omers
> is to have registration interval in SIP client to be 120 seconds, but I'm
> very interesting in to have a list of ill behaving routers.
>
> On Sunday 16 December 2007, Olle E Johansson wrote:
>
>> 16 dec 2007 kl. 09.57 skrev Pavel Jezek:
>>
>>&g
Olle E Johansson wrote:
> 16 dec 2007 kl. 09.57 skrev Pavel Jezek:
>
>
>> shouldn't we have also qualifyfreq configurable?
>> default 'sip ping' frequency 60s isn't enough in many envrironments,
>> because udp nat translations on firewalls
shouldn't we have also qualifyfreq configurable?
default 'sip ping' frequency 60s isn't enough in many envrironments,
because udp nat translations on firewalls often timeouts quicker...
PJ
SVN commits to the Asterisk project wrote:
> Author: oej
> Date: Sun Dec 16 02:19:38 2007
> New Revision:
http://svn.digium.com/svn/asterisk/team/bbryant/sip-tcptls
1) I'm testing this branch with eyebeam softphone and sip over tcp,
annoying issue is, that peers are after some minutes unregistered
without any reason (I must restart softphone to register with asterisk
again)
logs are continuously fil
Russell Bryant wrote:
> Pavel Jezek wrote:
>
>> - not working when bridge two channles with different jb implementation
>> - eg. sip/h323/skinny & iax
>>
>
> I'm not exactly sure what you mean here. You generally don't want to use a
> ji
Mihai Balea wrote:
>
> On Oct 10, 2007, at 12:47 PM, Pavel Jezek wrote:
>
>> generaly jitterbuffer in asterisk have some design problems, eg:
>> - not working when bridge two channles with different jb implementation
>> - eg. sip/h323/skinny & iax
>> - jb is a
I'm using one pickup group for each office with more than one user ;-)
Eric "ManxPower" Wieling wrote:
Just how many pickup groups do you need?
If you are assigning one pickup group for each extension then your
design is wrong.
Pavel Jezek wrote:
if this limitation is rea
if this limitation is really true, it is challenge for some rework,
because with 64 pickup groups limit, it's usefull only for small
companies :-(
Philipp Kempgen wrote:
Dov Bigio wrote:
Is there any possibily of having more than 0-63 pickup/callgroups
Not unless you invent so
l a script and saved the result into an Asterisk variable.
http://www.pbxfreeware.org/app_backticks.c
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks
Regards,
## nini @ www.modulo.ro ##
Pavel Jezek wrote:
any idea, how to do something like this, but in correct/functional
form? ;-)
quite off topic, but would be nice, if sip show/list peers shows also
some text description for each peers (eg. text from callerid= or new
field description= in sip.conf?)
PJ
Steve Edwards wrote:
On Wed, 25 Oct 2006, Kevin P. Fleming wrote:
Steve Edwards wrote:
What was the inspiration
John Lange wrote:
Specifically, callers trying to record their voicemail greetings will
have jittered audio.
Setting jbenable=yes in zaptel.conf in order to dejitter sip audio is
very confusing and I know there will be a _lot_ of people wondering
about this besides me.
I strongly agree wit
Hi, you can use chan_skinny, that is included with asterisk,
today, maybe haven't so many functions like chan_sccp but in can be
better, if more people will use it and improve it.
it has also great advantage, that is fully supported and always working
with new asterisk versions.
PJ
__
2/bill-52'
== Spawn extension (default, 324, 3) exited non-zero on 'IAX2/honzat-50'
-- Hungup 'IAX2/honzat-50'
Andrew Kohlsmith wrote:
On Tuesday 05 September 2006 08:27, Pavel Jezek wrote:
I'm experiencing iax2 connection drops after three-ten minutes
(c
I'm experiencing iax2 connection drops after three-ten minutes
(connection was successfully established before drop)
connection is dropped in one direction only, so one site hear, but
oposite site no,
this messages appears in asterisk console
any idea?
PJ
issue do not appear in asterisk 1.
Asterisk SVN-trunk-r41849
kernel 2.6.17-2mdv
when entering application waitexten _and_ if moh class name is
specified, asterisk crashes,
waitexten without moh class, e.g. simple WaitExten(10|m), is working
fine (but no moh is played)
's' =>1. Playback(beep)
gcc version 4.1.1 20060724
GNU Make 3.81
make[1]: Nothing to be done for `all'.
make[1]: Nothing to be done for `all'.
[LD] abstract_jb.o acl.o aescrypt.o aeskey.o aestab.o alaw.o app.o
ast_expr2.o ast_expr2f.o asterisk.o astmm.o autoservice.o callerid.o
cdr.o channel.o chanvars.o cli.o conf
Hello developers,
according to discussion in -users list in the past, seems that
possibility of setting some minimum jitterbuffer value in iax2 new
(current) implementation will be very welcome.
I think, that can't be very hard to implement this user configurable
treshold parameter, but can so
very interesting, so it means that this "bridge code" is currently in
asterisk, or you have some patch for this?
I would like to test this! :-)
PJ
Paul Cadach wrote:
I just acknowledge we have H.323 native bridge code that support RTP "move"
(like re-invites for SIP) and all works fine
betwee
Hello,
I'm testing asterisk branch from Olle
(http://svn.digium.com/view/asterisk/team/oej/test-this-branch/README.test-this-branch.html),
but still crashes with chan_sccp (http://chan-sccp.berlios.de), when I
tried to make call from ci$co phone (skinny firmware),
I'm not developer, so I can o
Hi, sorry, if I'm reporting this problem to incorrect mailing list...
PJ
gcc -D_FILE_OFFSET_BITS=64 -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include
-D_REENTRANT -D_GNU_SOURCE -O6 -Wno-pointer-sign -march=i686
-DZAPTEL_OPTIMIZATIONS
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