Hi All
I would Like to run a macro in a meetme conference when a user presses a
certain digit sequence, but I cannot seem to find how to do this , is it
possible?
if so how?
Thanks for you help
Robb
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I'm having real problems with my connection to BT, it is a home line, but
after a while it sets an alarm and only a restart of asterisk resets it
could some one look at the below configs and suggest any changes to make
this more reliable
Thanks for your help
Robb
asterisk version 1.6.1.10
its on the Avaya site, but basically,
you need to have the sip image and the 96xxsettings.txt and the
96xxUpgrade.txt on the route of a web server, the sertup the phone to read
its files from the web server, normally this is by pressing hold then
a-d-d-r then # set the ipaddress, callsvr and
Hi
thwe PrivacyManger app states thast you can use a context to match against
for the input , but gives no real examples or explaination, does anyone have
a an example context for this
Thanks in advance
Robb
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Hi
I wonder if anyone has any sugestions
I have had a TDM400 for a couple of years, and I have always had problems
with noise on the line, so tonight I have been doing some research and have
found that if I load the CPU dahdi_test has almost perfect results and no
noise
dahdi_test
Opened
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged isdn
channel in 1.6 and 1.4 using the following;-
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten =
I'm using ISDN30 for a bridged application
in all the old versions of asterisk the time slot number is shown in the
channels and dstchannel fields of the cdr
I understand this has chaned in 1.8,is there a way of getting the time slot
information stored somewhere at the end of the call so this
both show transfercapability DIGITAL
Regards
Robb
On 16 June 2011 23:40, Richard Mudgett rmudg...@digium.com wrote:
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged isdn
channel in 1.6 and 1.4 using the following;-
any reason why this would happen, should I report a bug on the issue
tracker?
Robb
On 17 June 2011 19:55, Richard Mudgett rmudg...@digium.com wrote:
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged
isdn
Hi
I'm using asterisk 1.6.2.18.1
I'm having a problem where only the first four digits are collected in the
cdr when the call is dialed overlap but if the call is dialed en-block the
whole dialed digits are recorded
chan_dahdi.conf
[trunkgroups]
[channels]
language=uk
switchtype=euroisdn
On 18 Jan 2013 15:22, Klaus Darilion klaus.mailingli...@pernau.at wrote:
Hi!
I want to forward a call to another destination if the outgoing call leg
has an rtptimeout. But as far as I see there is no way to find out if the
hangup was due to a rtp timeout or any other reason. I thought that
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