
Hi Jeng,
From the Asterisk CLI type stop gracefully and it will do exactly what you
described (stop accepting calls and shut down when all calls have
completed). If you don't want to stop accepting calls, but still want to
stop Asterisk when there are no active calls, you can use stop when
Jeng Yu wrote:
This would be the ultimate graceful shutdown; perfect
for routine system maintenance tasks on production
servers handling continuous traffic.
if [ `asterisk -rx show channels verbose|awk '/active calls/{print
$1}'` -eq 0 ]
then asterisk -rx stop now
fi
--
asterisk linkedin group
I have created an asterisk linkedin group for anyone interested.
http://www.linkedin.com/e/gis/45252/66270A773F53
Thank You,
Steven BerkHolz
- MCSA - MCSE -
Manager of Information Systems
HIROTEC AMERICA
Board member of
Connectech Greater Detroit
www.connectech.org
Alex Robar wrote:
Hi Jeng,
From the Asterisk CLI type stop gracefully and it will do exactly what
you described (stop accepting calls and shut down when all calls have
completed). If you don't want to stop accepting calls, but still want to
stop Asterisk when there are no active calls, you
Mark J Elkins wrote:
The 7.x.x firmware on the snoms IMHO is still alpha, or just barely
beta quality, and sadly for the snom370 there is no other firmware
available. Snom (usually very good) have just dropped the ball in a
BIG way with this new branch of firmware.
Software version
I haven't found outcall that confusing though I do agree that a TAPI
Driver that makes use of the available outlook call functions will make
for the easiest, most streamlined user experience.
I also agree that these convenience and little feature are very
important especially with Microsoft
Yes Asterisk feets your needs 100%.
You need 2 Asterisk each one with Telephony cards (ISDN or Analog) and I
would recommend you to setup an IAX trunk to interconnect both servers.
Each Asterisk will act as a Media Gateway.
Look for IAX trunk over the wiki it's all there.
good luck.
Best
Atis Lezdins wrote:
As I've heard before -
Asterisk is a hacker's tool that would allow nearly unlimited
possibilities if you know how to configure it.
And in addition to that Asterisk is the dratted phone
system (http://bugs.digium.com/view.php?id=10740)
LOL
Regards,
Philipp Kempgen
--
And whats going to happen at this secret LinkedIn group clubhouse? :-)
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Kovář Jan wrote:
I have two location. Each in different country. Both locations have Siemens
HiPath - different type and software. I can not use card that would allow me
to connect those PBXs using SIP. But I have some free ISDN and analog ports
in both PBXs. Is it possible to use Asterisk
Does anyone know how to customize the order of the soft keys on a 7960
running SIP? All the documentation I could find is CallManager
related. Specifically, I want to move the transfer function to the
first set of buttons during a call.
___
--Bandwidth
Thanks, All! And thanks, Oquendo! I will experiment
with this suggestion. I was actually thinking in terms
of a situation where it would be done
non-interactively.
Jeng
--- J. Oquendo [EMAIL PROTECTED] wrote:
Jeng Yu wrote:
This would be the ultimate graceful shutdown;
perfect
for
Try this:
queue-thankyou = /dev/null
On Nov 30, 2007 10:02 AM, [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
Short of replacing a sound file with a sound file containing only a
short period of silence, is there any way to suppress certain sounds
from playing during queue
Atis Lezdins wrote:
Anciso, Roy wrote:
Is there a way to tell asterisk to beep every few seconds rather than
play MOH.
I suppose you mean the beeps you hear when dialing any phone number -
the ringing indication? I'm not sure that this would work with queues,
however it works
Hi all,
I'm new in the list, and I have a problem upgrading from asterisk 1.2 to
asterisk 1.4:
There is a diference from asterisk1.2 to asterisk1.4 in AMI events.
When I do a call to a queue (with the same extensions.conf dial plan)
with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid
Although these are web based might be some interesting api's to utlise.
http://www.dancewithshadows.com/tech/text-to-speech.asp
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
Is there a way to dynamically alter the sip.conf properties of a SIP peer
in runtime without doing a SIP reload?
I am specifically thinking of enabling reinvites for users dynamically
based on whether they are registered from a public address.
--
Alex Balashov
Evariste Systems
Web:
Hello,
I'm trying to decide between the foneBRIDGE2 ($1135) and foneBRIDGE2-EC
($1610).
Has anyone here directly compared the two? Would we really suffer
without the onboard echo cancellation? The manufacturer's site doesn't
really give much helpful information about choosing one over the
Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a
64 bit 4200+ box
would there be any noticable lag or delay to bring each one of them into
a PAGE mode. so one speaker can talk out on all 230 SIP clients for a
message.
Would this work?
Thanks,
Jerry
Yes, in the queue config add:
leavewhenempty = yes
The users will enter the queue, but exit quickly and continue with the
dialplan.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Pauly
Sent: Sunday, December 09, 2007 12:33
To:
I don't think you can do much with them. This is a good guide on the
options you do have:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/i
pp7960/addprot/sip/admin/8_0/sipaxd75.htm
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
For those using Cisco 7911g phones, I am running into an issue with one
the Cisco demo phones we have. The 7961 works great with asterisk no
problems However, the 7911g gets audio clipping when recording
voicemails or the unavailable message. Also when a call is transferred
using the 7911g the
Jerry Geis wrote:
Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a
64 bit 4200+ box
would there be any noticable lag or delay to bring each one of them into
a PAGE mode. so one speaker can talk out on all 230 SIP clients for a
message.
Would this work?
Thanks,
All,
I'm looking for a creative way to do this...
I've got a Polycom 650 as well as a standard cordless phone connected to
the Zap channel...
My Dialplan in extensions.conf looks like this:
exten = 9007,1,Dial(zap/1r4SIP/polycom-9007,20,r)
exten = 9007,2,Answer
exten =
On Dec 10, 2007 1:17 PM, Jerry Geis [EMAIL PROTECTED] wrote:
Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a
64 bit 4200+ box
would there be any noticable lag or delay to bring each one of them into
a PAGE mode. so one speaker can talk out on all 230 SIP clients for a
You can also do
asterisk -rx stop gracefully
From any sort of script / crontab, etc.
On Dec 10, 2007 10:36 AM, Jeng Yu [EMAIL PROTECTED] wrote:
Thanks, All! And thanks, Oquendo! I will experiment
with this suggestion. I was actually thinking in terms
of a situation where it would be done
Eric C. wrote:
Hello all,
Just starting to setup asterisk v 1.4.11 and need to run three distinct phone
systems for three different companies.
So far, I have inbound lines going to the appropriate dial plan within the
extensions.conf file. I'm using
exten = _X.,1,NoOp(FROM NUMBER:
I;m not sure how your solution would work... but I thought I'd throw out some
ideas that we are having to implementing faxing here on a new install.
We are going to be bringing in a PRI and routing all the DIDs from our
existing copper lines to the PRI (including fax DIDs)... the solution we
Hey does anyone have any suggestions for this? Is this an asterisk issue or
a phone issue? Thanks!
Regards,
Zaheer
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zaheer K.
Master
Sent: Friday, December 07, 2007 11:22 AM
To: 'Asterisk Users Mailing List -
Contexts Its all about Contexts.
You can place the three groups of extensions/ dial plans in different
contexts.
The only dilemma you are going to encounter is that all three companies
will need to share the same company directory, and the same parking lot
(For ParK)
However, may orgs
How about fax machines talking directly to spa2102 and then out the pri or
am I missing something?
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Watson
Sent: Monday, December 10, 2007 4:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Matthew Mackes wrote:
Contexts Its all about Contexts.
You can place the three groups of extensions/ dial plans in different
contexts.
The only dilemma you are going to encounter is that all three
companies will need to share the same company directory, and the same
parking lot (For
If you mean faxing in audio it's hit or miss. We do it here and maybe
have an error every 6 pages or so. I wouldn't sell it to a customer as
a solution.
How about fax machines talking directly to spa2102 and then out the
pri or am I missing something?
Hello Folks.
I'm wondering if anyone has any helpful hints.
I recently upgraded to 1.4.11, and I'm having problems with pickup, both
directed, and the pickup feature.
My server is on the public internet, and all phones are behind a NAT router,
somewhere else on the public internet.
When a
I am using bandwidth.com for our sip provider and I can send faxes from a PAP2T
with no problems and it make it more complicated I can send faxes from a remote
office thought our main office and then out to or sip provider. Example.. PAP2T
--- asterisk --- IAX2 asterisk --- bandwidth.com ---
Hello,
Since last few days I have noticed some people complaining that their call
gets disconnected while they are in the middle of the conversations. Looking
in the log I found these error messages,
Dec 10 11:18:56 DEBUG[8833] channel.c: Bridge stops bridging channels
SIP/5060-b7a03560 and
Hi,
On Mon, 2007-12-10 at 13:00 -0500, Kevin DeGraaf wrote:
Hello,
I'm trying to decide between the foneBRIDGE2 ($1135) and foneBRIDGE2-EC
($1610).
Has anyone here directly compared the two?
Nope.
Would we really suffer
without the onboard echo cancellation? The manufacturer's site
How about connecting the fax machine directly to TDM card?
I was under the impression that this type of the solution is very solid ...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Adam Moffett
Sent: Monday, December 10, 2007 4:50 PM
To:
On Tue, Dec 11, 2007 at 12:44:27AM +0100, Patrick wrote:
Hi,
On Mon, 2007-12-10 at 13:00 -0500, Kevin DeGraaf wrote:
Hello,
I'm trying to decide between the foneBRIDGE2 ($1135) and foneBRIDGE2-EC
($1610).
Has anyone here directly compared the two?
Nope.
Would we really
Hi Michelle,
We added to the bounty for this feature some time ago[1], and had a
developer all lined up. He was unwilling to proceed, because Digium said
that our work would never get accepted because they were already
working on it. The IMAP support in 1.4 must have been it - doesn't work
for
On Mon, 2007-12-10 at 15:26 -0800, Jai Rangi wrote:
Is this the right place to post this error message and expect for the
solution.
I am using asterisk-1.2.12 on FC5. I will appreciate if someone can give me
some hints to get rid of this problem.
I doubt you'll get much response, unless
Thank you Jared,
I had the same feeling. But my servers are in production, doing great except
this problem. So i was hoping if someone had that same issue and if there
is/was an easy fix for this.
-Jai
On Dec 10, 2007 4:09 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2007-12-10 at 15:26
Hi
Does anyone have any recommendations of an SMS gateway which you can
just sign up for on a pay-as-you-go basis for testing, for use with
Asterisk?
Thanks
Robert McNaught
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
I guess that's my question. Is this the standard method of doing faxing?
Just point the PRI DIDs to a TDM and hang fax machines off of the ports?
If this is the generally accepted method of doing things (ie, it works very
reliably) then I'm good to go. The utilization of 3102s on the links
Hi all,
I have a TDM400 with all FXO in it. I can make outgoing out but the
call will be dropped between 20-30mins suddenly. Below is the message
shown in the log in the time the call drop.
[Dec 10 23:23:32] DEBUG[3613] dsp.c: ast_dsp_busydetect detected busy,
avgtone: 200, avgsilence 75
[Dec
Does anyone know how I can append to different user recorded voice files within
a dial plan? For example Asterisk ask caller a question and records the
answer, then ask another question record the answer to the end of the first
answer - so when it's played back, all the answers are in one
HI,
I have tried to add the context but it still doesn't work.
On Dec 9, 2007 11:36 PM, F6HQZ [EMAIL PROTECTED] wrote:
Hi,
Your extension 100 doesn't exist in the context where you have your PickUp
instruction.
You must include the context containing your phones into the context used by
Sorry for my late reply:
Here is my zapata.conf
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
define channels
context=incoming ; Incoming calls go to [incoming] in extensions.conf
signaling=fxs_ks ; Use FXS signaling for an FXO
rilawich,
in the CLI type the following:
CLI dialplan show [EMAIL PROTECTED]
then
CLI dialplan show [EMAIL PROTECTED]
-or-
CLI dialplan show [EMAIL PROTECTED]
and see if * recognizes the x100 in either of those...
daveC
Rilawich Ango wrote:
HI,
I have tried to add the context but it
I am planning to upgrade my asterisk to
Asterisk
1.4.15http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.15.tar.gz
Zaptel
bart,
one way is to write the recorded files to a known directory, then
launch an AGI script to use sox to combine/concatenate the two...
if you cat them into a known filename, just use the playback() cmd to
play it.
do you need specifics?
daveC
Bart Fisher wrote:
Does anyone know
speaking of multi-casting voice. since it isn't likely to get the ip
phones changed, could an app_multicast do the job?
has anyone thought of doing that?
daveC
Kristian Kielhofner wrote:
On Dec 10, 2007 1:17 PM, Jerry Geis [EMAIL PROTECTED] wrote:
Using asterisk 1.4 with 100M or 1000M
silvia,
I don't know how to pickup the message but if it is getting into the
dailplan as a variable, you can send it to an AGI() script as a
parameter...
AGI(my_script.php,${IM_TEXT})
if you give me an example of what you have already, perhaps I can think
on it more...
daveC
cimsi wrote:
thx. I got it.
On Dec 11, 2007 12:55 PM, dave cantera [EMAIL PROTECTED] wrote:
rilawich,
in the CLI type the following:
CLI dialplan show [EMAIL PROTECTED]
then
CLI dialplan show [EMAIL PROTECTED]
-or-
CLI dialplan show [EMAIL PROTECTED]
and see if * recognizes the x100 in
On Tuesday 11 December 2007 06:35:24 dave cantera wrote:
!DOCTYPE html PUBLIC -//W3C//DTD HTML 4.01 Transitional//EN
html
Please send to the list in text and not HTML
--
Dave Cotton
___
--Bandwidth and Colocation Provided by
tim,
sounds like a problem I had with bandwidth... too many devices
communicating on the same network connection to the internet...
have you tcpdump'd or used a bandwidth tool to see what the usage is?
nat=yes or nat=no? should be yes..
did you change the router between upgrades?
just some
On Dec 11, 2007 12:19 AM, dave cantera [EMAIL PROTECTED] wrote:
speaking of multi-casting voice. since it isn't likely to get the ip
phones changed, could an app_multicast do the job?
has anyone thought of doing that?
daveC
Dave,
That's just it - I think at least Snom and Linksys/Sipura
google app_pickup2, i just found it myself...
oh, still have the URL up... here it is..
http://www.thorsten-knabe.de/linux/asterisk/pickup.jsp
Lukassky wrote:
Hi everybody again.
A week ago I started a new Term about Pickup group over IAX or mISDN. I've
set all the config up with callgroup
On Sun, 9 Dec 2007 18:15:43 -0600, Tilghman Lesher
[EMAIL PROTECTED] wrote:
No, but you can use func_odbc with a backend SQLite driver.
OK, I'll give ODBC a shot, or call SQLite through a PHP script
instead. Thanks.
___
--Bandwidth and Colocation
Hi friends.
I want to monitor my system (Asterisk 1.4.15 with PostgreSql) in real time, I
am using CentOS 5.1 and try with the article in
http://www.voipphreak.ca/archives/382, but I got:
asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_snmp.so:
undefined symbol: init_agent
I need
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