Dear All
A consultation, i have a server in production with an asterisk digium te412p
card, the server is a slackware 12, whith asterisk compiled by hand.
under this scheme i have 2 e1 isdn over whith i receive calls that are
served by several agents queues. the server is already working at
Hello all,
I need to install asterisk for 900 sip users with 2 PRI ports.
It is posible to handle this number of calls/extensions with only one
asterisk machine?
Which is the best way to install that? two asterisk with openser. One
asterisk with openser .
Is it necesary run a SER server on
Hi Kate,
have you tried the busydetect parameter in zapata.conf?
Take a look here for other useful parameters:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
Giorgio.
Lists wrote:
Hi all,
When I do a test call into the box (which is running latest version of
Trixbox) it all
On Tue, Jul 08, 2008 at 09:34:44PM -0700, Trevor Peirce wrote:
Steve Totaro wrote:
For security, how about an authentication retry setting in the sip
configuration? After X amounts of failed auth or registration
attempts, block IP for Y amount of time. It would seem fairly easy to
do
With an ISDN10/20/30/etc, I would just put all the lines into an
'incoming' context - and make sure that incoming context doesn't have
any includes (unless you really need them...)
Can someone please have a look at below to see if this would be the best
and secure practice of using context in
How many calls do you expect to be going at one time?
Do you have any sip trunks for the users to call out on? Unless this ratio
really works for you I'm not sure a 15 to 1 ratio works for most people.
I wouldn't just depend on a single server for this purpose.
I'll leave it to the cluster guys to
Does anyone have experience with setting distinctive ring in SIP in such
a way?
I have done this for internal call with
;;;grandstream;;;
exten = _12X,1,Set(_ALERT_INFO=http://127.0.0.1\;info=internal)
exten = _12X,2,Dial(SIP/${EXTEN},30,tTr)
http://www.grandstream.com/asteriskfaqs.html
Hi all,
when enabling blind and attended transfers in features.conf, these only seem
to work when I enable voicemail for a particular user. How can this be? Can I
have transferrring without voicemail?
Using Asterisk 1.4 by the way.
Thank you!
Bart
On Tuesday 08 July 2008, Florian Hackenberger wrote:
On Tuesday 08 July 2008, Matt Riddell wrote:
Maybe the feature digit timeout?
The problem turned out to be related to several bugs in iaxclient with
alsa support. The softphone was unable to open the audio device and
that seemed to cause
Maybe 400 calls at one time. By the momento there aren`t voip trunks
maybe in the future.
About cluster, Which cluster solution will could be good option?
Which solution could I use to do load balancing between two asterisk machines?
Thanks again.
Voipcrazy
2008/7/9 Tom Moore [EMAIL
Hi all,
I want to write a script to test my asterisk server at predefined intervals.
The script will make a call to asterisk server. For that the extension is
defined which will answer the call wait for a second and then hangup. If
asterisk server receives the call successfully then it will
Hi guys,
I've been looking for a table layout that I should be using for cdr logging
to a Mysql database.
Everything I find seems to be a little different.
What should I be starting with before I start adding custom fields in the
future?
Also note I want to import some Master.csv files in to this
On Wed, 2008-07-09 at 10:17 +0200, voip crazy wrote:
Maybe 400 calls at one time. By the momento there aren`t voip trunks
maybe in the future.
[snip]
I need to install asterisk for 900 sip users with 2 PRI ports.
It is posible to handle this number of calls/extensions with only one
Hi!
I want to test Asterisk--Siemens HiCom integration using Q.SIG instead
of ISDN. I did not find any documentation about Asterisk und Q.SIG.
Thus, I wonder is it sufficient to set switchtype from euroisdn to
qsig or are there any other things which I have to take care of?
Thanks
Klaus
voip crazy wrote:
Hello all,
I need to install asterisk for 900 sip users with 2 PRI ports.
Is this correct? 60 channels (assuming an E1 connection, not a T1)
between 900 extensions means only 1 in 15 people can be on the phone at
once - which is a pretty low ratio.
If this is indeed
Hi All,
would Asterisk 'transcode' H.245 alphanumeric DTMFs
to an H.245 signal / rfc2833 H.323 device over G.729 codec ?
Thanks for supporting,
.TF
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Hy, i have a big problem with asterisk instance. Sometimes my instace fall
down on zerro and i must manually enter asterisk in console. Can someone
eyplain me how to make cron job and make conditions if asterisk fails?
Thx
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On Mon, 07 Jul 2008, Matt Riddell wrote:
Example :
exten = s,5,ChanIsAvail(SIP/604,s)
exten = s,6,Dial(SIP/604,15,wotr)
exten = s,106,NoOp(Matthieu)
exten = s,n,ChanIsAvail(SIP/605,s)
Won't work because Dial exit to 7, and line 7 don't exist
but
exten =
Hi,
I'm having a problem to receive inbound call from my sip provider. I used to
be OK, I may I have change something (for example I switched from asterisk
1.4.20.1 to 1.4.21.1). Could that be a bug? (I doubt that and I guess it a
configuration problem on my side!)
I have basically a sip
Hi,
How do i send proper message when hanging up?
[from-trunk]
exten = _1234,1,Dial(SIP/${EXTEN}|30|t)
exten = _1234,n,Hangup
With that, the other end receives a call reject if i don't answer the
phone, but the telco said they need something like No Answer instead
of Call Reject.
Is
Hello all!
I'm having problem with the calls that come through my asterisk box
and back out to our legacy pbx, it seems to be that even if the call
is ringing and not picked up yet, zap reports the line as answered,
why is it doing that?
Also it wont stay connected for the correct amount of rings,
Are asterisk and the phone on the same lan? I see you have nat=no. Do
you see the phone adapter registered?
Emmanuel Favre-Nicolin wrote:
Hi,
I'm having a problem to receive inbound call from my sip provider. I used to
be OK, I may I have change something (for example I switched from
Conrad Wood wrote:
Unless I am mistaken and there *is* some way to run 400 simultaneous
calls over 2 PRIs...
Traditionally, there hasn't been. But now that they've got that Large
Hadron Collider going... :-)
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel:
On Wed, 2008-07-09 at 13:42 +0200, Jerome Poggi wrote:
I use them before some patch. But this example work :
exten = s,5,ChanIsAvail(SIP/604,s)
exten = s,6,Dial(SIP/604,15,wotr)
exten = s,7,NoOp(Nopnopnopnopnop)
exten = s,10,NoOp(Matthieu)
and this not :
exten =
Hello,
Asterisk version 1.4.21.1
Can anybody tell me what I am doing wrong or why the Read application
does not accept the # key as input? My read statement:
exten = s,n,Read(uchoice|thankyouforcalling|3||1|1);
In the prompt thankyouforcalling it says press pound for a company
directory along
Hello all!
Hello!
I'm having problem with the calls that come through my asterisk box
and back out to our legacy pbx, it seems to be that even if the call
is ringing and not picked up yet, zap reports the line as answered,
why is it doing that?
could be that the PBX *answers* the line
They are on the same lan
the adapter is registered
sip show peers
Name/username HostDyn Nat ACL Port Status
sippyskypeuser/sippyskype 192.168.2.765070 OK (1 ms)
1000/1000 192.168.2.76 D 5061 OK (1 ms)
Bart Coninckx wrote:
Hi all,
when enabling blind and attended transfers in features.conf, these only seem
to work when I enable voicemail for a particular user. How can this be? Can I
have transferrring without voicemail?
Using Asterisk 1.4 by the way.
Thank you!
Bart
I think
On Wed, 2008-07-09 at 10:08 -0400, John Millican wrote:
Is it because Read exits with a # terminated string
Absolutely. The Read() application stops reading digits when the user
enters a #.
--
Jared Smith
Training Manager
Digium, Inc.
___
--
Tzafrir Cohen wrote:
On Tue, Jul 08, 2008 at 09:34:44PM -0700, Trevor Peirce wrote:
I was recently introduced to fail2ban. It's a nice tool that will watch
log files and when it notices too many failed authentication attempts
(SSH, FTP, Password protected web sites, asterisk) it will run
Very interesting article. I guess we won't know much more for another few weeks:
http://www.breitbart.com/article.php?id=080709124916.zxdxcmkxshow_article=1
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On Wed, Jul 9, 2008 at 10:50 AM, C F [EMAIL PROTECTED] wrote:
Very interesting article. I guess we won't know much more for another few
weeks:
http://www.breitbart.com/article.php?id=080709124916.zxdxcmkxshow_article=1
I thought this was common knowledge. I remember hearing about the flaw
I don't see anything obvious right away other than have you confirmed
that the phone is actually working? Can you get it to ring? With my
Sipura adapters that use Linksys software I can view the call status in
the Info section which if you have that panel might tell you if the
adapter thinks
Hi to all
is it possibile (via AMI or dialplan) to disable the DTMF tone on a
particular channel?
Thanks in advance
--
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
C F wrote:
Very interesting article. I guess we won't know much more for another few
weeks:
http://www.breitbart.com/article.php?id=080709124916.zxdxcmkxshow_article=1
Interesting! I just went there and the Check your DNS link failed.
Anyone else?
Rod
--
Hi all,
when enabling blind and attended transfers in features.conf, these only seem
to work when I enable voicemail for a particular user. How can this be? Can
I
have transferrring without voicemail?
Using Asterisk 1.4 by the way.
Thank you!
Bart
I think some clarification
Hi,
has anyone worked with nxtvox(www.nxtvox.com) fxo cards? What is their
quality?
James
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Register Now:
Snip
On Wed, Jul 9, 2008 at 10:50 AM, C F [EMAIL PROTECTED] wrote:
Very interesting article. I guess we won't know much more for another
few weeks:
http://www.breitbart.com/article.php?id=080709124916.zxdxcmkxshow_artic
le=1
I thought this was common knowledge. I remember hearing about the
Nhadie wrote:
Hi,
How do i send proper message when hanging up?
[from-trunk]
exten = _1234,1,Dial(SIP/${EXTEN}|30|t)
exten = _1234,n,Hangup
Ask the telco what cause code they expect:
asterisk*CLI core show application hangup
-= Info about application 'Hangup' =-
Well, i tried to see a noop with:
exten = 205,1,Dial(Zap/g1/205)
exten = 205,2,NoOp(After Dial ${DIALSTATUS})
exten = 205,102,VoiceMail
exten = 205,103,NoOp(After VM ${DIALSTATUS})
to see i got any dial status changes, i got nothing.
Let me explain my path and mabye it might help.
A User calls
On Wednesday 09 July 2008 09:03:07 Jared Smith wrote:
On Wed, 2008-07-09 at 13:42 +0200, Jerome Poggi wrote:
I use them before some patch. But this example work :
exten = s,5,ChanIsAvail(SIP/604,s)
exten = s,6,Dial(SIP/604,15,wotr)
exten = s,7,NoOp(Nopnopnopnopnop)
exten =
On Wednesday 09 July 2008 08:24:25 Kevin Leinenweaver wrote:
Hello all!
I'm having problem with the calls that come through my asterisk box
and back out to our legacy pbx, it seems to be that even if the call
is ringing and not picked up yet, zap reports the line as answered,
why is it doing
On Wednesday 09 July 2008 09:08:50 John Millican wrote:
Can anybody tell me what I am doing wrong or why the Read application
does not accept the # key as input? My read statement:
exten = s,n,Read(uchoice|thankyouforcalling|3||1|1);
In the prompt thankyouforcalling it says press pound for a
I don't think that this is the exploit that they are talking about.
What you say is too simple and requires too much to achieve (do it the
right time when a request is asked and quicker than the intended DNS
server).
On Wed, Jul 9, 2008 at 12:01 PM, Alexander Lopez [EMAIL PROTECTED] wrote:
Snip
On second note after reading CERT it looks like thats exactly what it
is. Another case where the media is over dramatizing something.
On Wed, Jul 9, 2008 at 1:17 PM, C F [EMAIL PROTECTED] wrote:
I don't think that this is the exploit that they are talking about.
What you say is too simple and
Hi additional question on music on hold.
scenario: both using x-lite
two extensions 101 and 102
101 calls 102
102 click onhold button
101 hears music (the music is what 101 uploaded)
102 offhold call
101 press onhold button
102 hears music (the music is what 102 uploaded)
call put offhold
call
Shouldnt it be able to detect the ringing ringer and a busy tone and
differentiate?
i'd think this would be a feature that would be implemented.
but atleast with the option g i saw that i says it was answered when
the line first picked up.
On Wed, Jul 9, 2008 at 9:59 AM, Tilghman Lesher
[EMAIL
i just discovered something that would work.
exten = 205,1,Dial(Zap/g1/205,,gr)
the g and r option make it work the way it should. i dont quite know
why r would help, but it did!
On Wed, Jul 9, 2008 at 10:22 AM, Kevin Leinenweaver
[EMAIL PROTECTED] wrote:
Shouldnt it be able to detect the
2008-07-08 - app_swift v1.6.2 released for Asterisk 1.6.x code-base
---
Added support for handling multiple dtmf input
Added support for input timeout and max input digits (similar to
AGI's get_data)
Ignores DTMF if no timeout and max digits
There is no performance impact if you use AGI or DeadAGI. The only
difference is, if you use AGI it will not continue executing the dialplan if
the calling party hangsup the call. DeadAgi, will continue executing the
dialplan and its upto the applications responsibility to hangup the channel.
So,
Does anyone have any experience getting inbound ANI information from a
CAMA/MF/EM Wink trunk on Asterisk?
Is this only do-able with a PRI interface? Any information would be
helpful.
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Hi,
Has anyone managed to get 2 AVM ISDN Fritzcard's working in with
a 2.6 kernel system?
Yes, with Suse 10.2/10.3 and chan_misdn.
are there any cheap ( 100 EUR) PCI-express ISDN-cards out that are
supported by Asterisk 1.4.x?
I would like to replace my SFF Pentium III with two PCI
I have a Sangoma A200DX, and am trying to bridge an FXO channel with FXS for
modem connectivity.
I have Zap/8 as a Fax Machine
Zap/5 is my outside line. When a call rings in on Zap/5 it immediately calls
Zap/8 and bridges the channels. I see it doing a native bridge on the two. I
have echo
I am seeing the following seg fault when using a SIP connection
to Console/Dsp. It takes quite a long time to happen but it eventually
happens.
nothing else is on this box. just alsa and asterisk running sip and
console/dsp.
What should I do now?
Jerry
Program received signal SIGSEGV,
On Wed, Jul 9, 2008 at 3:28 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
I have a Sangoma A200DX, and am trying to bridge an FXO channel with FXS
for modem connectivity.
I have Zap/8 as a Fax Machine
Zap/5 is my outside line. When a call rings in on Zap/5 it immediately
calls Zap/8 and
I set it up in general because my voice lines(ports 1-4) had very low volume,
and callers complained about outgoing as well, upping both to two seemed to
resolve them.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, July 09, 2008 2:39 PM
To: Asterisk
Then see if you can set the speed of your fax to something very low and work
up from there until you get to the fastest reliable speed.
On Wed, Jul 9, 2008 at 3:44 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
I set it up in general because my voice lines(ports 1-4) had very low
volume, and
is it only cell phone calls that don't work? or is it any external call
coming in over your lines?
What type of inbound lines do you have? I;m guessing analog lines... if
thats the case what type of signalling are you using?
if its only cell calls and not all external calls then I have no idea
Hi Matt,
It is only cell phone calls. We have two POTS lines coming in. I am
using TDM400 digium card.
This is my zapta.conf. Looks like i'm using signalling fxs_ks is this
right?
Thanks for your help.
Kate
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
2008-07-09 - app_swift v1.2.2 released for Asterisk 1.2.x code-base
---
Added support for handling multiple dtmf input
Added support for input timeout and max input digits (similar to
AGI's get_data)
Ignores DTMF if no timeout and max digits
Hi Giorgio,
Thanks for the suggestion. It works a treat.
Kate
Giorgio Incantalupo wrote:
Hi Kate,
have you tried the busydetect parameter in zapata.conf?
Take a look here for other useful parameters:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
Giorgio.
Lists wrote:
Hi
On Wednesday 09 July 2008 14:06:56 Thameem Ansari wrote:
There is no performance impact if you use AGI or DeadAGI.
There is a performance impact, in terms of the time it takes for
the process to start up. It may be measured in fractions of a second,
but there certainly is a performance penalty.
It looks like it's 19:
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
Nhadie wrote:
Hi,
How do i send proper message when hanging up?
[from-trunk]
exten = _1234,1,Dial(SIP/${EXTEN}|30|t)
exten = _1234,n,Hangup
With that, the other end receives a call reject if
On Wed, Jul 9, 2008 at 5:00 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 09 July 2008 14:06:56 Thameem Ansari wrote:
There is no performance impact if you use AGI or DeadAGI.
There is a performance impact, in terms of the time it takes for
the process to start up. It may be
I used to run into this when I first started palying around with some T1
cards in asterisk. I fixed it with the busydetect and busycount options as
already mentioned.
Try adding:
busydetect=yes
busycount=5
to your zapata.conf for the group thats having trouble. When the cellphone
hangs up
I'm trying to build a simple accept/reject screening app for inbound calls that
* forwards to my cell phone. Basically I want * to announce the caller ID and
then let me press 1 to accept the call or 2 to reject the call and send the
outside party to voicemail.
I've been messing around with
Yes this solution worked well.
Thanks
[EMAIL PROTECTED] wrote:
I used to run into this when I first started palying around with some T1
cards in asterisk. I fixed it with the busydetect and busycount options as
already mentioned.
Try adding:
busydetect=yes
busycount=5
to your
Jerry Geis wrote:
I am seeing the following seg fault when using a SIP connection
to Console/Dsp. It takes quite a long time to happen but it eventually
happens.
nothing else is on this box. just alsa and asterisk running sip and
console/dsp.
What should I do now?
Jerry
Program
Hi all,
I am wanting to change the sound files from the standard ones to a New
Zealand voice pack.
I have copied the files into the /var/lib/asterisk/sounds directory and
chowned them to asterisk:asterisk and chmod 420 to match the existing
files but the system is still using the original
On Wednesday 09 July 2008 16:50:09 Steve Totaro wrote:
On Wed, Jul 9, 2008 at 5:00 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 09 July 2008 14:06:56 Thameem Ansari wrote:
There is no performance impact if you use AGI or DeadAGI.
There is a performance impact, in terms of
Hi,
we have a strong echo on an Astribank running on
Asterisk 1.2.29-BRIstuffed-0.3.0-PRE-1y-s
The echo is on the opposite site of the analogue phones
connected to the Astribank.
I tried to use fxotune. It is a production system, so I cannot
shutdown Asterisk for long, I did:
CLI zap destroy
On Wed, 9 Jul 2008, Steve Totaro wrote:
On Wed, Jul 9, 2008 at 5:00 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 09 July 2008 14:06:56 Thameem Ansari wrote:
There is no performance impact if you use AGI or DeadAGI.
There is no significant difference (in Asterisk 1.2) in the
On Wed, 9 Jul 2008, Tilghman Lesher wrote:
On Wednesday 09 July 2008 16:50:09 Steve Totaro wrote:
On Wed, Jul 9, 2008 at 5:00 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
DeadAGI is not recommended and is not supported for channels which are
not already hungup (and invoked from the h
Klaus Darilion wrote:
Hi!
I want to test Asterisk--Siemens HiCom integration using Q.SIG instead
of ISDN. I did not find any documentation about Asterisk und Q.SIG.
Thus, I wonder is it sufficient to set switchtype from euroisdn to
qsig or are there any other things which I have to take
Jared Smith wrote:
On Wed, 2008-07-09 at 13:42 +0200, Jerome Poggi wrote:
I use them before some patch. But this example work :
exten = s,5,ChanIsAvail(SIP/604,s)
exten = s,6,Dial(SIP/604,15,wotr)
exten = s,7,NoOp(Nopnopnopnopnop)
exten = s,10,NoOp(Matthieu)
and this not :
exten =
H...you could build a site that used a voip service provided to
hook into people's mobiles or home phones
That way there's very little software development needed, and it 'just
works'.
You could deliver the full 'net phone' stuff as phase 2.
later,
PaulH
Dean Collins wrote:
Hi Paul, yes you could build it but the client I have in mind really
just wants a preformatted solution eg, someone has already set it up and
just gives them a cut and paste job to make it work on their website.
Cheers,
Dean
-Original Message-
From: Paul Hales [mailto:[EMAIL
Tilghman Lesher wrote:
On Wednesday 09 July 2008 09:08:50 John Millican wrote:
Can anybody tell me what I am doing wrong or why the Read application
does not accept the # key as input? My read statement:
exten = s,n,Read(uchoice|thankyouforcalling|3||1|1);
In the prompt thankyouforcalling
On Wednesday 09 July 2008 19:15:03 Steve Edwards wrote:
On Wed, 9 Jul 2008, Tilghman Lesher wrote:
On Wednesday 09 July 2008 16:50:09 Steve Totaro wrote:
On Wed, Jul 9, 2008 at 5:00 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
DeadAGI is not recommended and is not supported for
On Wednesday 09 July 2008 22:18:58 John Millican wrote:
Tilghman Lesher wrote:
On Wednesday 09 July 2008 09:08:50 John Millican wrote:
Can anybody tell me what I am doing wrong or why the Read application
does not accept the # key as input? My read statement:
exten =
Hi
On Thu, Jul 10, 2008 at 01:45:02AM +0200, Udo Schacht-Wiegand wrote:
Hi,
we have a strong echo on an Astribank running on
Asterisk 1.2.29-BRIstuffed-0.3.0-PRE-1y-s
The echo is on the opposite site of the analogue phones
connected to the Astribank.
So this is an FXS module.
I tried
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