Hi Team,
I would like capture SS7 Error Code in CDRs, Specifically of outbound call
from the asterisk. calls generated using .call file.
In extension.conf extens gets excuted on successful call only , So that on
h extension reason of hangup is captured. But i am not aware of any
provision that
Hello list,
We are interested in configuring an Asterisk based GSM server in Africa
an I would like your recommendations. The requirements are:
1) A system with at least 8 GSM cards
2) 2 pstn lines from here (the US) will directly link with 2 lines on
the GSM server and the rest of the U.S.
I am setting up my dialplan with quite some outbound numbers. We have a
block of 100 DID's, for which some of them will go direct to specific
phones. I am struggling how to solve this, so I am searching for a little
advice. These are my concerns.
I could set the DID in the sip.conf using
On Tuesday 27 March 2012, Roland wrote:
I am setting up my dialplan with quite some outbound numbers. We have a
block of 100 DID's, for which some of them will go direct to specific
phones. I am struggling how to solve this, so I am searching for a little
advice. These are my concerns.
I
On 3/26/2012 1:11 PM, bilal ghayyad wrote:
If it possible, then is it possible to be a configuration per user?
Just expanding on Jim's answer-
to allow user example with password secret from 192.168.0.*, do
something like:
in /etc/asterisk/sip.conf:
[example]
type=friend
secret=secret
I would like to fetch my extensions from the database. I created a dynamic
hint, but doesn't seem to work. The BLF on my phone doesn't change when the
state of the extension changed.
This is in my dialplan:
exten = _ZXX!,hint,${SIP_BYEXT(${EXTEN},${CONTEXT})}
exten = _ZXX!,1,Verbose(3, Search
I have done this successfully in 2 ways depending on your requirements.
Usually, I just set the callerid number right in the SIP, this is the easiest
and cleanest in my opinion. Worth mentioning that I always set the callerid in
the SIP regardless, this way I know that internal calls, trunk
Hello
We have a Genband C3 Switch and a couple of customers that operate asterisk
PBXes connected via SIP Trunk. All of them still use some 1.6.X asterisk and
this works fine.
One customer uses a 1.8 version and has a very strange problem:
Asterisk 1.8.10.0-1digium1~lucid built by pbuilder @
Thank you for your answer.
First of all: I didn't name my sip accounts the same as my extension
numbers. So I cannot just Dial SIP/${EXTEN} in my case. Also this 100
number block is mapped to extensions for our primary location, but we will
be connection other locations where extension numbers
Hi
What should be the optimal settings for HWEC related parameters in
wanpipe1.conf to be able to receive Fax over T.30 on a Sangoma A108DE Card ?
In particular, should I enable echo ? If yes, to what parameter ? What
other values like RXGAIN, TXGAIN should I tune ?
TDMV_HW_DTMF
= YES
-- Executing [s@from-pstn-3:2] Verbose(DAHDI/3-1, CALLERID is
XX) in new stack
CALLERID is XX
-- Executing [s@from-pstn-3:3] Verbose(DAHDI/3-1, Time is
20120327-204307) in new stack
Time is 20120327-204307
-- Executing [s@from-pstn-3:4] Dial(DAHDI/3-1,
SCCP/1000SCCP
Hi All,
i am working on video setup within asterisk my simple question is asterisk
support RFC-5168.
if yes then in which version ?
thanks
Dhaval
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