On Tue, Jul 26, 2011 at 12:37 AM, Nikhil d.nik...@cem-solutions.net wrote:
I am using asterisk as a client not as a server. For client I need features
like transfer ,call forward ,multiple lines as in normal IP Phones like
CISOC,polycom.
In asterisk ,we have chan_alsa driver that will
On Wed, Apr 13, 2011 at 11:34 AM, Andreas Sikkema h...@ramdyne.nl wrote:
On 4/12/11 1:21 AM, Don Kelly wrote:
Continuing top posting...
The same argument could be made for any commercial solution. Why use
Asterisk when we could throw $4,000 at our problem for a commercial
solution?
I'd
I understood that option worked the other way around so attacker
thinks peer name is invalid even when they hit a real one.
On Wed, Jan 19, 2011 at 2:23 AM, ad...@3a.hu wrote:
Hi List,
i've been receiving several sip registration probes in the last month, and
as this server is a testing site
On Fri, Jan 14, 2011 at 6:31 PM, Tim Nelson tnel...@fudnet.net wrote:
You've been officially added to my kill file [1]. The lists are here to get
suggestions and assistance with various issues [2]. They are *NOT* your one
stop shop for everyone doing your homework [3][4][5][6][7][8][9]. You
Sounds like they just need to be told its a hilariously bad idea to host
anything important on a cellphone.
On Mon, Nov 29, 2010 at 1:20 PM, Gordon Henderson
gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote:
On Mon, 29 Nov 2010, Gilles wrote:
Hello
Some SOHO prospects only
You didn't give full details...
which port is unreachable? 5060? some random RTP port? Did you forward udp
or tcp or both? Also why did you type in gmail when outlook asked you for
your name? :)
Is virtual server the same as dmz option?
On Sun, Nov 21, 2010 at 4:26 AM, gmail gres...@gmail.com
Why do you have A,(demo-thanks) shouldn't it it be A(demo-thanks)?
eg:
exten = s,n,Dial(SIP/jazzey/1703111,120,A(demo-thanks))
On Sat, Nov 13, 2010 at 6:38 PM, Thomas Perron thomas.per...@gmail.comwrote:
Here is a very very basic config. But, not working (:
I simply want to dial the DID
On Sat, Nov 13, 2010 at 8:00 PM, Thomas Perron thomas.per...@gmail.comwrote:
i am running 1.4.37 and am hosted on Rackspace.
I feel like a took a step back by using the Cloud server service since
I am having a little trouble proving that my basic configuration is
working.
Nevertheless, I
I fiddled with the demo version of swift a year or so ago and I had better
sound quality if I used the non-8khz versions and had app_swift or asterisk
convert it for me (not sure, giving app_swift a regular version seemed to
JustWork(tm)
On Sat, Oct 23, 2010 at 3:24 PM, Zeeshan Zakaria
On Thu, Oct 7, 2010 at 11:25 PM, Fazil Amaan fazil_d...@yahoo.com wrote:
Hi,
I cannot get asterisk to start again after the g729 install failed.
kindly advise what's the problem.
Thank's
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On Wed, Oct 6, 2010 at 7:03 AM, Zarko Zivanovic outlaw...@gmail.com wrote:
Hello,
I would need a little help about using 16 bit wav or mp3 files for moh on
asterisk 1.2.x
When i try to use these files as moh, the caller gets disconnected.
Please advise.
Regards,
Z. Zivanovic
On Tue, Oct 5, 2010 at 1:40 PM, Roger Burton West ro...@firedrake.orgwrote:
I now have an OpenVox A400P and it is working well. Thanks to Ade
Vickers for the recommendation, which I second.
However, I need to make a slow transition between a conventional
multiple-extension setup and a full
On Wed, Oct 6, 2010 at 12:50 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
This is such an annoying issue whenever it comes up. The sender and receive
always receive the source public IP no matter what in the IP packets but
then SIP packets go out with something like 192.168.0.20.
In
On Sat, Oct 2, 2010 at 4:37 PM, bruce bruce bruceb...@gmail.com wrote:
Thanks Roger.
I will be trying this box to see what I can do. Otherwise, I'd probably
have to find a list of all of the Rogers (The ISP providing internet to
these boxes) IPs to at least limit the attacks to Rogers ISP.
On Wed, Sep 15, 2010 at 6:04 AM, Jeff LaCoursiere j...@sunfone.com wrote:
On Tue, 14 Sep 2010, Joe Freeman wrote:
Anyone have a good provider for International (US/Canada at least) 800
termination/origination? I have a customer that had us port one of their
800 numbers and apparently
On Sun, Sep 12, 2010 at 10:43 AM, Richard Stuppi rich...@stuppi.com wrote:
I work in a small office and have fallen into the role of network support
based on knowing enough about networking to be dangerous.
Our office is moving from DSL to a T1. Were using Asterisk as our PBX and
I'm
On Sun, Sep 12, 2010 at 10:05 AM, Jonas Kellens jonas.kell...@telenet.bewrote:
Hello,
everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I get
the following :
[Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username
mismatch, have 32990900, digest has
I hadn't heard about them until your mailing list post.
On Thu, Sep 9, 2010 at 8:48 AM, Barry Fawthrop ba...@isscp.com wrote:
Does anyone have a packet capture of a 3Com 3102 phone registering with
an NBX that I could take a look at ?
What is the expected traffic flow, all I get is the
On Thu, Sep 9, 2010 at 10:25 AM, Carlos Chavez cur...@telecomabmex.comwrote:
Is there an archive of security advisories for Asterisk? We
recently
upgraded a customer from 1.2 to 1.4 and now they are asking for
documentation of all security and bug related fixes. I know the
On Mon, Sep 6, 2010 at 12:32 PM, Barry Miller asterisk-us...@notanet.netwrote:
After upgrading my small test system from Debian Etch-Lenny via a
complete reinstall, I find my g729 hostid has changed. Same machine,
same CPU, same NIC! It doesn't seem reasonable that I have to burn
my one
Looks like they have twitter. Its good that you mentioned them in the
subject unlike the guy who wrote an hour and a half ago with subject Global
Outage?
http://twitter.com/vitelity
http://twitter.com/vitelityWe are currently experiencing network
difficulty on Vitelity's core router. We are
On Wed, Sep 1, 2010 at 6:09 AM, Barry Fawthrop ba...@isscp.com wrote:
Has any advancement been made to get 3102 operational in either a SIP or
H323 asterisk environment.
A post back in time mentioned a downloader service.
From the posts and articles I have read, the NCP is acting like a bootp
On Sat, Aug 14, 2010 at 7:33 AM, Lyle McKarns
lyle.mcka...@nexusmgmt.com wrote:
By a mixed environment I mean some Asterisk servers running on AMD and some
running on Intel
Thanks,
Lyle J. McKarns
---
Networking/Linux Engineering Team
n|m Nexus
On Sun, Aug 1, 2010 at 10:34 PM, Janu Mukherjee janu.mu...@gmail.com wrote:
Hi all,
I have the following problem. I want to
Call -- Asterisk AGI Answer -- Create File - Copy File Asterisk
-- Play File -- Finish Call
For now we are using sshfs to map the directories. I now want to achieve
On Mon, Aug 2, 2010 at 5:37 AM, Tino t...@sparksupport.com wrote:
Hello,
I would like to know whether there is a way to associate a TV media server
with Asterisk. Is it possible to access TV Chanels in the Telephone Sets.
Anybody have any tips or documents related to this please let me
On Mon, Aug 2, 2010 at 9:56 AM, mattias m...@mjw.se wrote:
Is a mail server built in in asterisk now
Like in elastix
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On Mon, Aug 2, 2010 at 12:15 PM, mosbah abdelkader
mosbah.abdelka...@gmail.com wrote:
Thanks for your reply.
My configuration is correct. It works with ssh: many attacks have been
stopped. Also, the config has worked for asterisk one time: I have seen that
in the fail2ban.log file.
--
On Sun, Aug 1, 2010 at 7:12 PM, ast guy ast...@gmail.com wrote:
Hi,
I have made a fresh install of asterisk-1.6.2.10 and when I register my soft
phone it gives following error. Rest are default configurations.
32.454370 MY_IP - ASTERISK_SERVER_IP SIP Request: REGISTER
On Thu, Jul 29, 2010 at 4:05 PM, jwexler jwex...@mail.usa.com wrote:
On Thu, Jul 29, 2010 at 10:15 PM, Paul Belanger wrote:
MAC Address? Are you sure? Why would your ISP care about level 2? I
could understand IP address (level 3). If this is the case, you will
need to spoof your MAC.
On Wed, Jul 28, 2010 at 6:38 AM, Randy R randulo2...@gmail.com wrote:
Hi,
Since f2b is one of the topics du jour here, I was wondering if
someone would mind telling me what these pf stats mean:
Evaluations: 964303 Packets: 12176 Bytes: 648408 States: 0
Looks like pf examined nearly a
My guess is on spammers signing up the spamtraps for mailing lists ;)
On Wed, Jul 28, 2010 at 6:45 PM, Sam aster...@net153.net wrote:
Just a note, the asterisk mailing list server continually gets
blacklisted over at
http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering
On Wed, Jul 28, 2010 at 6:06 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Everyone,
This is probably more related to Linux than to Asterisk. Analogue channels on
a system were un-responsive on Monday morning. Apparently something happened
over the weekend and the router went off or it lost
On Tue, Jul 27, 2010 at 12:50 PM, Roderick A. Anderson
raand...@cyber-office.net wrote:
Anyone tried installing Asterisk in a AWS server?
\\||/
Rod
--
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On Tue, Jul 27, 2010 at 4:16 PM, Randy R randulo2...@gmail.com wrote:
Kyle Kienapfel wrote:
On Tue, Jul 27, 2010 at 12:50 PM, Roderick A. Anderson
raand...@cyber-office.net wrote:
Anyone tried installing Asterisk in a AWS server?
I'd think twice about trying this, taking into account
On Mon, Jul 26, 2010 at 6:14 PM, Adolphe Cher-aime achera...@gmail.com wrote:
To have your asterisk box reachable from internet you must configure static
nat on your router to get sip traffic to the public Ip redirected to your
internal ip. Make sure that sip and rtp traffic are not bloked by
On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam resea...@businesstz.com wrote:
I am having a problem understanding the way to retrieve some parameters to
asterisk via AGI or what ever method that fits. I have an executable
program that accept one parameter (CALLERID) and return customer status
from
On Sun, Jul 25, 2010 at 9:04 AM, Muro, Sam resea...@businesstz.com wrote:
Kyle Kienapfel wrote:
On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam resea...@businesstz.com
wrote:
I am having a problem understanding the way to retrieve some parameters
to
asterisk via AGI or what ever method that fits
What sort of options does audacity give you for wave files?
r...@missy:/usr/share/asterisk/sounds/en# file /tmp/test.wav
/tmp/test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM,
16 bit, mono 8000 Hz
Try to match that.
On Sat, Jul 17, 2010 at 3:52 PM, David Shauger
It looks like theres no much information out there about using realtime moh
Have you tried making an extension that goes to MusicOnHold(testmoh)
On Fri, Jul 16, 2010 at 8:21 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello list ?!
Is there anyone that can point me to the documentation
Use google or wikipedia if you don't know enough to ask a real question :)
On Sun, Jul 11, 2010 at 7:29 PM, mohamed daif mohamed.d...@gmail.com wrote:
Dear All.
I want to become a wholesale VoIP traffic Provider , and i don't have a
experience about the software used this career .
I ask
On Tue, Jul 6, 2010 at 11:08 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
- Original Message -
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to
On Wed, Jul 7, 2010 at 10:34 AM, Adil Zaaraoui adilzeaara...@yahoo.fr wrote:
Dear list.
Is it possible to use both IAX2 and SIP protocole during a dial?
Illustration:
I have peer A communicate with my Asterisk using IAX2 protocole.
I have peer B communicate with my Asterisk using SIP
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that
we are unable to URI dial our clients. We run a multi-tenant server
and have set sip.conf to forward calls to a
On Mon, Jul 5, 2010 at 5:05 AM, Randy R randulo2...@gmail.com wrote:
PS: http://www.ayera.com/teraterm/
I'm pretty sure there was a last update or patch or something because
Whats different about teraterm compared to putty? I know back in the
day I used to send files to my linux box with
On Sun, Jul 4, 2010 at 4:32 AM, amit salunkhe amitsalunkh...@gmail.com wrote:
Dear ALl
Can we use Asterisk for only for transcoding?. if yes how many concurent
call we can transcode with help of Astetrisk?
trans-coding is one of the bigger CPU hogs, amount of calls depends on
CPU and what
I use a release of putty called putty tray available at
http://haanstra.eu/putty/ for its URL clickability
This is what ubuntu does, for some reason not for screen sessions
export PROMPT_COMMAND='echo -ne \033]0;${us...@${hostname}:
${PWD/$HOME/~}\007'
One of the distros sets it so that it says
For those codecs an interfaced DSP might be the only option due to
lack of, or expensive software options.
I had an easier time looking into MELPe than I did with CVSD, so I
looked around just a little bit to satiate my curiosity
On Wed, Jun 16, 2010 at 9:23 AM, Jeff LaCoursiere j...@sunfone.com wrote:
On Wed, 16 Jun 2010, Randy R wrote:
On Wed, Jun 16, 2010 at 5:16 PM, Jeff LaCoursiere j...@sunfone.com
wrote:
pretty much giving up on Skype for Asterisk (and Skype for SIP) now
that I realize that they'll be
On Thu, Jul 1, 2010 at 3:50 PM, Kyle Kienapfel doctor.w...@gmail.com wrote:
On Wed, Jun 16, 2010 at 9:23 AM, Jeff LaCoursiere j...@sunfone.com wrote:
On Wed, 16 Jun 2010, Randy R wrote:
On Wed, Jun 16, 2010 at 5:16 PM, Jeff LaCoursiere j...@sunfone.com
wrote:
pretty much giving up on Skype
It did not look to me like he was suggesting you switch to JACK+SIP
for your application. The SIP suggestion is to isolate your problem.
How are you listening to the voicemail? Through jack or?
--
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http://en.wikipedia.org/wiki/G.729
Looks like theres A and B and no A/B so theres nothing to worry about
On Thu, Jun 3, 2010 at 9:09 AM, Alejandro Cabrera Obed
aco1...@gmail.com wrote:
Dear all, I've read that Asterisk supports only the G.729 A audio
codec. I have several Grandstream IP phones
== Registered translator 'amrtolin' from format unknown to slin, cost 4000
== Registered translator 'lintoamr' from format slin to unknown, cost 32002
Probably shouldn't be listing it as unknown
Have you tried using that AMR codec beyond commands in the asterisk cli?
Did the patch apply
On Sat, Mar 13, 2010 at 1:40 PM, Thomas Perron thomas.per...@gmail.com wrote:
DID number A.
I have a DID (a regular line from Verizon). number A.
Can I have A ported to my SIP provider?
Then, interface the A DID to my system so that I can build a solution.
I want to write an IVR for the A
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
Has example
exten = s,1,Answer
exten = s/9184238080,2,Set(CALLERID(name)=EVIL BASTARD)
exten = s,2,Set(CALLERID(name)=Good Person)
exten = s,3,Dial(SIP/goodperson)
for white list
exten =
Hi, I stumbled upon mentions of a SILK codec last night on skypes
skype for sip information page. I tried looking into it further and
found some blog and mailing list posts from 2009 but I can't find any
mentions of anything other than skype using the codec. Has the codec
not gotten anywhere so
I don't think this mailing list is intended for posts advertising
peoples services.
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To UNSUBSCRIBE or update options visit:
strip_ampersands(${EXTEN})?
On Sun, Feb 14, 2010 at 10:56 AM, C F shma...@gmail.com wrote:
On Sun, Feb 14, 2010 at 3:26 AM, Olle E. Johansson o...@edvina.net wrote:
14 feb 2010 kl. 03.25 skrev C F:
Excellent and very informative article, Thanks Olle.
You're welcome.
I ran thru lots of my
use two contexts, one for internal numbers, and one for outside, and include
the inside phones in the outside context.
On Fri, Feb 12, 2010 at 12:39 PM, cool dude cool_dudeof...@yahoo.co.inwrote:
i had configured asterisk with a minimum dial plan, made 10 extentions.
below is extensions and
this on list, thanks
On Fri, Feb 12, 2010 at 12:48 PM, Kyle Kienapfel doctor.w...@gmail.comwrote:
use two contexts, one for internal numbers, and one for outside, and
include the inside phones in the outside context.
On Fri, Feb 12, 2010 at 12:39 PM, cool dude cool_dudeof...@yahoo.co.inwrote
From a technical point UDP and TCP ports are separate, a server
listening for TCP requests on port 80 wont see any UDP traffic on that
port unless it explicitly opens a UDP socket. Tunneling in on UDP port
80 might be possible if the routing rules that are in place dont
specify to allow only TCP
check the output of running configure for any mentions of problems with libspeex
On Mon, Feb 8, 2010 at 8:09 AM, nedo nodo nedo.n...@gmail.com wrote:
Hi,
I would like to add support for speex codec in Asterisk.
In Ubuntu 9.10 the procedure is the following:
1) sudo apt-get install speex
You should take a look and see if any SIP packets are going out that
mention Connected Line 0317998955 as either something is or isn't
sent out from the asterisk server.
On Sat, Feb 6, 2010 at 4:30 AM, Magnus Benngård
magnu...@inputinterior.se wrote:
Gentlemen,
Did tryout CONNECTEDLINE
May as well check anything that you can. I usually start with
memtest86 when i'm curious about system stability.
On Fri, Feb 5, 2010 at 2:59 PM, Danny Dias ing.diasda...@gmail.com wrote:
Hello my friends,
My asterisk is going down randomly, following you will find some errors that
i could
Evenly distributed? like with conferences? or with Mixmonitor having
two sides to record?
On Thu, Feb 4, 2010 at 4:39 PM, Lyle Underwood lyleunderw...@gmail.com wrote:
If call recordings were stored in stereo and the callers were evenly
distributed along the stereo spectrum. BAM.
Just a cool
Any updates on this? It looks like I can't update CDR(userfield) from
inside such a macro and have it written to the cdr record.
[macro-pstn-trigger]
exten = s,1,noop()
;exten = s,n,DumpChan()
exten = s,n,verbose(${DIALEDPEERNUMBER})
exten = s,n,verbose(cdr userfield ${CDR(userfield)})
exten =
Where are these urls being input into asterisk?
On Tue, Feb 2, 2010 at 11:11 PM, Alex Balashov
abalas...@evaristesys.com wrote:
On 02/03/2010 02:03 AM, Olle E. Johansson wrote:
2 feb 2010 kl. 11.20 skrev BERGANZ Francois:
Hello all,
Does asterisk accept uri tel: instead of sip: ?
No,
You should phone up the emergency people on a non-emergency number and
ask them about that as well.
On Thu, Jan 28, 2010 at 10:58 AM, mir shahnawaz shahnawaz...@gmail.com wrote:
Thanks for your reply. Yes POTS lines are coming into the building but
I have multiple rooms. Suppose a person is
What happens when you dial with a handset? Is this delay caused by the
asterisk or is the telco doing it?
On Thu, Jan 28, 2010 at 2:57 PM, Danny Nicholas da...@debsinc.com wrote:
Greetings all,
This was most likely covered in one or more of the 15K
emails I tried to
Can you link the howto or other documentation you are following to set this up?
What version of asterisk?
Did you edit extconfig.conf?
Heres a howto for 1.4.x
http://hostseries.com/asterisk-realtime-installation-guide/
On Wed, Jan 27, 2010 at 8:39 AM, ahmed magdy amagdy.ibra...@gmail.com wrote:
You'd need RTP ports open for asterisk then.
Transfers and parking can be done at the SIP level, asterisk doesn't
have to be in the RTP path, as it can reinvite itself into the
callpath as necessary.
On Wed, Jan 27, 2010 at 5:23 AM, Vincent codecompl...@free.fr wrote:
Hello
I think I finally
If the computer is the same as the phone, one can't whine about
breaking one while talking on the other :)
On Wed, Jan 27, 2010 at 7:09 AM, Karl Fife karlf...@gmail.com wrote:
On Mon, Jan 25, 2010 at 12:07:55PM -0600, Karl Fife wrote:
From: cb c...@mythtech.net Sent: Sunday, January 24, 2010
The playback command is designed to work with multiple formats
If the channel in question is gsm it'll use a .gsm file before a .wav file
if the .wav file is in the directory, is it playable by asterisk?
(8000hz sample rate, etc etc)
On Tue, Jan 19, 2010 at 8:20 AM, Danny Nicholas
The 8 probably comes from the T1, does the telephone number end with an 8?
The playback of ss-noservice might be a fallback ensuring that
*something* happens when a call comes in
On Sun, Jan 10, 2010 at 1:31 PM, Edwin Quijada
listas_quij...@hotmail.com wrote:
Hi!
I have an T1 line for using
Going along the internet between us and canada doesn't add much
distance, but bouncing back and forth between east and west coast
does.
On Tue, Jan 5, 2010 at 11:25 AM, Max McGraw max.mcg...@gmail.com wrote:
hello,
we have been using a couple of US based
VoIP providers for outbound calls
On Tue, Jan 5, 2010 at 5:24 AM, Arun Sasidhar
arun.sasid...@cabotsolutions.com wrote:
Hi,
I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The phone
display is showing Unknown when there is an incoming
Having a trunk version available is a convention, but not a requirement of
using subversion. You'll probably want to check out a branch.
On Sat, Oct 24, 2009 at 7:57 AM, Olivier oza-4...@myamail.com wrote:
Hello,
I'm rather new to svn, so please, forgive me if this question sounds naive
but
Your question doesn't seem clear on which way you have set up already.
Phone - Asterisk - legacy PBX - one of the 23 extensions
Or the other way?
The capabilities of your specific existing PBX and asterisk need to be
matched up. With an FXS you could have the other end dial with DTMF a string
On Wed, Oct 21, 2009 at 10:15 PM, Olivier oza-4...@myamail.com wrote:
2009/10/21 Leif Madsen leif.mad...@asteriskdocs.org
Olivier wrote:
Hi,
Siemens Gigaset line of products include an integrated web browser with
which firmware download is possible.
The trouble is you need to
I thought google pulled uploading to that site after they bought youtube.
On Thu, Oct 22, 2009 at 4:05 PM, Ron Arts ron.a...@neonova.nl wrote:
http://video.google.com/
Free, no length limit, and they seem to have plenty of bandwidth...
Regards,
Ron Arts
NeoNova BV
John Todd schreef:
Your best option without a local asterisk server is to set up the remote
server to do reinvites when calls are going local-local
The calls will end up routed through your internet router, but not beyond
that.
Downside: might have to make each ip phone available via port forwards
If you're
Your best option without a local asterisk server is to set up the remote
server to do reinvites when calls are going local-local
The calls will end up routed through your internet router, but not beyond
that.
So by placing canreinvite=yes in sip.conf, the RTP-traffic would flow
between
It should be reproducible in some way, how was asterisk installed on the
server its having a problem? If its from source compare the
apps/app_voicemail.c from whats in production with whats getting compiled in
the lab.
when imap is used only one format is stored
you could specify just one format:
warning message about being sure to delete all messages not using that
format? I would think not but it's a dire enough message that I thought
I had better ask - John
On Wed, 2009-10-21 at 14:02 -0700, Kyle Kienapfel wrote:
It should be reproducible in some way, how was asterisk installed
apache is CaSeSeNsItVe where did you get the link with SLN capitalized?
http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-sln16-1.4.9.tar.gz
On Fri, Oct 2, 2009 at 11:22 AM, Mark Hulber asterisk.ad...@hulber.comwrote:
It looks like there's a problem with the
I checked the source for reading of configuration options but I didn't see
anything in vm_execmain()
This is the line of code that is bothering you
cmd = get_folder2(chan, vm-savefolder, 1);
On Mon, Sep 28, 2009 at 8:41 AM, Mike l...@virtutel.ca wrote:
I am looking to
It's been my experience that when asterisk does a dns lookup, for externhost
or to do a SIP register, it blocks the whole server. Not sure if 1.6 has
that problem or just 1.4 though as my internet has been stable while im
awake these days
On Sun, Aug 30, 2009 at 5:54 PM, Alex Samad
Disclaimer: I'm just a guy
Step one seems to be to delay account activation on your end. ;)
I do know that Les.net sits on payments from unverified paypal accounts for
14 days.
With all the fraud going around these days you might have to prove that the
buyer is the payer. I think you're limited
why is CROSS_ARCH=Linux? is this something the AVR32 distro is doing, or
something you did? it should be something line avr or avr32
On Thu, Jun 18, 2009 at 3:08 AM, Paulo Santos paulo.r.san...@sapo.ptwrote:
Greetings everyone,
I'm trying to compile asterisk for an AVR32 (Atmel NGW100).
For determining security risks, its specific to how your dialplan is set up.
If a person connects to your asterisk, what can they do? what happens? did
you set the incoming context to one with outgoing dialing rules?
Also for filtering calls, you'll probably want to either look at the
incoming sip
What are these dns queries for? I'd like to disable them but I cant
find any obvious reference to them in the asterisk source.
I'm running Asterisk 1.4.21.2
I call voicemail and immediately hang up:
I called from a sip client called line1, but I have no idea where
08c5b9e0 is coming from...
I can only suggest the most obvious cause without knowing how its
configured, sorry.
Take a look at the default context in sip.conf
for me:
[general]
context=default
my default context doesn't exist, so if a call comes in from an
unknown user, asterisk complains about not matching whatever
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