Hi,
Could someone tell me where are the good places in chan_iax to put trace
points when I experience strange delays in NEW processing?
I tried to output some debug after every stage of socket_process / case
IAX_COMMAND_NEW, but it all takes max 30ms. However, sometimes in a
normal call I get a
f sound itself will not change. Also, unless
you use a wideband codec, you won't be able to send more than 8000Hz
over the line for most standard codecs (u/alaw, gsm, etc.). Just play
the .wav you already have - you cannot get a better quality than that
in the original file.
--
KTHXBYE,
2009/9/14 Olle E. Johansson :
> Make sure that each device has a TRANSFER_CONTEXT dialplan variable.
What about a situation where sip devices register at a proxy in front
of many asterisks and asterisks authorise all calls from that proxy?
I.e. I don't have any devices that asterisk would know abo
2009/9/14 Matt Riddell :
> For every billable item we use a code for the account and store it in...
>
> accountcode :)
I'm not sure that actually answers my question... If you have a A->B
call and set accountcode for A on it, then B does a blind transfer,
how do you set the correct accountcode the
2009/9/9 Stanisław Pitucha :
> I've got different customers that may use the same asterisk. Each user
> can blind-transfer a call to whatever place they want. But of course
> the transferring side should be billed for it.
> What can I do to see the difference between
I'd like to know which side did the
transfer - but whichever side does it, I get back to context 'default'.
Any ideas?
--
Kind regards,
Stanisław Pitucha, Gradwell Voip Engineer
T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com
Gradwell – Internet fo
ne
noticed ;)
Exact scenario I'm using is described in the bug:
https://issues.asterisk.org/view.php?id=15833
Thanks for any help.
--
Kind regards,
Stanisław Pitucha, Gradwell Voip Engineer
T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com
Gradwell – Intern
ptor and the application
will still write to the file (it's just named differently).
The logger restart just does a { close(); open(); } and you're logging
to a new 'log' again. This way you don't lose any messages during the
rotation.
--
Kind regards,
Stanisław Pitucha, Grad
2009/8/26 Paul Herman :
> I use Bria and eyebeam and it seems that asterisk doesn't send RCTP
> keepalives when a SIP channel is on hold.
Slightly related: https://issues.asterisk.org/view.php?id=15466
It also affects integration with OCS for me.
___
--
Probably the easiest way: put an opensips box in front of asterisk. It
can handle multiple registrations on the same username. If you have
multiple registrations, it will do a parallel fork and work just like
you wanted.
You just have to make sure that phones register on opensips, not asterisk.
_
2009/7/24 Louis-David Mitterrand :
> This used to work fine in 1.4:
>
> exten => 2131/,1,NoOp(reject3: ${CALLERID(num)})
> exten => 2131/,n,Playback(no_unknow_callerid_here)
> exten => 2131/,n,Hangup
>
> And now, after upgrading to 1.6.1.x it matches every callerid.
I'm not su
rg/view.php?id=15466) but maybe someone here
has some ideas how to make it work?
--
Kind regards,
Stanisław Pitucha, Gradwell Voip Engineer
T: 01225 800 851 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com
Gradwell - Internet for Business People
Phone Services | Business Broadband | Ema
Hi,
I noticed something bad happening on our systems lately. We have lots
of asterisk threads running, but most of them are completely idle -
strace doesn't show anything happening there. The only thread doing
work seems to do everything. I see it sending mysql queries, writing
logs, sending both S
- "michel freiha" <[EMAIL PROTECTED]> wrote:
> Did you try "show translation"
That shows a table of times taken by translation... I'm asking about codecs
used by a channel on a certain call.
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Hi,
I'm trying to access audionativeformat / other codec variables in the hangup
handler of a call (with ${CHANNEL(audioreadformat)}), but I get no response.
Also 'core show channel ...' doesn't list those variables. Are they always set
by asterisk, or only in some scenarios? It's a simple SIP-S
Hi,
I was wondering if there's any sense in increasing audiobuffer above the
minimal '2' in meetme, if every channel is already dejittered before
(Local/.../nj - as described at:
http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/)
Will it help in anythin
Hello,
I've got a problem with rtp handling by siemens c450 and similar. I experience
a couple seconds of silence between early media and normal call (normal call's
rtp is dropped by phone). This is caused by SSRC changing (even though marker
bit is set). I have all relevant patches applied - it
Add /usr/sbin to your PATH, or run /usr/sbin/asterisk.
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Hey
I'm looking for an advanced scenario for sipp, that can be used for testing
asterisk. Mainly I'm interested in making random calls between sipp
pseudo-users. Did anyone try to do something like this?
Or has anyone got an example scenario with working loops?
Thanks
__
capi_counter++);
--->8---
Calls seem to work - did I break something big? Anyone else tried that before?
Stanisław Pitucha
Gradwell Dot Com
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- "Rizwan Hisham" <[EMAIL PROTECTED]> wrote:
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="584760da"
> Authorization: Digest username="bernart48", realm="asterisk", algorithm=MD5,
> uri="sip:[EMAIL PROTECTED]:9060", nonce="584760da",
> response="948d3923bf2df47eca17c5727
opping calls
- maybe some proper site for project... so I won't spam this maillist
---
Stanisław Pitucha
Gradwell Dot Com
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- "Tzafrir Cohen" <[EMAIL PROTECTED]> wrote:
> Interesting. One thing thoough: what's the license of your code?
It's MIT - I forgot to add that. I'll stick the banners to files soon, with
next update to the package. (along with some fixes, etc)
t do any other modification or audiostream translation - only message
passing.
If someone's interested -- code + short doc is available at
http://www.gradwell.com/tmp/iax_proxy.tar.gz
Development will continue - any opinions / comments / contributions are
appreciated.
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