On Friday 08 November 2013, akhilesh chand wrote:
When I calling a number from web, my softphone show me Answer and
Decline bottoms, and then I have to click Answer to call the number. it
seems it is two step to calling the number. If I type the number direct to
my client softphone, it calls
A few bags for tie wraps. A couple ups's. And a good switch are usually the
best defence.
Following advice from another PBX discussion list I have seen. I prefer the
slap to the back if the head over the shockplate.
On Nov 7, 2013 10:53 PM, Eric Wieling ewiel...@nyigc.com wrote:
I don't have
I certainly agree that the first and best solution is to deal with the
hardware issues, and we've started working on that already.
I'll investigate the suggested Asterisk ideas and post here if anything
works for my purposes.
Mitch
On 11/08/2013 12:13 AM, Mikhail Lischuk wrote:
Mitch
I had a SIP problem on an 11.5.0 system that I look after. It registers
with a SIP trunk provider, and at one point the provider had an issue that
caused registration to fail.
The problem was that Asterisk did not keep retrying, and it was not until
it was restarted that registration was
From: Tony Mountifield t...@softins.co.uk
To: asterisk-users@lists.digium.com
Sent: Friday, November 8, 2013 10:39:25 AM
Subject: [asterisk-users] 11.5.0 - SIP registration not retrying after
failures
I had a SIP problem on an 11.5.0 system that I look after. It
registers
with a
Hello, I have a fully functional Asterisk Server, I want to configure this
server to be able to process call from Skype, can someone point me to a
howto? or if there are suggestions on best way to approach this problem.
Thanks,
--
Hi folks,
I've been trying to evaluate T.38 termination providers for a document
fax service we currently use (really!) US Robotics Courier modems for,
on POTs lines. I managed to make it work through one of our existing
upstreams, and reliably, but sadly they are the one upstream I was
In article 9519872.915.1383925949785.JavaMail.myoung@myoung-laptop,
Michael L. Young myo...@acsacc.com wrote:
From: Tony Mountifield t...@softins.co.uk
To: asterisk-users@lists.digium.com
Sent: Friday, November 8, 2013 10:39:25 AM
Subject: [asterisk-users] 11.5.0 - SIP registration not
Buy SIP Channel from Skype and you can configure it as sip trunk on
asterisk box.
Mitul
On Friday, November 8, 2013, motty cruz wrote:
Hello, I have a fully functional Asterisk Server, I want to configure this
server to be able to process call from Skype, can someone point me to a
howto? or
Hello Mitul, I did purchase a sip channel from skype, and configure it on
Asterisk
in Sip.conf
register = motty2342342:mypass...@sip.skype.com/motty2342342
[skype]
username=motty2342342
secret=mypass123
type=friend
qualify=yes
context=from-skype
host=sip.skype.com
in a testing skype account
We, along with a lot of other people, have a phone number that is pretty
important to us. Yesterday, our VoIP provider went down... won't call
any names VI, but it was pretty bad...
Our goal is to create a script within asterisk, that will place a call out
one SIP trunk provider (not the one
It's look like our test Did script, which was test a list of our did
number. it generate call files which call to a number and, after answer
play file for a 4 sec. After this, we send email for manager with excel
file where everely Did number noted mark. like this. sorry for my english.
08 нояб.
Hello,
Same issue happens on one of our Call Center installation (using Asterisk
1.6) - random unresponsive Asterisk with self heal after 2-3 minutes.
Because we could not find the root cause (till now - many thanks Ishfaq) we
end up by nightly restart on Asterisk.
We are using CLI commands more
13 matches
Mail list logo
