hi,all
one thing confused me these days. i don't know which method to choose,
and don't know which one is better perfoermance than another when in
production system.
i can save dialplan in the extension table , i also can write dialplan
in extension.conf with MYSQL commmand to fetch data from
Hello!
I tried using sendtext() in the Asterisk dialplan to send a SIP MESSAGE to Bria
or a recent Eyebeam on my mac. I know it used to work, but right now I get 100
trying and nothing else from the softphone.
Anyone that knows what's going on here?
Thanks,
/O
--
How does Asterisk select which of its IP addresses to use to send as the
address to use for RTP connections? I want to be able to use a specific one.
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
I'm getting dozens of these at a very high rate:
[Jan 20 09:15:27] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but
based on stale nonce received from ' sip:1...@gnat.com;tag=as5f1a9480'
[Jan 20 09:15:28] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but
based on stale
Perhaps this will help
http://lists.digium.com/pipermail/asterisk-users/2007-August/194341.html
--
Danny Nicholas
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner
Sent: Wednesday, January 20,
Does anyone know if you can use the Polycom Norstar Clarity
speakerphones with Asterisk?
Model number is 2501-03308-001 'C'
Is it a SIP handset or analog style unit (or worse proprietary).
Cheers,
Dean
--
http://lists.digium.com/pipermail/asterisk-users/2005-July/110220.html
phone is using old authentication challenge, you may have restarted
asterisk, or did a sip reload, if the message is driving you batty, reboot
the phone.
-Original Message-
From:
On 20 Jan 2010, at 14:39, Dean Collins wrote:
Is it a SIP handset or analog style unit (or worse proprietary).
I'd say analogue.
W--
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asterisk-users
According to what I see on Ebay, it is an Analogue handset. You would have
to hook it to an FXO port.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Will Payne
Sent: Wednesday, January 20, 2010 8:57 AM
To: Asterisk Users
Cool, I have a spare daughterboard port for a S110M so will hook it up
into that.
Anyone got a S110M they want to sell cheap?
Cheers,
Dean
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
On 1/17/2010 3:25 PM, shawn bright wrote:
Hey all,
i love your name, btw.
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
The AstLinux Team would like to announce that the 0.7.0 version of
AstLinux is available for download. There have been many significant
updates in this release including updating to the latest Asterisk
Release (1.4.29), moving to DAHDI (2.2.0.2) along with several other
system updates.
For a
FXS port is correct answer.
FXO ports are for PSTN or PBX lines
FXS SUPPLIES battery and ringing, and receives DTMF ( or pulse ) dialing
FXO receives battery and supplies DTMF ( or pulse ) dialing
Danny Nicholas wrote:
According to what I see on Ebay, it is an Analogue handset. You would
I never get that one right :(
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent: Wednesday, January 20, 2010 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I continued trying.
Now I reached 2 results.
1.
Asterisk ver1.6 or more has bug .
When you want to use jitter and PLC and want to see packet-log , you will
set ' jblog=yes ' on 'sip.conf '.
But Asterisk can't make log-file.
In /tmp/ packet-log-file will be made, if jb-modules work
On Wed, Jan 20, 2010 at 4:40 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
The AstLinux Team would like to announce that the 0.7.0 version of
AstLinux is available for download. There have been many significant
updates in this release including updating to the latest Asterisk
Release
You have a powerful name, yourself.
sk
On Wed, Jan 20, 2010 at 9:38 AM, Sean Bright sean.bri...@gmail.com wrote:
On 1/17/2010 3:25 PM, shawn bright wrote:
Hey all,
i love your name, btw.
--
_
-- Bandwidth and Colocation
Hi All;
I have a Plocyom 320 model, it supports 2 extensions (line 1 and line 2), when
configuring line 1, then I have to determine the username and password and IP
address of the server to register on it. And same thing when configuring the
line 2.
How can I receive (and make call) using the
On 01/17/2010 09:25 PM, shawn bright wrote:
Hey all,
We have been using a TDM400 card at work to provide our IVR.
We we have upgraded our server and now require the same capability,
but on a card that goes into a PCI Express.
Any suggestions would be greatly appreciated.
oh, and it has to
I use the 331, and only have 1 line assigned, and each phone has a call
limit of 10, if another call comes in, they can answer it, and it would put
the other caller on hold, you can then switch between callers by using the
up/down keys.
-Original Message-
From:
On Wed, Jan 20, 2010 at 11:00 AM, randall rand...@songshu.org wrote:
On 01/17/2010 09:25 PM, shawn bright wrote:
Hey all,
We have been using a TDM400 card at work to provide our IVR.
We we have upgraded our server and now require the same capability,
but on a card that goes into a PCI
What is the configuration of the TDM400?
Sangoma makes a nice card as well., I think the A200 is available in PCIe
and supports from 2-4 and I think the A400 does 2-24
If you just answer 4 lines.. you could always just use a SIP Gateway, and
not use any PCIe card.
If you have a pbx, maybe a
On 01/20/2010 06:00 PM, randall wrote:
On 01/17/2010 09:25 PM, shawn bright wrote:
Hey all,
We have been using a TDM400 card at work to provide our IVR.
We we have upgraded our server and now require the same capability,
but on a card that goes into a PCI Express.
Any suggestions would
If you do a core show channels during an active call, you'll see that 800
is actually 800-x during a call. FWIW, you probably want a call-limit
of something like 3-5 to cut down on phantom calls left by park, transfer,
etc.
-Original Message-
From:
Hi,
I am running a Asterisk 1.6 box in our Data Centre, and have a number of users
connecting to that box, as their PBX.
Calls in and out work fine, as does voicemail.
The PBX at the Data Centre has an External IP, Nat’d to it by the firewall, and
the relevant ports are open.
The Office
Hello;
Thanks alot for your help and advise.
This is good for receiving, what about making calls? If already I have a call
at line 1 and need to place another call, how to do this? Do I have to
configure a new extension for line 2 (so two extensions for the same phone), or
I can do it with
Admittedly I didn't read your SIP debug (on the mobile), but do you have
reinvite=no set for the extensions and SIP trunks (providers)?
This sounds on the surface like a classic case of the Mondays. Erm reinvites I
mean.
snip
1. Incoming call from pstn/viop provider
2. Call is answered by a
On 01/17/2010 02:25 PM, shawn bright wrote:
We have been using a TDM400 card at work to provide our IVR.
We we have upgraded our server and now require the same capability, but
on a card that goes into a PCI Express.
Any suggestions would be greatly appreciated.
oh, and it has to work with
I had the exact same issue and it was caused by a crappy firewall at the phone
site. Once they swapped it out with a box that did NAT correctly, the issue
went away. I don't think you said if the phone site is being NAT'd or
firewalled and when you mentioned the debugs below, you said
I am using H.323 to create a trunk between Asterisk and Avaya IP Office system.
Everything is working correctly, Asterisk
can call Avaya and vise versa. Now I create a conference room with a
user pin in Asterisk. Avaya can call into the conference room, but can
enter the pin number. The error
At 5:59 PM on 19 Jan 2010, __ wrote:
Test case:
We have e1 trunk and multi-channel sip line. Clients waiting in the
queue, which can handle 30 clients. They listen mellody and their
position, while waiting. The system can handle only 5 clients at the
moment. As soon
This is the setting i am using for Avaya CM to Aseterisk. (and pinf code
is working when dialing from Avaya to Asterisk
conference)sip:/etc/asterisk# cat ooh323.conf
[general]
bindaddr=213.88.138.183
port=5088
context=inputinterior.se
dtmfmode=rfc2833
;h323id=may day
;callerid=may day
Hi All,
I'm using Asterisk 1.4 branch and checking the status of some SIP
Peers with the functions ${SIPPEER(101:status)} and the result is OK
(48 ms). Seems to work fine.
Now I would like to use the function CUT to set a variable with the
'OK' portion of the status OK (48 ms) and then do some
On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson jmr.richard...@gmail.com wrote:
Hi All,
I'm using Asterisk 1.4 branch and checking the status of some SIP
Peers with the functions ${SIPPEER(101:status)} and the result is OK
(48 ms). Seems to work fine.
Now I would like to use the function
The Asterisk Development Team is pleased to announce the release of
DAHDI-Linux and DAHDI-Tools version 2.2.1.
DAHDI-Linux 2.2.1, DAHDI-Tools 2.2.1, and DAHDI-Linux-Complete
are available for immediate download at
http://downloads.asterisk.org/pub/telephony/dahdi-linux
This email is not a question, but a potential solution to any who have
had the same issue I have faced.
If you have agents logged in to multiple queues at the same time,
Asterisk does not handle the answering of those queues in any set order
or sequence. It has no way of prioritizing calls
Hi all!
I've been reading this list for a few weeks and now this is my first
post. :-)
I'm planning to build a new VoIP telephone system at our company. It's
just a small company with not more than 3-4 employees.
The telephone system is not so important for us because each employee
has
On 23:28, Wed 20 Jan 10, Felix Tiefenthaler wrote:
Hi all!
I've been reading this list for a few weeks and now this is my first
post. :-)
I'm planning to build a new VoIP telephone system at our company. It's
just a small company with not more than 3-4 employees.
The telephone system
I'll second that notion - next up, why bother with POTS/PSTN when Asterisk
offers chan_mobile that would allow a dedicated cell-phone to be your
line?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michiel van
Hello,
We are recording our calls to queues by putting the appropriate options in
our queue.conf. This is all working properly.
We would now like to set the MixMonitor option to adjust the caller volume
(which is very quiet). With the regular MixMonitor application, we would
just add the v4
Hi,
I installed asterisk-1.6.2.0.tar.gz and linphone 3.2.1 for the clients on both
linux and windows vista. I have a problem on the windows linphone client, such
as failing registration. Linphone on Linux works well, so the communication
between linphone on linux works well. I wonder someone
Wednesday, January 20, 2010, 11:41:48 PM, Michiel wrote:
Forget about virtualization!
...
Virtualisation is nice for test-setups, but thats it. for any real job
it's a major pain in the ass and makes stuff bork beyond imagination.
Well. Why do you use computer? There're slide-rule. You can
On Thu, 21 Jan 2010, Gergo Csibra wrote:
Wednesday, January 20, 2010, 11:41:48 PM, Michiel wrote:
Forget about virtualization!
...
Virtualisation is nice for test-setups, but thats it. for any real job
it's a major pain in the ass and makes stuff bork beyond imagination.
Well. Why do
Jeff LaCoursiere wrote:
On Thu, 21 Jan 2010, Gergo Csibra wrote:
Wednesday, January 20, 2010, 11:41:48 PM, Michiel wrote:
Forget about virtualization!
...
Virtualisation is nice for test-setups, but thats it. for any real job
it's a major pain in the ass and makes
The playback command is designed to work with multiple formats
If the channel in question is gsm it'll use a .gsm file before a .wav file
if the .wav file is in the directory, is it playable by asterisk?
(8000hz sample rate, etc etc)
On Tue, Jan 19, 2010 at 8:20 AM, Danny Nicholas
On Wed, Jan 20, 2010 at 10:18 PM, C. Chad Wallace
cwall...@lodgingcompany.com wrote:
At 5:59 PM on 19 Jan 2010, __ wrote:
Test case:
We have e1 trunk and multi-channel sip line. Clients waiting in the
queue, which can handle 30 clients. They listen mellody and their
Beside the port number and the alaw, the only difference is the dtmf. I added
this into my ooh323.conf and it still didn't work.
dtmfcodec=127
dtmfmode=rfc2833
I also tried: dtmfmode=h245signal
This is to an Avaya IP Office 500.
--
At 3:09 AM on 21 Jan 2010, __ wrote:
On Wed, Jan 20, 2010 at 10:18 PM, C. Chad Wallace
cwall...@lodgingcompany.com wrote:
At 5:59 PM on 19 Jan 2010, __ wrote:
Test case:
We have e1 trunk and multi-channel sip line. Clients waiting in
Thursday, January 21, 2010, 12:53:09 AM, Jeff wrote:
On Thu, 21 Jan 2010, Gergo Csibra wrote:
Wednesday, January 20, 2010, 11:41:48 PM, Michiel wrote:
Forget about virtualization!
...
Virtualisation is nice for test-setups, but thats it. for any real job
it's a major pain in the ass and
My development system for asterisk is a virtual CentOS 5.4 world running under
Fusion on my MacBook. I am usually only doing a few calls at a time. I have an
IAX trunk to our office Asterisk PBX so I can access the PRI line there. I do
meetme rooms and recording of calls and all seems to work
yes. asterisk can playback wav file . but need transfer to 8000hz.
Using WAV files
Asterisk has codecs for wav (pcm), gsm, g729, g726, and wav49, all of
which can be used for Playback and Background. However, Asterisk does
not understand ADPCM WAV files. To convert your WAV files to a format
On Wed, Jan 20, 2010 at 4:28 PM, Felix Tiefenthaler
tiefenthale...@gmail.com wrote:
Now my big question: What kind of virtualization should I run on the
Server? I have already used VMware ESXi and Proxmox.
It would be very nice if there was a way to make snapshots (for
backup purposes).
On Wed, Jan 20, 2010 at 4:41 PM, Michiel van Baak mich...@vanbaak.infowrote:
Virtualisation is nice for test-setups, but thats it. for any real job
it's a major pain in the ass and makes stuff bork beyond imagination.
You're right, I doubt that whole Amazon cloud thing will ever catch
On Wed, Jan 20, 2010 at 5:53 PM, Jeff LaCoursiere j...@jeff.net wrote:
Pretty crappy analogy. Just because you *can* do something doesn't mean
it is production ready. But then the OP said it wasn't all that
important, so I would say go Xen and tell us how it works out. I think
you will
Hello,
- ---
|Sip Softphone|---|Internet||F.W|-|Asterisk|
- ---
IP addresses: a.b.c.dq.w.e.r
The SIP softphone(x-lite) is configured to
Hi,
I am currently using asterisk to record all incoming calls. My setup is as
follows, the asterisk server has a two TE120P cards one of which
sends/receives calls from the carrier and the other is connected to a
Siemens HiPath 3000. All calls that come into asterisk use MixMonitor to
record
On Wednesday 20 January 2010 14:57:38 JR Richardson wrote:
On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson jmr.richard...@gmail.com
wrote:
I'm using Asterisk 1.4 branch and checking the status of some SIP
Peers with the functions ${SIPPEER(101:status)} and the result is OK
(48 ms). Seems
I am currently using asterisk to record all incoming calls. My setup is as
follows, the asterisk server has a two TE120P cards one of which
sends/receives calls from the carrier and the other is connected to a
Siemens HiPath 3000. All calls that come into asterisk use MixMonitor to
record
Make sure u have the correct DTMF over IP (or what it is named in IP
Office, thats the CM name) setting on the signal-group. In my case: DTMF
over IP: rtp-payload
On Wed, 20 Jan 2010 16:11:58 -0800 (PST), hin lee wrote: Beside the
port number and the alaw, the only difference is the dtmf. I
On 01/20/2010 11:28 PM, Felix Tiefenthaler wrote:
Hi all!
I've been reading this list for a few weeks and now this is my first
post. :-)
I'm planning to build a new VoIP telephone system at our company. It's
just a small company with not more than 3-4 employees.
The telephone system is not
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