Hi
Login as user asterisk and then try it
e.g. sudo su asterisk
and then try to execute the command.
Last time I had this issue I had to set the shell environment for
asterisk with the
chsh
command
On Wed, 2013-06-19 at 13:03 -0500, Daniel - Asterisk wrote:
Hello everyone,
I'm trying
Hello,
I'm stressing an Asterisk 11 platform with incoming calls from sipp 3.1.
I've dedicated a context to sipp in my dialplan.
Everything works OK expect that calls from sipp comes in with a CallerID
set to sipp and this sipp value is stored in CDR.
1. I can change the value of the CallerID
On 11.06.2013, at 0:24, Sean Darcy wrote:
Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no
success:
Silk is enabled only after asterisk restart.
for silk work need codecs.conf with silk configuration
res_format_attr_silk.so - loaded
codec_silk.so - loaded
please
hi,
i need some small sip video endpoint for cloud videoconference (like
bluejeans)
i have this idea
VIDEO OUT
TV or projector with HDMI
VIDEO IN
cameras with h264 hw enconding
- http://downloads.element14.com/raspberry-pi-camera/
http://downloads.element14.com/raspberry-pi-camera/
-
Hi all,
I have two phones that I've been comparing (different manufacturers).
To debug call quality issues on one of them, I've been using calls from
the phone to our main DID, so 3 SIP sessions exist (phone asterisk
then asterisk provider, and the providerasterisk for the DID).
The bad
Have you checked whether the same codecs, or codecs with the same bandwidth
requirements, are used?
jg
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On 06/20/2013 11:56 AM, jg wrote:
Have you checked whether the same codecs, or codecs with the same
bandwidth requirements, are used?
Yes, it is ulaw everywhere.
- Mike
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On 06/20/2013 11:56 AM, jg wrote:
Have you checked whether the same codecs, or codecs with the same
bandwidth requirements, are used?
Here's an example of a simple outgoing call. Everything is ulaw. The
192.x.x.x phone has roughly twice the packet count of the provider
session. The lost
On Thu, 20 Jun 2013, Dr. Michael J. Chudobiak wrote:
Here's an example of a simple outgoing call. Everything is ulaw. The
192.x.x.x phone has roughly twice the packet count of the provider session.
Would running wireshark yield any clues?
--
Thanks in advance,
There's one thing on my list of things to check, that could be relevant for
your problem, too.
Packet count is one thing, transferred data is another one. If one phone uses smaller UDP
packages, then the packet count should increase in reciprocally. I have read some comments on
the net that
Packet count is one thing, transferred data is another one. If one phone
uses smaller UDP packages, then the packet count should increase in
reciprocally. I have read some comments on the net that smaller packages
are preferable because lost packages have less impact on the hearable
audio.
Aha.
On Thu, 20 Jun 2013, Satish Barot wrote:
Would you mind sharing a sample of your pthread-ed C AGI? This will help
someone like me who has written AGI in Perl/PHP and now exploring C AGI.
The source code for this particular AGI is about 600 lines and uses my own
AGI library (written before
Aha. I overlooked that some phones had ulaw:10 in sip.conf, instead of the standard ulaw:20.
That explains the packet count difference. It seems my call quality issues are coming from
something else.
... and this explains how to set the packet size. Answer to get answers, or so.
jg
--
Hi all,
I'm getting ready to setup SIP/TLS and SRTP. But I have a few questions.
The first one is that I was reading an article at:
https://supportforums.cisco.com/docs/DOC-15381
That indicated that Asterisk doesn't support TLS as an OPTIONAL transport.
It's either all or nothing.
Help!
I have providers configured that send me incoming calls via SIP. There's
one that seems to make trouble. As soon as I get a few concurrent calls
through this peer, Asterisk CPU load goes up to 250%, audio becomes
laggy and I get hundreds of these per second in the logs:
[Jun 20
I need to block any audio before there is a connect, in SIP. How do I tell
the DIAL application to behave like that?
Yours
Philip
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Please also have a look at the gateway boxes from berofix
(http://wiki.beronet.com/index.php/Main_Page). I am not affiliated but
have used different products from them over last few yeas and all have
survived and are stable.
Documentation is open and free on their wiki. They provide updates. They
Mike Diehl wrote:
Hi all,
I'm getting ready to setup SIP/TLS and SRTP. But I have a few
questions. The first one is that I was reading an article at:
https://supportforums.cisco.com/docs/DOC-15381
That indicated that Asterisk doesn't support TLS as an OPTIONAL
transport. It's either all or
Hello jg:
When mutt is called from Asterisk's dialplan there's no output at mail.log
When I use:
echo | /usr/bin/mutt -s New fax earohua...@gmail.com -a
${FAXDEST}/${tempfax} /tmp/ocurrencias.txt 21
replacing FAXDEST and TEMPFAX with proper values, the output is as follows:
Jun 20 16:16:16
Have you tried calling a bash script that in turns calls mutt. That way you
could debug much easier, adding echo to a log file.
Sent from my iPhone
On 2013-06-20, at 5:27 PM, Daniel - Asterisk earohua...@gmail.com wrote:
Hello jg:
When mutt is called from Asterisk's dialplan there's no
Hi Elder!
I am currently busy with a problem of one of my customers. I am pretty sure that your setup
requires only minor changes to make things work. I have several setups that use mutt exactly in
the same way as you are trying to do (except that I am using postfix instead of sendmail, but
On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp jc...@digium.com wrote:
Mike Diehl wrote:
Hi all,
I'm getting ready to setup SIP/TLS and SRTP. But I have a few
questions. The first one is that I was reading an article at:
https://supportforums.cisco.com/docs/DOC-15381
That indicated that
When I type: asterisk -rx core show channels
I usually get
Channel Location State Application(Data)
SIP/pstn--03 7807574622@internal: Up Dial(SIP/77807574622@pstn-9998
SIP/pstn-9998-03 (None) Up AppDial((Outgoing
Hi
You can do,
core show channels verbose
Kind Regards
On Thu, Jun 20, 2013 at 6:45 PM, Joseph syscon...@gmail.com wrote:
When I type: asterisk -rx core show channels
I usually get
Channel Location State Application(Data)
SIP/pstn--03
Eloi,
My responses are inline.
Thanks a lot for this detailed answer :
You're welcome. Thank you for responding. A lot of people forget to do so and
future list readers are left wondering whether or not the proposed solution
worked.
- I managed to have it working disabling auth message
What happens when we increase the queue frame size in channels.c
if ((queued_frames + new_frames 128 || queued_voice_frames +
new_voice_frames 96)) {
Be default it is 128 and 96 if i increase it to 256 and 192 what will
happen? will it impact to default behavior?
Regards,
Gopal.
--
On Thu, Jun 20, 2013 at 6:55 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
What happens when we increase the queue frame size in channels.c
if ((queued_frames + new_frames 128 || queued_voice_frames +
new_voice_frames 96)) {
Be default it is 128 and 96 if i increase it to 256
actually when i get the message my call volume is around 180 to 200
calls will that matter... and some calls being transferred to other
location and some are making outbound calls, some are inbound...
Is there any way for optimization?
On Fri, Jun 21, 2013 at 5:57 AM, Richard Mudgett
On Thu, Jun 20, 2013 at 5:10 PM, Mike Diehl mdiehlena...@gmail.com wrote:
On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp jc...@digium.com wrote:
Mike Diehl wrote:
Hi all,
I'm getting ready to setup SIP/TLS and SRTP. But I have a few
questions. The first one is that I was reading an
On Thu, Jun 20, 2013 at 10:54 PM, Steve Edwards
asterisk@sedwards.comwrote:
On Thu, 20 Jun 2013, Satish Barot wrote:
Would you mind sharing a sample of your pthread-ed C AGI? This will help
someone like me who has written AGI in Perl/PHP and now exploring C AGI.
The source code for
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