Hello, what about: This feature means you can restart Asterisk after
a failure (or asterisk restart itself with safe_asterisk), and keep
existing calls up with only a few seconds of audio dropped. That
would be a feature! there is a other pbx that has this feature...
Anyone else would like to see
I use the latest spandsp source from the freeswitch git.
There you have also a changelog documenting the differences. Steve Underwood
commit here the latest changes in spandsp source.
http://fisheye.freeswitch.org/changelog/freeswitch.git/libs/spandsp
Kristijan
2012/1/11 Olivier
remove the c argument
Kristijan
2011/10/7 Administrator TOOTAI ad...@tootai.net:
Hi,
I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken from
deb http://packages.asterisk.org/deb lucid main) including dahdi from this
same repository. No FFA involved.
On incoming calls
Gergo, why do you want to use mISDN? Use Dahdi. Or do you want to use
a very exotic
isdn card which is only supported by mISDN? tell us more.
My long time experience with mISDN_v1 is, v1 has major echo and fax
problems. because the audio signals are transported very unsynchronic
because of the
?
On Tue, Sep 13, 2011 at 7:37 PM, Kristijan Vrban vrban.l...@googlemail.com
wrote:
hello Virendra,
thx for your response. but after i made clear to the carrier that i
want the dmtf only via rfc2833
and not via rfc2833 and SIP-INFO simultaneously, the problem is fixed.
Kristijan
2011/9
.
On Fri, Aug 26, 2011 at 3:11 PM, Kristijan Vrban vrban.l...@googlemail.com
wrote:
Hello, i have a carrier that send DMTF via rfc2833 and SIP-INFO
simultaneously. That has the effect, that asterisk read every dtmf
twice. and yes, it's mainly the carriers mistake. but is there a
configure option
Hello, i have a carrier that send DMTF via rfc2833 and SIP-INFO
simultaneously. That has the effect, that asterisk read every dtmf
twice. and yes, it's mainly the carriers mistake. but is there a
configure option, that asterisk accept only one DMTF method for
inbound dtmf?
Kristijan
--
use gdb (The GNU Project Debugger) to take a look into the core dump
gdb asterisk core.sip.pbx.tld-2011-08-26T08:07:35+0200
Kristijan
2011/8/26 Jonas Kellens jonas.kell...@telenet.be:
Hello,
Today I restart the MySQL-DB (/sbin/service mysqld restart) and I could no
longer connect to
did you find al solution for this issues? i fight with the same problem.
kristijan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Hello is use realtime sip-peers in 1.8, and have the problem, that when a peer
is loaded from database, the devstate is AST_DEVICE_UNAVAILABLE and
the the peers
can not be called from the queue. because the app_queue only calls
agens in state
AST_DEVICE_NOT_INUSE or AST_DEVICE_UNKNOWN.
My
http://code.google.com/p/zaphfc/
2011/5/23 Patrick Lists asterisk-l...@puzzled.xs4all.nl:
Hi,
I would appreciate some advice on the following: how does one use a single
BRI port HFC chipset based ISDN cards with the latest DAHDI, libpri and
Asterisk 1.8?
For Asterisk 1.4 I would first
choise. I had remembered when we shifted 1.2 to first
release of 1.4 and we had many issue. Same thing right now I'm dealing with
1.8 things take time to stabilized.
Good luck!!
--
Sent from my iPhone
On Apr 15, 2011, at 8:33 AM, Kristijan Vrban vrban.l...@googlemail.com
wrote:
Security only
Security only fixes: 2011-04-21 So in six days, no more bugfix patches will
committed into 1.4-branch :(
Is a prolongation possible? Because 1.4 is so reliable now. It would
be a great loss.
And no, 1.8 is not (yet) a replacement.
Kristijan
--
also the Answer is redundant
in logger.conf console = warning,error,notice,debug,fax
2011/1/24 David Backeberg dbackeb...@gmail.com:
On Mon, Jan 24, 2011 at 2:53 PM, Bryant Zimmerman brya...@zktech.com wrote:
I am testing out inbound faxing using res_fax and res_fax_spandsp.so
My system
there is no reason not to use 1.8 when you start a new installation.
1.8 is the new five years long term support version
Kristijan
2010/12/15 bilal ghayyad bilmar...@yahoo.com:
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
3 for advanced options does not have an effect _while_ the
vm-message is played. all other options like 7 delete or 6 next
message are
working. after the vm-message is played, it's working. the question:
is this intentional? or a bug?
the available documentation does not describe this case.
if you dont need asterisk as a fax maschine, just disable
res_fax_spandsp with make menuconfig.
if you want fax support, first remove all old spandsp lib/header, and
install the latest spandsp on you system from:
http://www.soft-switch.org/downloads/snapshots/spandsp/?C=M;O=D
then again
use the init script from debian
http://svn.debian.org/viewsvn/pkg-voip/asterisk/trunk/debian/asterisk.init?revision=8502view=markup
the one from the asterisk source seems to be broken, if have the same issue
kristijan
2010/8/16 unsero...@aol.com:
I am using Debian Lenny, not RedHat.
Hello, i just had some fax abortions because of some packet loss. so i
startet to examine in the pcap recording
from the res_fax_digium, if the T.38 EC mode redundancy was really
used. So i watched into it, and compared it
with a t.38 pcap from spandsp (same asterisk setup, but with app_fax)
and i
Hello, i just made the reproducible watching:
I send a Fax from asterisk (trunk) with spandsp (latest snapshot) via
T.38 - Audiocodes Mediant 2000 (FW 5.60.43.5) - PSTN Fax
With Internal timing Enabled, the Fax break after the first quarter
from the first page is transfered.
With Internal timing
Hello, when is use SIP realtime and i (re)start asterisk, then all SIP user are
Unavailable, and never go to Not in use because the phones are
registered on opensips.
And res_config_mysql does load the user only, when the SIP does a call
(or get called)
an then chan_sip give app_queue the
I can confirm this issue. And my setup is similar to yours.
kristijan
2010/4/19 Positively Optimistic positivelyoptimis...@gmail.com:
Good day..
We have what I consider to be a large dialplan (-= 1501 extensions (2559
priorities) in 99 contexts. =-)
If we have more than 10 or so channels
Asterisk 1.4.29
BLF-SUBSCRIBE go to internal IP (ngrep output):
U 2010/03/02 11:34:06.013515 212.78.xxx.xxx:2048 - 62.134.xxx.xxx:5060
SUBSCRIBE sip:1...@62.134.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP
212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport..From:
good question. i never investigated this issue more exact. Any other
T.38 more knowing here if this is possibly anyway?
Kristijan
2010/2/12 Deepesh D deep.d2...@gmail.com:
Hello,
Is it possible to use asterisk in T.38 pass through mode with reinvite?
My fax calls are getting disconnected
it's because after ReceiveFAX (i use asterisk 1.6.2.1 with
spandsp-0.0.6pre17), it jumps to the hangup exten. So this is my
dialplan for ReceiveFAX:
[fax-in]
exten = s,1,ReceiveFAX(/tmp/fax-${CDR(uniqueid)}.tif)
exten = s,n,Hangup()
exten = h,1,NoOp(### FAXSTATUS: ${FAXSTATUS})
exten =
On a well set up system you should be able to send or receive those
pages all day. If you can't, you probably have timing issues in your
Asterisk setup.
This is a uncleared question. What does timing issue exactly mean?
1) Enable internal timing and use one of the res_timing_*.so (with
timing source.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kristijan
Vrban
Sent: Friday, January 08, 2010 4:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
super quick asterisk in gdb howto:
compile asterisk with DONT_OPTIMIZE (in make menuconfig - Compiler flags)
gdb asterisk
run -cvv
wait for the crash
bt
bt full
and now make the patch :)
Kristijan
2009/12/24 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Dec 24, 2009 at 12:13:55PM
magnus, simple answer: just use the latest version available. and if
something is not working inside the t.30/t.38 protocol, try the latest
spanpshot: http://www.soft-switch.org/downloads/snapshots/spandsp/?C=M;O=Dand
if something i still not working, give a good description how to
reproduce the
Hello, short question: is there a possibility to use asterisk as an outbound
proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly
workarounds, everything.
What is want to build is:
SIP Phone - via TLS/SRTP - Asterisk as outbound proxy - via UDP/RTP -
VoIP-Provider
So
-oslec
Kristijan Vrban
2009/8/27 DHAVAL INDRODIYA dhaval.it01...@gmail.com
hi all,
any one know, about echo cancellation with digium card,
is it actually needed or it okay if we dont purchase because it increase
price which half of new card,
regards
Dhaval
hello, i made a experimental patch for libpri to have NT/PTMP mode,
answers please on asterisk-dev at:
http://lists.digium.com/pipermail/asterisk-dev/2009-May/038455.html
Kristijan
2009/5/14 Kristijan Vrban vrban.l...@googlemail.com
good news, i just made my isdn device ring! ok, after
some basic nt_ptmp
functionality. Stay tuned :)
Kristijan
2009/5/12, Tzafrir Cohen tzafrir.co...@xorcom.com:
On Tue, May 12, 2009 at 08:05:49AM +0200, Olivier wrote:
2009/5/12 Kristijan Vrban vrban.l...@googlemail.com
For those also need NT over PtMP, i started a initial patch for it. Very
For those also need NT over PtMP, i started a initial patch for it. Very
limited at the moment, only one incoming call to chan_dahdi from one
device is possible. But i was pleasantly surprised that NT-ptmp is working
anyway
Get the patch here: http://bugs.digium.com/view.php?id=15048
Kristijan
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