Greetings,
We have an old analog phone system running Asterisk 1.2.13 (not my choice lol).
Everything has been working wonderfully until today. The site is experiencing
dropped and missed calls. When I tried calling the site, I did get through
however the CLI was flooded with hundreds of copies
Am 25.01.2012 um 20:24 schrieb Alec Davis:
> Great to hear it's working for others.
>
> Regarding inclusion into 1.8 branch, as it's a new feature, it would only
> ever go into trunk, unless there is an outcry from the community.
Outcry! :-)
> To assist others implementing this, and from a dif
Great to hear it's working for others.
Regarding inclusion into 1.8 branch, as it's a new feature, it would only
ever go into trunk, unless there is an outcry from the community.
To assist others implementing this, and from a different viewpoint, would
you mind documenting how you implemented it.
Hello Guys,
I am trying to convert files that are .wac to mp3 after mixmonitor command is
called but it doesnt execute the command, I tried the command in terminal it
worked, any help please ... below is my dial plan
exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F
-
Hi,
How can I play a sound file from the middle and end it after a certain
number of seconds?
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Read the documentation. If I do Set(a=${CHANNELS()}) a will return a list
of channels with a space between each entry like you originally said. I did
my test using Set(a=${CHANNELS(1107)}) and it returned a=SIP/1107, therefore
I logically assumed that ${CHANNELS(miq8)}) would return SIP/miq8. If
I appreciate your 2-cents worth.
However, I do not believe they have access to machine
If so, they are clever to create three failures in the logs for my benefit
before entering the correct one for hijacking.
Additionally, I have a lot of sip extensions to hijack and he keeps going for
This is actually an interesting concept however I do think I want to restrict
dialing during a specific time period.
If someone is in the office, I would have to reprogram the route so allow
dialing which adds overhead.
Again, I do like the concept though.
Thanks,
--E
From: aster
Can you please elaborate on rate limiting. Not how to implement it but rather
how implementation is beneficiary.
Reading up on it, it appears that it just checks the tcp connections and denys
connection if limit is passed.
In my thoughts, this is essentially a live fail2ban monitor in res
Hi
I'm using asterisk 1.8.7.0 on a ContOS 5.6 machine
Earlier today I spotted multiple occurrences of the following type of
warning
handle_response_invite: just did sched_add waitid(1955858) for
sip_reinvite_retry for dialog etc...
I recognised this as a hung channel, which I've not had since upg
Am 23.01.2012 um 23:25 schrieb Alec Davis:
>
>> How can I test this solution on a 1.8.8.1 system ?
>> If I'm not mistaken, diff
>> https://reviewboard.asterisk.org/r/1619 do not apply to 1.8.8.1.
>
> I've just checked out 1.8.8.1 and download my patch from
> https://reviewboard.asterisk.org/r/
Hi,
I was provided with a bunch of ip addresses to be configured on my asterisk
server.
the first batch contains the signalling which has 2 ip addresses
the second batch contains media which has 4 ip addresses
How will I configure it to my asterisk box? Should I simply add the
signalling to my s
I use ChanSpy successfully all the time. You do not have to specify the
full channel, just the prefix which is the peer name. As you can see it
also states 'This includes the audio coming in and out of the channel
being spied on.'
Have you tried giving it a go?
-= Info about application 'ChanS
This could work, yes.
But the context is not always the same.
Also ${CHANNELS(miq8) will return nothing...
Jonas.
On 01/24/2012 08:47 PM, Danny Nicholas wrote:
Did a little research on this using my Asterisk 10.0. This should
work for you.
exten => 1246,1,answer()
exten => 1246,n,set(
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