Why do you want to use asterisk? It would be better to use kamailio or
opensips for that purpose.
09.06.2012 1:32 пользователь Ignacio Gonzalez mylan...@gmail.com
написал:
Hi everybody, I want to use asterisk for build a SIP Proxy and REGISTAR,
Is AsteriskNow a Sip Proxy already? or, do i
Don't blame me guys, but if you are a new to VOIP and aint't using asterisk
for a while, checkout freeswitch (freeswitch.org). Also do not forget at
least about this
http://www.opensips.org/html/docs/modules/1.6.x/load_balancer.html for
loadbalance to your pbx boxes. Just my 2 cents.
2012/5/9
AFAIK, this version is EOL. Sorry.
2012/5/4 Jonas Kellens jonas.kell...@telenet.be
**
Hello,
where can I place an Asterisk 1.6.2 backtrace taken from a core dump file ?
The file is to big to pass to this mailing list (it exceeds the maximum
kb).
Kind regards,
Jonas.
--
Check it out, thank you.
3 апреля 2012 г. 20:27 пользователь Niccolò Belli darkba...@linuxsystems.it
написал:
Hi,
If someone is interested I made Debian Squeeze Packages:
http://www.linuxsystems.it/**2012/04/asterisk-1-8-11-0-**
debian-squeeze-packages-with-**t-38-gateway-queue-hints-and-**
It seems like simle deadlock.
2012/2/3 Jonas Kellens jonas.kell...@telenet.be
**
Hello list,
using asterisk 1.6.2.22
What is wrong with Asterisk when the CLI becomes unresponsive ?!
I can login to the CLI with /usr/sbin/asterisk -r but whatever I type,
nothing happens.
The only thing
http://sangoma.com/products/hardware_products/digital_analog_hybrids/b601.html
2012/1/23 bilal ghayyad bilmar...@yahoo.com
Hi All;
Is there a telephony card that contains analoge ports and E1s at the same
time?
Regards
Bilal
--
ldd
2012/1/16 Olivier oza_4...@yahoo.fr
Hi,
I've recently upgraded a system from 1.8 to asterisk 10 and also
updated spandsp while doing so.
I wondered what is the safest and easiest way to check from command
line which libraries a running Asterisk system is currently using
(just like
Externip support per device in sip.conf
http://edvina.net/products/edvx/
2011/12/20 giovanni.v i...@keybits.org
Il 20/12/2011 6.07, Anton Kvashenkin ha scritto:
you can add exterin= in sip.conf for each trunk
I think this can be used only in [general] section not on peers
definition
AFAIK you can add exterin= in sip.conf for each trunk, correct me if i'm
wrong.
2011/12/20 Douglas Mortensen d...@impalanetworks.com
Hello,
** **
I have a SIP provider whom I may want to have multiple trunks with, rather
than just adding more channels to the individual trunk. I have
module show like sql
ls -l /usl/lib/asterisk/modukes/*sql*
2011/11/29 mahesh katta maheshka...@flexydial.com
yum info mysql* check with this
Reg:
Mahi.
On Tue, Nov 29, 2011 at 3:20 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
hello list
i have one question how to check
This is the right place to ask the question: what is the best practice to
run asterisk safe_asterisk-like method, but without _using_ safe_asterisk
shell script. Actually, what is safe-asterisk doing? It restarts asterisk
process and renames core dumps for better understanding of the scenario of
I think that you should use asterisk with this compile flags when you
actually have deadlocks or often crashes.
2011/11/23 Stefan Schmidt s...@sil.at
Am 23.11.11 11:39, schrieb Ishfaq Malik:
Hi
How much impact on performance do DONT_OPTIMISE and BETTER_BACKTRACES
have on a busy (13000+
Maybe this can help
http://www.russellbryant.net/blog/2011/03/04/debugging-the-asterisk-dialplan-with-verbose/
2011/11/14 Tristram Cheer trist...@tristramcheer.com
Hi All,
Hopefully this is considered on-topic for this list.
I'm using DumpChan(1001) in a Macro called debug in order to debug
Is anybody using pci-passthrough?
2011/11/9 Nick Khamis sym...@gmail.com
Hans,
Thank you so much for your response. We will be moving everything to VM
soon.
Cheers,
Nick.
On Tue, Nov 8, 2011 at 6:11 PM, Hans Witvliet aster...@a-domani.nl
wrote:
On Mon, 2011-11-07 at 11:45 -0500,
Turn off faxdetect on this peer.
2011/11/2 Christian Tardif christian.tar...@servinfo.ca
Hi,
I have a 1.6.2.6 fax installation with a FFA license which seems to be
installed correctly (in fax show stats, I see that I have 1 Digium G.711
licensed channel, and 1 Digium T.38 licensed
Which phone model do you use? 192.168.33.90 doesn't belong to
192.168.23.0/24, so the first thing that i will be doing is check routing
tables on the phone. Or for more help you could draw you network layout and
poste here.
--
_
Not info about networl settings. Please give output of
ip l
ip -4 a
ip ro
2011/11/1 Danny Nicholas da...@debsinc.com
Hello listers,
Another opportunity presents itself in my 1.4 to
10.0 conversion. My asterisk is set up for 192.168.23.xx and most of my
phones
Capture pcap with tshark or tcpdump for the future analysis with wireshark.
Ngrep is also handy tool for captaring, say, INVITE. You can use grep like
this: tail -f /var/log/asterisk/full | egrep --color -w
'chan_sip.*SIP/911|pbx.*SIP/911'
Interesting technique from Astresk Cookbook, Debugging
You can use tcpdump portrange 1-2 udp
2011/10/20 Andrew Higgs andrew.m.hi...@gmail.com
Hi Isabel,
Could you not just filter out after the fact using something like
Wireshark?
Regards
On Thu, Oct 20, 2011 at 3:28 PM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Dear all,
** **
What do you use _now_ to run asterisk: safe_asterisk, init-script, from
command-line? What distribution do you use?
To run asterisk from command line as user asterisk, just run asterisk -U
asterisk (asterisk user should be created), or edit /etc/asterisk.conf to
run as user asterisk. In production
Run tcpdump with portrange 1-2 (or what range of ports you use for
rtp) and dial test ivr to see what happening.
2011/10/17 gincantalupo gincantal...@fgasoftware.com
**
Hi John,
there is no firewall:
snom -- pbx -- patton -- pstn
It happens ONLY with IVRs. Normal calls are fine.
dahdi show status?
2011/9/27 salaheddine elharit salah.elharit...@gmail.com
thank you evryone for you help and support
i make this command lspci -d d161:* and i get :
09:02.0 Communication controller: Digium, Inc. Wildcard TE210P dual-span
T1/E1/J1 card 3.3V (rev 02)
that is mean i have
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