Displaying 20 results from an estimated 1000 matches similar to: "Asterisk crashes on high IO load"
2010 Nov 11
3
Limit Call Duration with L-option of Dial : announcement
Hello,
Limiting the call duration with the L-option of the Dial()-command is
working fine, however the announcement is not played.
Dialplan :
exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000))
The call lasts for 11 seconds, but 5 minutes before time runs out an
announcement should come. I hear no announcement, not on caller-side nor
on
2011 Mar 02
1
GSM-Card for Asterisk / recommendation needed
Hi,
I am trying to setup a GSM-Card for Asterisk. I currently use a "vgsm I"
from voismart (http://www.voismart.it/) but the driver is very bad
(compile-problems and no echo cancellation).
Is there anybody out there who can recommend me another piece of
hardware (pci card)? I need 1 or better 4 gsm-ports. Should be stable
and have an echo cancelltaion feature. And of course it
2011 Jun 08
1
PRI hangup request, cause 18
We have 2 PRI from AT&T
And all is well but only few numbers having following issue. We are getting hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and this issue raised
[Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got hangup request, cause 18
[Jun 7 17:57:10] DEBUG[24856] sig_pri.c: Not yet hungup... Calling hangup once with icause,
2011 Mar 03
2
Converting MP3 files to wav for Asterisk
Hi,
I am running a service where I play full songs but MP3 files kept on
crashing my server. I resorted to wav but the quality is really poor
after converting..or even sometimes not audible at all! Do you guys
know of a better way I can convert mp3 to wav and restore quality?
Below is the script I am using, I also tried the steps at
2010 Dec 15
5
Which version to use: 1.4 or 1.6 or 1.8
Hi All;
I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8?
For example, when to decide that I have to go for 1.6 or I have to go for 1.8?
Regards
Bilal
2011 Nov 21
2
Continue AGI after Dial() following caller hang up?
Hello,
We would like to continue a Perl AGI after a Dial() it has done completes
following caller hangup. We would like to do this in the same AGI, and not
using a new AGI from the 'h' extension. It works fine when the called party
hangs up and the 'g' option is used, but not for caller hangup.
Is this possible?
If not a confirmation that this is the case would be very helpful.
2017 Feb 16
2
Soft SIP phones that support TLS - Asterisk version 13.13.1
Microsip (Windows) is free and small.
2.5Mb download, 10Mb RAM usage, does everything I need and configuring
TLS is a doddle.
http://www.microsip.org/
On 16 February 2017 at 13:04, Max Grobecker
<max.grobecker at ml.grobecker.info> wrote:
> Hello,
>
> I'm a big fan of PhonerLite.
> It's more poplar in Germany, but also available in English language.
> This client
2017 May 06
2
Need to restart Asterisk if remote server not working?
Max Grobecker <max.grobecker at ml.grobecker.info> schrieb:
Hello Max,
> I'm also a customer of the DTAG.
> Yesterday, the messed a bit with their DNS entries...
Maybe they tried again to repair a working system... :)
> If you are NOT using their DNS resolvers you got a "wrong" IP address back
> that was not working. Besides that, you should disable SRV lookups
2016 Sep 15
2
Tricking asterisk to think the call has ended, but it was continuing on the other side
No, there is no Music On Hold starting and the bad thing is the call
duration reported by asterisk was just few seconds while the call duration
reported by the provider was few thousand seconds, the max allowed. So they
will be able to terminate the call on the asterisk side and have it run on
the provider side.
Leandro
2016-09-15 19:18 GMT+02:00 Max Grobecker <max.grobecker at
2017 Mar 29
2
How to have callers not being billed when in waiting queue ? [SOLVED]
Thank you very much, Max, for this valuable and informative answer.
Offline billing must be quite complex to set up as several telco may be
involved (or origination,transit or termination).
Moving to normal landline fare seems much simpler !
Thanks again
2017-03-28 21:41 GMT+02:00 Max Grobecker <max.grobecker at ml.grobecker.info>:
> Hi,
>
> in Germany, this kind of regulation
2014 May 24
1
"transmit_silence" not properly recognized on 1.8 ?
Hello,
I've got a problem at the moment, that setting "transmit_silence = yes"
seems to have no effect on Asterisk 1.8-Certified.
Although it's enabled and "core show settings" confirms, that it is
really enabled, there are no RTP packets sent by Asterisk when waiting
for DMTF input or when "Wait()" is called.
Also, there seems to be a small gap of 2 or 3
2012 Jun 18
4
Asterisk 1.8.13.0 / problem with cdr logging (mysql, odbc)
Hi,
I am trying now for over 4 hours setting up cdr-logging via odbc into a
mysql database. But with no success. Do you have any hint for me?
cat /etc/odbc.ini
------------------
[MySQL-asterisk]
Description = MySQL ODBC Driver
Driver = MySQL
Socket = /var/run/mysqld/mysqld.sock
Server = localhost
User = xxx
Password = xxx
Database = asterisk
Option = 3
Port =
and
/etc/odbcinst.ini
2013 Jan 23
1
DAHDI: How to supress notification of changing CallerID on transfer?
Hello out there,
I'm running an Asterisk 1.8.15-cert1 with DAHDI.
Today I noticed that Asterisk is signalling to the calling party the
current internal CallerID whenever I put a call to another internal phone.
Example:
Customer calls 020212345-555
-> IVR answers and puts caller to the chosen queue
-> Someone picks up the phone (Internal ext. 321)
-> CallerID shown on customers
2014 Nov 26
5
Strange Issue: asterisk deleted
Hi,
I looked for asterisk in /usr/sbin using the commands ls and find and whereis and it was not there.
I know that the process is killed because when I start asterisk using the command asterisk -vvvvc it starts and then it exits and the word killed is wrote on the console.
Ever time I copy a new executable to /usr/sbin either using cp command or make install it gets deleted too.
Now I used
2013 Feb 18
3
Dialplan / check / tool
Hi,
I am wondering, if there is any tool available, which performs a check
for suspicious entries in the dialplan. For example a non existing
AGI-Script or a double assigned extension ike that:
[context]
exten => *100*,1,AGI(test_app.pl)
...
exten => 190,1,Answer()
...
exten => *100*,1,AGI(never_reached.pl)
...
A "normal dialplan reload command" would echo no warning or
2013 Sep 03
1
Sip-Client / type=peer / Why can this client place calls?
Hi,
I am using Asterisk 11.5.1. As far as I understood, the following
configuration allows a sip client only to receive calls (type=peer) but
not to place calls
(http://www.voip-info.org/wiki/view/Asterisk+sip+type). Why can I place
calls though with this config?
sip.conf
...
[thorsten]
type=peer
host=dynamic
context=my_context
nat=force_rport,comedia
secret=...
dtmfmode=rfc2833
disallow=all
2014 Sep 18
1
Voice-Recognition / ASR / with barge in
Hi there,
I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine
:-) But I am wondering if there is a solution/application which will
enable me to implement voice recognition while playing a voice file
(barge in). So that the caller hears a voice file and can interrupt it
with his voice.
Currently (on our platform) the caller has to wait for the end of the
voicefie. Then we play
2011 Mar 03
2
Sangoma PCI vs PCI Express card
Hey Guy,
I have quick question. I am purchasing Sangoma A102D card but i am confused between PCI and PCI Express. Which card would be good for me.
Definitely PCI Express is advance but i just want to know is there any major difference, like quality, performance etc..
-Satish
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2010 Nov 12
1
Asterisk 1.6.20 / CDR issue with app-dial / bug or feature?
Hi,
it's me again with a cdr-issue. I have the following example
extensions.conf:
# dial in
exten => 100,1,Playback(hello)
exten => 100,n,Dial(local/200,20,rtg)
exten => 100,n,Playback(pleasewait)
exten => 100,n,wait(10)
exten => 100,n,Playback(goodbye)
exten => 100,n,Hangup
# for local dial
exten => 200,1,Playback(hello)
exten => 200,n,wait(10)
exten =>
2013 Sep 03
1
Asterisk 11.5.1 / TLS and Media Encryption / Blink as Client / no audio
Hi,
I use Asterisk 11.5.1 and it works fine. :)
Now I want to use TLS and media encryption. I followed this guide:
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
When I place a call via Blink-Client (0.5.0) I get connected and Blink
shows 2 locks. The blue lock shows "Signaling is encrypted using TLS"
and the orange lock shows "Media is encrypted using