On Saturday 23 June 2012, neo haux wrote:
Actually I can start and receive SIP calls (PC client, iphone client)
but I have an issue with calling external number throught PSTN
(certified-asterisk-1.8.11-cert2).
I notice the number you Dial()led didn't start with a zero.
Check with your telco
Le 21/06/2012 09:52, Ishfaq Malik a écrit :
On Wed, 2012-06-20 at 20:04 +0200, Stefan at WPF wrote:
Hello,
1) I am wondering what is the best practice to monitor if there are or
were problems with SIP calls on my Asterisk box.
[...]
I've not used this myself but had a look at the site and I
Hi all,
I am trying to control the whole call using a FastAGI script. To that
effect I launch a FastAGI script (written with asterisk-java).
Basically, I want to DIAL from within the FastAGI script. When the call
ends I want to control the hangup (if executed at the remote end), and
depending on
On 06/24/2012 07:53 PM, Mitchell Johnson wrote:
I'm testing a few IAX trunk scenarios in a controlled lab. From server2
extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across
the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the
6001 phone,
Hi All,
I have a simple urgent question that I couldn't find the answer yet, can we
customize the voicemail attachment format *per user* in asterisk *1.2 *(like
all receive wav attch but one or two users receive attch in gsm format)? if
yes can you show me how please?
--
Khalid Touati
--
On Fri, Jun 22, 2012 at 08:07:54PM +1200, Alec Davis wrote:
Have a look at the latest blacklist sample in dahdi trunk
http://svnview.digium.com/svn/dahdi/tools/trunk/blacklist.sample?view=log
file: blacklist.sample
...
# Some mISDN drivers may try to attach to cards supported by DAHDI. If
2012/6/25, Tzafrir Cohen tzafrir.co...@xorcom.com:
On Fri, Jun 22, 2012 at 08:07:54PM +1200, Alec Davis wrote:
Have a look at the latest blacklist sample in dahdi trunk
http://svnview.digium.com/svn/dahdi/tools/trunk/blacklist.sample?view=log
file: blacklist.sample
...
# Some mISDN drivers
On 06/22/2012 05:12 PM, bilal ghayyad wrote:
One of the problems I faced with Polycom is the voice volume and ring volume,
it is low.
When it rings, even if it is maximum volume, still it is weak.
When I talk and I set the volume to the maximum, I still feel the voice volume
is low and would
On Mon, Jun 25, 2012 at 9:23 AM, khalid touati khalidtou...@gmail.comwrote:
Hi All,
I have a simple urgent question that I couldn't find the answer yet, can
we customize the voicemail attachment format *per user* in asterisk *1.2
*(like
all receive wav attch but one or two users receive
On Mon, 2012-06-25 at 10:45 -0500, Kevin P. Fleming wrote:
On 06/22/2012 05:12 PM, bilal ghayyad wrote:
One of the problems I faced with Polycom is the voice volume and ring
volume, it is low.
When it rings, even if it is maximum volume, still it is weak.
When I talk and I set the
On 25 Jun 2012, at 16:58, Jeff LaCoursiere wrote:
Actually we get that complaint a lot too (Polycom ring volume). We
typically install in hotel environments, and in their back office the
environment can be noisy, as well as in their restaurants.
I imagine in a typical office environment
We have the ringer volume issue with some customer environments as well. We
use Grandstream phones in a lot of installs so we just upload a custom
ringtone with the db pushed up on it a bit.
We are testing the Digium phones and have concerns if we will be able to
use them for the high noise
- Original Message -
We have the ringer volume issue with some customer environments as
well. We use Grandstream phones in a lot of installs so we just
upload a custom ringtone with the db pushed up on it a bit.
We are testing the Digium phones and have concerns if we will be able
to
I am looking for a CDR report tool that will link extensions to the
user's names... are there any that offer this feature? We are using
trixbox 2.8.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Hello,
Does SendFAX have the ability to put the caller ID and timestamp on the fax?
If so, is there a way to adjust the timezone used for the timestamp?
Thanks for any assistance.
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Alejandro,
Try the 'g' option to Dial():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
- *g*: When the called party hangs up, continue to execute commands in
the current context at the next priority
On 25 June 2012 20:17, Alejandro Recarey alexreca...@gmail.com wrote:
Hi all,
In what way was my question not meaningful? Not enough details?
Here's our current receive fax route:
sender fax machine - telco - E1 line - sangoma card - asterisk
We're currently using free fax for asterisk.
I have read that fax over voip is not reliable, but is it the same
case for faxes
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