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
,
Stanisław Pitucha, Gradwell Voip Engineer
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2009/9/9 Stanisław Pitucha s...@gradwell.net:
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 the channels here
2009/9/14 Matt Riddell li...@venturevoip.com:
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
2009/9/14 Olle E. Johansson o...@edvina.net:
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
to know which side did the
transfer - but whichever side does it, I get back to context 'default'.
Any ideas?
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Kind regards,
Stanisław Pitucha, Gradwell Voip Engineer
T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com
Gradwell – Internet for Business People
Phone
;)
Exact scenario I'm using is described in the bug:
https://issues.asterisk.org/view.php?id=15833
Thanks for any help.
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Kind regards,
Stanisław Pitucha, Gradwell Voip Engineer
T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com
Gradwell – Internet for Business
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.
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Kind regards,
Stanisław Pitucha, Gradwell Voip Engineer
T: 01225 800 831 | F: 01225 800 801 | E: s
2009/8/26 Paul Herman her...@cleverbridge.com:
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 vindex+lists-asterisk-us...@apartia.org:
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
) 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 | Email Website Hosting
Can
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
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
- 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 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
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 -
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
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=948d3923bf2df47eca17c572713af2c7, opaque=
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.
Stanisław Pitucha
Gradwell Dot Com
- 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)
Stanisław Pitucha
Gradwell Dot Com
proper site for project... so I won't spam this maillist
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Stanisław Pitucha
Gradwell Dot Com
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