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
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 a
>
> 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
> reco
On Wednesday 20 January 2010 14:57:38 JR Richardson wrote:
> On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson
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 to work fine.
> >
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 call
Hello,
- ---
|Sip Softphone|---|Internet||F.W|-|Asterisk|
- ---
IP addresses: a.b.c.dq.w.e.r
The SIP softphone(x-lite) is configured to regis
On Wed, Jan 20, 2010 at 5:53 PM, Jeff LaCoursiere 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 only have t
On Wed, Jan 20, 2010 at 4:41 PM, Michiel van Baak wrote:
>
> 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.
;)
Virtualiza
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" purpose
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
which
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 we
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 th
At 3:09 AM on 21 Jan 2010, __ wrote:
> On Wed, Jan 20, 2010 at 10:18 PM, C. Chad Wallace
> 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
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.
--
On Wed, Jan 20, 2010 at 10:18 PM, C. Chad Wallace
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
>> position, while
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 wrote:
> Just
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 pai
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.
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
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 c
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"
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 va
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
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 it's
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 in
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
http://downloads.asterisk.o
On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson 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 CUT to set a varia
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 so
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
disallo
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
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
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 "datacent
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
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.
1. Incoming call from pstn/viop provider
2. Call is answered by a user
3.
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 sam
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 user
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: asterisk-users-boun...@lists.dig
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 sugg
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 PRI
On Wed, Jan 20, 2010 at 11:00 AM, 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
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: asterisk-users-boun...@lis
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
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
You have a powerful name, yourself.
sk
On Wed, Jan 20, 2010 at 9:38 AM, Sean Bright wrote:
> On 1/17/2010 3:25 PM, shawn bright wrote:
> > Hey all,
>
> i love your name, btw.
>
> --
> _
> -- Bandwidth and Colocation Provided by
On Wed, Jan 20, 2010 at 4:40 PM, Darrick Hartman
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 (1.4.29), moving to DAH
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 correc
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: [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
>
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
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:
http:
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
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 Ma
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
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: asterisk-users-boun...@lists.digiu
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
--
__
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, 2
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 '"" ;tag=as5f1a9480'
[Jan 20 09:15:28] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but
based on stale nonce received f
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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as
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
--
___
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 dat
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