Hi guys,
Did you get some explanation? I'm suffering the exact issue. It could be I
need some additional dependencies than before?
Regards,
Elder
On Mon, Jul 25, 2011 at 7:59 AM, Soeren Malchow (MCon) <
soeren.malc...@mcon.net> wrote:
> Dear Shaun,
>
> First, thanks for you answer
>
> The inst
Could you please share a little sample showing how to get connected to AMI
with php?
Thanks a lot!
Elder
On Mon, Aug 1, 2011 at 3:40 PM, Kevin P. Fleming wrote:
> On 08/01/2011 03:35 PM, Paul Belanger wrote:
>
>> On 11-08-01 04:24 PM, Daniel - Asterisk wrote:
>>
>>
Hi guys, I hope you could help me.
I am trying to get connected through AMI but something is not working. Both
php code and manager.conf were working well in asterisk 1.4
1. Sometimes it gets connected and sometimes it doesn't:
== Connect attempt from '192.168.25.241' unable to authenticate
=
On the CLI write: sip show channels
If there are lots of bye channels you have the same problem than me.
I've tried waiting with the call generator -sipp- and channels
finished when there are a few. But they're not ending faster enough
when I send lots of concurrent calls.
Elder
2011/7/5, A E [G
ated between the
> endpoints.
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
> On Jul 4, 2011, at 3:29 PM, Daniel -
I'm trying to get working SIPp with media but something is wrong (it's
working well without media), please help:
This is the command I send at SIPp server:
./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err
This is the result I see:
Last Error: Aborting call on unexpect
al(SIP/intern,30)
same=>n,Hangup()
exten => 2006,1,Answer()
same=> n,WaitMusicOnHold(4)
same=> n,Hangup()
I'm using sipp.3.1.src.tar.gz and I have installed it this way:
..sip.svn# make pcapplay
Thanks in advance.
Elder
On Thu, May 12, 2011 at 2:51 PM, Steve Totaro
wrote:
>
) ?
in my opinion, more cores are better, because Asterisk ist multithreded and
each channel has a good chance to distribute to the cores.
is that right or what do you think?
Thanks for your answers.
Daniel
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Hello Everyone,
I wonder if someone could share a manual about using SIPp for Asterisk's
testing.
I'll be gratefull
Regards,
Elder Arohuanca
Lima - Peru
On Tue, Sep 30, 2008 at 12:59 PM, zac wolfe wrote:
> Sipp looks pretty good! I don't know how I missed this one. This would've
> saved me
Thank for the hint. I will have a look into it.
Daniel
-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von isr...@gmail.com
Gesendet: Freitag, 6. Mai 2011 15:22
An: Asterisk Users Mailing List - Non
ion to the predefined
address and port and send a simple TCP message (trigger) with caller and callee
ID and close the connection afterwards. Is this scenario possible without any
plugin like the Java API or similar?
Thanks,
Daniel
--
I found a solution that works fine for me
Set(var1=${SHELL(shellcommand)})
Bye Daniel
> Von: Daniel Knoll
> Datum: 16. April 2011 13:13:28 MESZ
> An: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Betreff: write system command output into a variable
>
"123 "
Can anyone help me ?
Thanx a lot for help
Daniel
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>> - upgrade policy - is it intended that someone who has Debian 6 with
>> the existing Asterisk 1.6 packages (from Debian's maintainer) can just
>> upgrade to the Digium package without moving or changing any config?
>
> There is nothing specific about the packages that is going to make this
>
> This effort is not intended to replace packaging of Asterisk in the
> official Debian or Ubuntu repositories. Our repositories are for
> providing access to major versions of Asterisk that are newer than what
> is included. We are exploring ways to work as closely as possible with
> the Debian
hy or Psi)
Google user frie...@gmail.com wants to make a voice call to
aster...@widgets.com - is it possible?
For this scenario, is gtalk.conf needed at all? Is gtalk.conf needed
for any Jabber server, such as the ejabbard instance described above?
Regards,
D
ple told me
they managed to set this up as a SIP/PSTN gateway.
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gt; directly). Thus, a 10/100Mbps Ethernet card was installed to provide
> the second port needed.
You can free the PCI slot if you use VLANs on the internal interface
to seperate the internal and external traffic. This requires a switch
with vlan support, shouldn't could much more than
igium hardware and you still need more than 1
if you want to avoid this being a single point of failure. But they can
be more flexible in some setups (multiple active asterisk machines
connecting simulataniously)
--
o filter, not
very elegant but works over any transport. I use this to do multitenant
billing on the remote server in places where I only want 1 IAX trunk.
Whether this is effective depends on your control of the local server.
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Finally I could get it to work by running a shell script which parsed
results from 'queue show' CLI command in dearch of 'Not in Use' members. It
was done with an AGI.
Regards,
Daniel
On Tue, Feb 8, 2011 at 11:52 AM, Carlos Chavez wrote:
> On Mon, 2011-02-07 at 10:44 -
or
your favorite scripting language.
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http://
s.digium.com] *On Behalf Of *Daniel - Asterisk
> *Sent:* Monday, February 07, 2011 9:38 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] About maxlen parameter in queues
>
>
>
> Dear list,
>
>
>
> I want to avoid sending ca
Dear list,
I want to avoid sending calls to a queue when it is full. From the fact that
'maxlen' must be at least 1 (I wish it could be zero but it isn't) I'd like
to know if there's a way to do it. Setting the Queue() timeout to a little
value is not the most suitable option.
I'm using asterisk 1
r the AMI way you could implement an event listener for PeerStatus
changes:
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Events
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found
by matching channel==dstchannel for all channels.
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e.
https://issues.asterisk.org/view.php?id=13243 explains settings codecs
(and its difficulties, but since you have full control there shouldn't
be a problem). Use GROUP and GROUP_COUNT to find out how many channels
are are active, use this to decide whether to use alaw or g726.
--
anging the the priority dynamically depending on
context and waittime should avoid the wrong reported holdtime.
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anything at all.
Daniel Tryba
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http://www.asterisk.org/hello
asterisk-use
r that understands asx like in the URL I
posted. But if you take a look at the content of the .asx you'll see
that it contains links to mp3 streams. You could pick one of them
manually, but expect the URLs to change in the fut
On Sat, Dec 25, 2010 at 04:04:59PM -0700, Dave George wrote:
> My server is being attached all day and fail2ban is not stopping the
> attack. I updated stamstamp to match fail2ban requirements.
How about posting your fail2ban config?
--
Daniel
le people using the
> dial plan but that is not available when you are listening to
> voicemail.
My guess the easiest hack is to create a local alias (/etc/aliases) that
will relay the mail to multiple users.
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our config (custum should be custom).
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h
d any record. have i placed my ResetCDR(w) correctly?
I had the same problem last week, my fix was adding:
[columns]
alias uniqueid => uniqueid
to cdr_mysql.conf
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Daniel Tryba
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nt to VoiceMail will skip vm-intro. So you only need to figure out
is unavail.gsm exists from the dialplan to add 's' to the arguments.
Implementing this in an AGI script should be trivial.
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ye
opener.
What I'm looking for is a extension that handles:
^\*\w+\*\d+$
I guess I'll have to catch _*. and manually check if it matches above
regexp.
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ccountcode=${CUT(EXTEN,*,2)}) [pbx_config]
3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config]
4. Set(CDR(accountcode)=${accountcode}) [pbx_config]
7. ResetCDR() [pbx_config]
8.
Hey Guys,
In which Version of Asterisk is "EventFilter:" in manager.conf working?
Higher than 1.6.2.10 or from the 1.8.0 Version?
Thank for your answer
Daniel
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Hi Godson Gera,
thank you for your answer.
if i understand correctly, the EventMask filter all until i define all event
that i need.
This is not really helpful, because i must define the categories that i need.
mhh
is there another Solution for my Problem?
Thanks a lot
Daniel
Am 09.12.2010
er .*@.*
>
> If anyone who is more familiar with the attacks or how to generate
> these messages would give me some assistance, or chime in on the
> sshguard-users list, that'd be most appreciated.
You could use SIPVicious to run attacks on your own servers:
http:/
> _X.,1,Set(GRUPPO=${EXTEN:-2:1})
exten => _X.,2,Set(ALLARME=${EXTEN:1:1})
exten => _X.,3,AGI(checkgroup.php|${GRUPPO})
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Hey Guys,
for debugging i need to read the Events from AMI. But i have a lot of unwanted
"RTCPSent" Events.
How can i filter this Events in Asterisk 1.6.2.x Version of Asterisk?
Thanks a lot for your answ
(e.g. Positron sells PCI cards that should do the trick if you want
something "internal")).
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tain the ipadress(es) of your internet connection(s), by only
explicitly listing internal ipadresses and hostnames. e.g.:
domain=10.2.3.4
domain=sip.example.com
The standard scanners will get a "Not a local domain" error, since they
only try the external ipadress to connect (f
abled the call will be answered in my SN4554.
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a dialplan where calls to internal
numbers check whether ${DB(CFIM/${EXTEN})} is empty (and do nothing) or
set (and dial that number instead).
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otification get send back to the sender).
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gister from Twinkle softphone, I get the
following:
-----
lun 13:41:56
Daniel, registration failed: 503 Service Unavailable
-----
Thanks for your reply.
Regards,
Daniel
--
erver by its IP, so I do
not think there is a problem on that side.
To enable debug I should use 'sip set debug'? from the Asterisk CLI? I
do not see any record in the CLI after running this command. However,
from Twinkle, for example, I see the following:
--
Is there a parameter in the Asterisk configuration
where also I have to "force" the use of an internal DNS server?
Thanks for your reply.
Regards,
Daniel
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, can it be? In this
case, is there a possible workaround?
Thanks in advance for your reply.
Regards,
Daniel
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specific (Debian stable with zaptel),
switching to MG2 made my problems dissapear but overall voice quality is
lower IMHO. You could try disabling ec all together and check if
clipping still occurs. But it does sound like an operator p
as registrar, location server,
within-dialog request routing
Hope it is useful for some people within this community.
Next step, naturally, is to upgrade the tutorial for latest Asterisk,
1.8.0, just needs some time to get familiar with it.
Cheers,
Daniel
--
Daniel-Constantin Mierla
Kamailio
your setup for echo cancelling?
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UDP traffic (can't remember the specifics).
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http:/
lding this is a the combination
of 2 examples (CIRC10 and CIRC11 from
http://ardx.org/src//guide/2/ARDX-EG-OOML-DD.pdf).
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New to Asteri
AT connected
to an * (behind NAT) without problems and without any portforwards at the
Grandstream side. nat=yes and canreinvite=no as always do the trick for
me.
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dmin to setup NAT routing without rewriting the external
adress/port.
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0$ router.
But what ports did you open? Only sip or also the RTP ports?
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Hey,
i forgot to ask, how can i get the user number from a caller he is in a
conference, i don't find a variable to us this for the current channel.
Only the command "meetme list " shows the usernumber, but i can't use
this out
Hi @ all,
what is the best way to to use features like MeetmeCount without leaving the
conference.
I use Meetme(,X) and MEETME_EXIT_CONTEXT=context, but the problem is that the
caller leave the Conference :(
Is it possible to press a key, without this obstacle?
Thanx for your answers
Daniel
the network statitics.
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atistics per call (can be retrieved from the RTCP reports in
asterisk). These probes and the analyzer software aren't bug free and
perfect but give a good indication of all historic calls. Once a problem
is spotted we move to test calls to trace the pro
I'd say you should try the
difference values for nat to see if one works with the NAT gateway or
use STUN like suggested elsewhere.
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On Wed, Oct 06, 2010 at 01:56:55PM +0200, Daniel Tryba wrote:
> Issuing the AMI Status command results in a list of active channels. But
> how to figure out which channels are related before the call is
> answered?
Anybody?
My workaround for this problem is setting a persistent variab
On Thu, Oct 07, 2010 at 02:24:59PM +0200, Daniel Tryba wrote:
> > It's the same account, the same password, but other agent.
> >
> > Can anyone help me with this please ?! I see no difference but there
> > must be !!
>
> The difference is the SNOM is using
Is nat for
this client set to 'yes' or something else?
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On Wed, Oct 06, 2010 at 01:56:55PM +0200, Daniel Tryba wrote:
> Issuing the AMI Status command results in a list of active channels. But
> how to figure out which channels are related before the call is
> answered?
CoreShowChannels gives a little bit of extra data in the originato
; 'trunk',
'CallerIDName' => '0031234567890',
'Accountcode' => '',
'ChannelState' => '4',
'ChannelStateDesc' => 'Ring',
'Context' => 'macro-dial-one',
'Ex
Has anyone a solution for me
- with "Meetme(,Ms)"asterisk plays "conf-invalid" if a room not exist
- with "Meetme(123,Ms)" asterisk plays not "conf-invalid" if the room not exist
and asterisk hangup
I am
Hello,
is it possible to check more than one condition for GOTOIF in the dialplan?
Or is the normal way to cascade the diaplan each GOTOIF?
The Background is that I would like to check more than 2 values from a
Variable, and then route the call based on the value.
Thanks for your help.
Daniel
session between the SIP device and Asterisk.
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e option (NAT Keep Alive
Enable).
But even with both these setting enabled NAT gateways sometimes seem to
lose track of SIP sessions (I have more trouble with Cisco devices than
Linux routers), setting the UDP session timeout to 10m seems to help
(default is something like 3m).
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aw and #3
> is slin; Need it the other way so I can do DAHDI--> IAX testing.
exten => 1234,1,Set(_SIP_CODEC=alaw)
exten => 1234,n,Goto(0234,1)
exten => 2234,1,Set(_SIP_CODEC=slin)
exten => 2234,n,Goto(0234,1)
Should do
ial" 2.1.0.4
in any debian version. If you are running stable you should either use
the backports version: http://packages.debian.org/lenny-backports/dahdi
Or make your own package from testing/unstable sources.
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On Fri, Sep 24, 2010 at 02:30:12PM -0400, Paul Belanger wrote:
> > Am I missing something?
> >
> DEBUG_THREADS
Thanks, I guess I should have RTFM :)
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DULES DEBUG_CHANNEL_LOCKS
MALLOC_DEBUG
$ make && make install
$ asterisk && asterisk -rx "core show locks"
No such command 'core show locks' (type 'core show help core' for other
possible comma
Last week I had a couple of outages one machine, the problem was that
Asterisk suddly stopped responding to UDP SIP requests. tcpdump show
requests arriving on the machine, sip debug log in asterisk doesn't show
anything for the UDP peers, TCP functions just fine.
In all 3 cases the log is somethi
Hi Paul,
i set Answer() .. just Cut the first, my fault.
is that the normal case, to treat errors like wrong conference Room?
Daniel
Am 07.09.2010 um 15:01 schrieb Paul Belanger:
> On Tue, Sep 7, 2010 at 3:11 AM, Daniel Knoll wrote:
>> Hi Kai-Uwe,
>> thank you for your answer.
awn extension (provider, 22, 8) exited non-zero on 'SIP/100-3b5c'
Any Ideas?
Thanx,
Daniel
Am 06.09.2010 um 23:54 schrieb Kai-Uwe Jensen:
> I use "MeetMe(,Ms)" in the Dialplan and if a Conference Room does't exist
> Asterisk play (conf-invalid.slin)
>
ay (conf-invalid.slin) and Asterisk Hangup the
Call.
there is a solution for the kind my problem?
Thanx and bye
Daniel
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Hello Everybody,
does anyone knows an opensource stresstest client for the IAX protocol, like
sipp?
Thanx for your answer.
Daniel
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New to Asterisk
omputer with one pci port (its pretty tight to the PSU wall but i
hope it will fit.. any interference because of that?)
thanks!
> On Mon, 2 Aug 2010, Daniel Petre wrote:
>
>> hello,
>> i just subscribed to this list, i discovered asterisk and i would
>> like to try it at h
Hi Group,
short question. is it possible to use
#include "asterisk/alaw.h" instead of #include "asterisk/ulaw.h"
in app_meetme.c or is ulaw required in meetme?
thanx for the answer.
Daniel
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hello,
i just subscribed to this list, i discovered asterisk and i would like
to try it at home on my personal pc.
the computer is a p4 at 3 ghz with 2 gb ram and 80 gb hdd, a 1 Mbit
guarranted connection and runs a gentoo linux.
i search about digium products but i can't find them in my area o
can i change the Format for Meetme to alaw which is the NativeFormat.
Thanks for your help.
Daniel
PS: i unload the format_sln16.so and format_sln.so modules.
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6,n,MeetMe(${CALLERID(num)},q)
Maybe it is helpful for all others.
Daniel
Am 12.07.2010 um 18:58 schrieb Daniel Knoll:
> Hi all,
> is it possible to send a Variable to another System via IAX Protocoll by
> using AMI / Orginate
> Like this:
>
> Action: Originate
> C
t; ?
In my Test it fails to transfer the Variable :(
What I doing wrong?
Thanx for your help
Daniel
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Hello guys,
I have this problem when a call is received in my PBX:
(Caller) --> (Redirecting Service) --> (E1 PRI) --> (Asterisk PBX) -->
(Internal Phone)
Reception works fine, but when conversation finishes and the agent at
internal phone hangs up, the call at caller's side is still alive for
m
ss)=music)
exten => ,n,Set(CHANNEL(language)=de)
exten => ,n,MeetMe(,Msp)
exten => ,n,Hangup()
thanx for help.
Daniel
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Hi Paul,
Yes, i can use iax2, but this is rather a redirect to another server as
connecting 2 confernce channels from 2 different server.
Can i join 2 dahdi (meetme) channels from different servers?
Regards Daniel
--Originalnachricht--
Von: Paul Belanger
Absender:asterisk-users-boun
Is it possible to join 2 meetme conferences (each on different server)
together, that if i load balance the callers, they can "see" altogether
something like a inter system communikation ?
Thanx for your he
ok, thanx for your answer.
Daniel
Am 20.06.2010 um 19:17 schrieb Tilghman Lesher:
> On Saturday 19 June 2010 10:47:07 Daniel Knoll wrote:
>> Hello Group,
>> what does the Compiler Option mean "LOTS_OF_SPANS" ?
>> The description is: "More than 32 DAHDI sp
store the SessionID in a
distributed filesystem or mysql database)
Or what is the best way, to load balance Meetme conferences?
I read abut openSIPS, thats sounds nice, but additionally i would like a kind
of asterisk channel interconnect for better load balance.
Thanx for your help.
Daniel
Hello Group,
what does the Compiler Option mean "LOTS_OF_SPANS" ?
The description is: "More than 32 DAHDI spans"
Does this mean, more than 32 DAHDI Channels ?
Thanx for help.
Daniel
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age mean?
Thanx for answers
Daniel
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Hi Michael,
Can you show us the output from:
"moh show classes" and "moh show files" Command
Or try it to set a new exten after setting the language with:
exten => 12345,n,Set(CHANNEL(musicclass)=personalised)
Daniel
Am 13.06.2010 um 12:35 schrieb Mickael Monsieur:
it's not so easy, because i use a mysql database (realtime) to write the room
number from a webapp into the table.
also i extend the meetme table for my web application :-/
any other things at least to show more logs from meetme or dahdi ?
Daniel
Am 12.06.2010 um 17:39 schrieb Thomas P
can anyone help me and maybe someone has also the problem as i and have an
solution.
I use:
asterisk-1.6.2.7
dahdi-linux-complete-2.3.0+2.3.0
asterisk-addons-1.6.2.1
Thanx a lot for any answers that helps me.
Daniel
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