Hello list,
with
the SNOM M9 DECT base station and handhelds, how can the range
best be expanded ?
Is there a DECT repeater that can be used ??
Is there a way to put some 'dumb' base station somewhere else on the
network to expand the range ?
Kind regards,
Jonas.
--
Thanks all,
I guess i will use syslog for as my choice.
Quyps
On Thu, Apr 8, 2010 at 10:16 PM, David Backeberg dbackeb...@gmail.comwrote:
On Wed, Apr 7, 2010 at 10:12 PM, Pham Quy qu...@vega.com.vn wrote:
Hi all,
I want to have a separate file to log what i need for my dialplan
Today, Chris Matthieu, Founder CEO of GetVocal, entered the
cloud-based communications market in February, with its launch of
Teleku.
Teleku is a new cloud-based telecom service that allows Web developers
to build and host phone applications that answer inbound calls and
initiate outbound
On 04/09/10 05:08, Steve Edwards wrote:
On Fri, 9 Apr 2010, Arkadi Shishlov wrote:
It would be essential to get your comments (in email or by leaving a
voice message) about sound quality if you could call the menu at
sip:1...@riga.beta.lv (actually, any number at riga.beta.lv)
I get:
I really like the idea. I will try to ask. I don't know if they will be able
to do that easily though. They ask a week or two for any changes to the hunt
programming.
Thanks,
Bruce
On Thu, Apr 8, 2010 at 3:29 PM, Edo edo.eku...@gmail.com wrote:
Hello.. maybe you can just have the telco do an
Hi,
I'm experiencing a few (but meaningful) cases of audio distortion (or bad
quality). I can't say yet how often this happens.
Please listen to the following sound file:
http://213.96.91.201/temp/distorted_audio_1.wav
This was recorded by Asterisk while the local SIP caller was dialing out a
Hello,
I am creating a call file with parameter Archive: yes. When it is
completed it is moved to directory outgoing_done. It works.
Now i want to execute a script when it is completed. Is there a
parameter/configuration for this?
--
Necati DEMİR
http://blog.demir.web.tr
On 04/09/10 15:34, Necati Demir wrote:
I am creating a call file with parameter Archive: yes. When it is
completed it is moved to directory outgoing_done. It works.
Now i want to execute a script when it is completed. Is there a
parameter/configuration for this?
You can write a script that
Do the call in a context and have the context run the script as a DeadAGI.
[call_and_do]
- exten = s,1,Dial.
- exten = h,1,Deadagi(.)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Necati Demir
Ok... They normally do it here within a few hours.. This will not be a
change to the hunt group just a forward immediately from one number to
another. If the number was functional you could even have done it yourself
using the forward code.
On Fri, Apr 9, 2010 at 6:04 AM, bruce bruce
- dotnetdub dotnet...@gmail.com wrote:
I would not think you'd need to worry about jitter on a normal 100mbit LAN
unless there is heavy traffic or people are running their PC's through the
phone (don't remember if the 501 has two ethernet ports...). Typically the
quality issues
Hi Vieri,
The sound I hear does not seem caused by packet loss, jitter or latency,
this problems usually produces a robotic or synthetic voice. It seems
produced by some kind of bad contact (most probable). It is strange that you
are seeing it using hard phones, I could bet on the headphones.
Do you seperate your voice and data networks?
On 9 April 2010 14:56, Tim Nelson tnel...@rockbochs.com wrote:
- dotnetdub dotnet...@gmail.com wrote:
I would not think you'd need to worry about jitter on a normal 100mbit
LAN unless there is heavy traffic or people are running
- dotnetdub dotnet...@gmail.com wrote:
Do you seperate your voice and data networks?
Un-top-posting...
Yes, I separate voice and data. Typically this is done using separate switches
where possible, other times, using VLANs with appropriate QoS. Regardless, your
phone and PC are
Are you sure writing to the right list???
Martin
- Original Message -
From: Sarfaraz Chougule
To: sarfaraz.choug...@gmail.com
Sent: Monday, April 05, 2010 4:54 PM
Subject: [asterisk-users] Please sign Petition - Stop Child Labour
Hello Friends,
Kind request to you all -
On Thu, Apr 8, 2010 at 10:33 PM, Alan Zheng machinecat1...@gmail.com wrote:
Hello All:
I saw there are app_fax and app_chanspy modules in 1.6.2.6, but there is NO
sample configure file for them.
Is anybody know how to use them, or where is the documentation for them?
If you read the code for
I am just guessing, but sometimes happened to me that the logic on dialplan
does not contain a hungup, so channels on spa3102 continues up even if users
have finished.
On CLI you should put core show channels, and see if there are channels to
sip/8028
On the [gw8028] context you send the call
Am 09.04.2010 13:10, schrieb Vieri:
Please listen to the following sound file:
I've experienced similar (well, vaguely similar) distortion on a
horstbox pro when echo cancellation is switched on for the zap
channels (ISDN).
Turning it off resulted in no distortion at all, but then i
Hi,
sounds for me like when i use an headset and the microfone handle
scratches on my beard while i talk ;)
maybe you have a network cable whitout screening. I had bad problems on
different phones which sounds like that you have cause of electric or
magnetic inteferences but when i changed
Hello everyone,
I'm fairly new to asterisk and this list. Currently I'm working on IAX
trunks to send/receive calls between 2 asterisk boxes with asterisk
1.6.1.1+asterisk gui 2.0. After some work in the gui, two boxes can
send/receive calls to/from the other just fine, the only problem I
have is
Hi all,
I have noticed something I can't solve regarding Asterisk (latest
1.6.0.x).
My server is set at the GMT+2 timezone. The clock is ok (I can get the
correct time at the terminal). But today I got a call at a time where
Asterisk should have gone 'off business hours'.
All log times
On 9 April 2010 16:46, Tim Nelson tnel...@rockbochs.com wrote:
- dotnetdub dotnet...@gmail.com wrote:
Do you seperate your voice and data networks?
Un-top-posting...
Yes, I separate voice and data. Typically this is done using separate
switches where possible, other times, using
I have res_fax setup and working for the most part. However, I'm seeing
some fax machines drop the connection on me -
Apr 9 17:33:11] NOTICE[30809]: res_fax.c:906 generic_fax_exec: Channel
'DAHDI/1-1' did not return a frame; probably hung up.
-- Channel 0/1, span 1 got hangup, cause 102
Hello my friends...
We are having some problems with the fax in our asterisk server...
We have:
Asterisk 1.4.21.2
Zaptel Version: 1.4.11
libpri version: 1.4.5
Digium Card TDM 410P
This digium card has 3 FXO ports and 1 FXS port where we have a fax machine
connected!
The problem is that we can
I am having serious problems connecting my client software to asterisk, i tried
x-lite would not connect, and i tried with twinkle too, it wouldnt, i cannot
get to call myself, i am not on a network, just trying all this out locally,
can i not get to connect without been on a network?
Hello my friends,
I want to make fax work in the following scenario:
My versions are:
Asterisk 1.4.21.2
WANPIPE Release: 3.4.7
Zaptel Version: 1.4.11
libpri version: 1.4.5
Digium Card TDM 410P
The E1 pri is connected to our Sangoma A102DE, we also have a SIP
Mediant Audiocodes 1000 where we
On Fri, Apr 9, 2010 at 5:17 PM, Danny Dias ing.diasda...@gmail.com wrote:
Hello my friends,
I want to make fax work in the following scenario:
My versions are:
Asterisk 1.4.21.2
WANPIPE Release: 3.4.7
Zaptel Version: 1.4.11
libpri version: 1.4.5
Digium Card TDM 410P
The E1 pri is
On Fri, 9 Apr 2010, ayodele abejide wrote:
I am having serious problems connecting my client software to asterisk,
i tried x-lite would not connect, and i tried with twinkle too, it
wouldnt, i cannot get to call myself, i am not on a network, just trying
all this out locally, can i not get
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