Hi,
I use asterisk 1.8.
My issue is : I have the same SIP members added to two queues. I use realtime
configuration and has set the field ringinuse=0 for both the queues. But if an
extension is answering the call in one queue, and some new call comes in the
second queue, still that extension i
Hi,
where did you change the ulimit? The following command should show you,
if your setting is correct:
asterisk -rx "ulimit descriptors"
In my installation I edited the limits here:
vi /etc/security/limits.conf
[...]
asterisksoftnofile 8192
asteriskhard
Hello Everyone,
We are currently having talks with various service providers, and
trying to determine what the best way is to interconnect in order to
have access to the PSTN network. As you know there are two ways of
doing this:
Traditional PRI: Have trunks grouped into a transport layer such as
Interesting.
You might want to consider paying some expert for consulting ?
Mitul
On Jun 22, 2013 7:21 PM, "Nick Khamis" wrote:
> Hello Everyone,
>
> We are currently having talks with various service providers, and
> trying to determine what the best way is to interconnect in order to
> have ac
Thank you mitul.
N.
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On Sat, 22 Jun 2013, Larry Moore wrote:
echo >> "$MSGFILE"
printf "%18s" "Sender: " >> "$MSGFILE"; printf "%-20s\n" "${REMOTESTATIONID}" >>
"$MSGFILE"
printf "%18s" "Pages: " >> "$MSGFILE"; printf "%-20s\n" "${FAXPAGES}" >>
"$MSGFILE"
printf "%18s" "Signal Rate: " >> "$MSGFILE"; printf "%-20s\
