get the
wrong time.
My system time (currently) Thu Jun 12 15:12:11 GST 2008
on asterisk i use EPOCH to look at the time, nbsp; NoOp(SIP/105101-00857e60,
DATE: 20080612-081147)
i would really appreciate any help. TIA
ron
--- On Thu, 6/12/08, Tilghman Lesher lt;[EMAIL PROTECTED]gt; wrote:
From
Hi Hans,
Can't you leave the picking up of the cli to the isdn line?
Even if it is an ISDN1 (just a B-channel and a D-channel),
the chances of tranferring channel info, like CLI, is better.
If a call comes in over the POTS line, then I still need to get CLI over it.
I'm not sure if the ISDN
Hello Fellow Users,
I am looking for a way - using certain software or other techniques - to
monitor, measure, and improve the quality of service for Asterisk system.
During the last while, it seems the quality has decreased and am trying to
look for ways to get things going well again.
Thanks,
Hi Tony,
Thanks a lot for the tips. I have turned on the logging and saw them in the
console. However, this applies only for startup, when I try to register a
user, which I cannot succeed, there is no logging done.
Do you think you can give me an idea why my user cannot register? All I want
to
then start asterisk but still i
get the wrong time.
My system time (currently) Thu Jun 12 15:12:11 GST 2008
on asterisk i use EPOCH to look at the time,
NoOp(SIP/105101-00857e60, DATE: 20080612-081147)
i would really appreciate any help. TIA
ron
--- On *Thu, 6/12/08, Tilghman Lesher
/[EMAIL
I appreciate the responses thus far but I am looking to find out what type
of security I should implement for the future. Being new to linux, not to
mention asterisk I didn't realize that someone could brute force into the
box and upload crap. With that in mind it seems that I would want to get a
Hi;
I would like just to know one thing:
Where did u find a good IAX IP Phone?
I am looking in the market since long time to buy such
device and did not find a reliable one till now.
Any advise?
Regards
Bilal
---
Hi,
I've been asked to spec up a small Asterisk
Dear All,
The documentation of the Dial Command, says the following about Option D:
D([called][:calling]) - Send the specified DTMF strings *after* the called
party has answered,
but before the call gets bridged.
However, in my experience, the timing the call get bridged is not
I am still looking to know if all of these h323's are
able to work as gatekeeper, so endpoint can register?
About chan_ooh323 and using It is clean the Asterisk
RTP stack (and can therefore bridge properly), and
doesn't creak under the bloat of OpenH323 like the
first two do:
The other two: how
Tilghman Lesher schrieb:
On Wednesday 11 June 2008 10:20:15 Brent Davidson wrote:
There is not, although I don't see any reason why it couldn't be done. There
is a ZapRAS application which performs much of this same function, although
it only works on ISDN lines (where the line signal is
on
gt; the system, then restart the system then start asterisk but still i
gt; get the wrong time.
gt;
gt; My system time (currently) Thu Jun 12 15:12:11 GST 2008
gt;
gt; on asterisk i use EPOCH to look at the time,
gt; NoOp(SIP/105101-00857e60, DATE: 20080612-081147)
gt;
gt; i would really
bilal ghayyad wrote:
I would like just to know one thing:
Where did u find a good IAX IP Phone?
I am looking in the market since long time to buy such device
and did not find a reliable one till now.
Any advise?
I haven't tried any yet; but http://x100p.eu have a few for sale; plus
Hi,
We can always count on Dean Collins to arrange interesting things and
this Friday, June 13th, we're LUCKY to have a very interesting offer
if you happen to be looking for an applicane.
See http://VoipUsersConference.org
This Friday the 13th we'll be hearing about the newest Asterisk
In article [EMAIL PROTECTED],
bilal ghayyad [EMAIL PROTECTED] wrote:
I am still looking to know if all of these h323's are
able to work as gatekeeper, so endpoint can register?
I think they all run only as a gateway, not a gatekeeper, but I'm not
100% certain.
About chan_ooh323 and using It
) Thu Jun 12 15:12:11 GST 2008
on asterisk i use EPOCH to look at the time,
NoOp(SIP/105101-00857e60, DATE: 20080612-081147)
i would really appreciate any help. TIA
ron
--- On *Thu, 6/12/08, Tilghman Lesher
/[EMAIL PROTECTED]/* wrote:
From: Tilghman Lesher
[EMAIL PROTECTED
On Thu, Jun 12, 2008 at 01:59:51AM -0700, Nhadie Ramos wrote:
hi mats,
i'm using 64-bit Ubuntu Server Edition 8.04
I just use GMT+0, but i'm on Singapore whcih should be at GMT+8, but if i use
GMT+8 the system does not give the correct time.
You should actually be using Asia/Singapure
I need to execute an action after a call is hangup. I just see the
command Dial has an option for that, the g option.
I configure the dial command as
exten = s,n,Dial(SIP/100,100,Ttg)
How should I add the line which the command will be executed after the
dial command in this example?
I don`t
What services do you need exposed to the internet and on what machines?
Does the fiber just terminate into your switch then? What type of
switch? Can you get access to the switch? If so you can probably
create access control lists.
You could put your own router in front to act as a firewall
hi sir,
i forgot to mention it was originally at Asia/Singapore, when i noticed
that asterisk has a wrong time, that's why i tried GMT instead.
regards,
Ron
Tzafrir Cohen wrote:
On Thu, Jun 12, 2008 at 01:59:51AM -0700, Nhadie Ramos wrote:
hi mats,
i'm using 64-bit Ubuntu Server Edition
Thanks for the response.
I have a tellabs 8813 switch provided from time warner. No I currently do
not have access to the switch. I am in the process of converting from analog
based dialers using dialogic hardware TO asterisk/ vicidial systems
I am strictly placing sip calls to my termination
Then I would think IPtables should work just fine for you. You have
local access to the * box? Even a simple NAT should probably work OK
with a little config tweaking.
Have a look here http://swik.net/iptables+sip
Thanks,
Steve
On Thu, Jun 12, 2008 at 7:03 AM, Mark Adams
[EMAIL PROTECTED]
Which is the way to run an AGI after hangup a call?
The problem I have is when the call dies the AGI dies too
I try the Dial command g option, but it does not work for me
Any clue will be welcomed.
Thanks
VoipCrazy
___
-- Bandwidth and Colocation
See, to get back to your answer, this is what I`m not understanding:
Again, this works fine. The problem is when I forward my calls to
another
outside line (using Polyocm phones), and need to know the ${did} value
at
that point. It's empty.
Right, so the call path is:
Provider --
Hi all,
I have setup an asterisk system which:
1. recieves incoming sip calls
2. ask the caller the number they want to dial, and then dial that number
3. after the caller is done talking and callee hangsup or even if the
callee does not answer the phone, the caller is asked for
On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
I have setup an asterisk system which:
recieves incoming sip calls
ask the caller the number they want to dial, and then dial that number
after the caller is done talking and callee hangsup or even if the callee
just add as many extensions as you want under the Dial command extension
keeping the extension number same:
exten = s,n,Dial(SIP/100,100,Ttg)
exten = s,n,Application here
On Thu, Jun 12, 2008 at 3:25 PM, voip crazy [EMAIL PROTECTED] wrote:
I need to execute an action after a call is hangup.
Hi,
With the g option, you just have to continue in the CALLER Dialplan, you
have nothing to do, just continue your Dialplan i.e:
exten= s,n,Dial(what you want) = and when the Called hangup you're goto
the next line
exten= s,n,Goto(where you want) or
exten= s,n, 'DO WHAT YOU WANT: playback,
LinkedIn
Josemar M#xfc;ller Lohn requested to add you as a connection on LinkedIn:
--
Ricardo,
I'd like to add you to my professional network on LinkedIn.
-Josemar M#xfc;ller
View invitation from Josemar M#xfc;ller Lohn
On Thursday 12 June 2008 03:23:46 Mark Adams wrote:
I appreciate the responses thus far but I am looking to find out what type
of security I should implement for the future. Being new to linux, not to
mention asterisk I didn't realize that someone could brute force into the
box and upload
Philippe Sultan wrote:
Friends,
a new dialplan application is now available for testing :
http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/
Sounds very cool.
See below for more comments:
The corresponding feature request is located here :
On Thursday 12 June 2008 03:50:30 Tobias Wolf wrote:
Tilghman Lesher schrieb:
On Wednesday 11 June 2008 10:20:15 Brent Davidson wrote:
There is not, although I don't see any reason why it couldn't be done.
There is a ZapRAS application which performs much of this same function,
I am also having this problem using includes based on time of day,
however the restart did not help and when enabled it finds no context
with extension 's'. This is for my incoming calls, see below...Any
Ideas!
On Thu, Jun 12, 2008 at 04:23:46AM -0400, Mark Adams wrote:
My situation seems unique because I am not using a router even at this
point. I was given a sheet of ip addresses and was told just to provision by
devices with the given ip's and they would handle the rest. My devices are
hooked
On Thu, Jun 12, 2008 at 07:03:52AM -0400, Mark Adams wrote:
I have a tellabs 8813 switch provided from time warner. No I currently do
not have access to the switch. I am in the process of converting from analog
based dialers using dialogic hardware TO asterisk/ vicidial systems
I am strictly
On Thu, Jun 12, 2008 at 08:02:24AM -0500, Tilghman Lesher wrote:
On Thursday 12 June 2008 03:23:46 Mark Adams wrote:
I appreciate the responses thus far but I am looking to find out
what type of security I should implement for the future. Being new
to linux, not to mention asterisk I didn't
On Thu, 2008-06-12 at 16:43 +0800, tcchan wrote:
However, in my experience, the timing the call get bridged is not
consistance,
Do you happen to be calling out over an analog phone line? In the case
of dialing out an analog line, we have no easy way of knowing when the
far-end has answered the
Tilghman Lesher wrote:
On Thursday 12 June 2008 03:23:46 Mark Adams wrote:
I appreciate the responses thus far but I am looking to find out what type
of security I should implement for the future. Being new to linux, not to
mention asterisk I didn't realize that someone could brute force
On Thu, 2008-06-12 at 01:23 -0700, bilal ghayyad wrote:
Where did u find a good IAX IP Phone?
I've had good success with my Allnet IP-7960 phones. They have the
ability in the firmware to either do SIP or IAX, and they even have a
mode where you dial one prefix to send the call out using the
On Thu, Jun 12, 2008 at 08:41:18AM -0500, Lyle Giese wrote:
Most recent hacks that I have first or second hand knowledge of
came from ssh issues. Most inexperienced admins will expose ssh
without using the 'allowgroups' option in their sshd_config and
will get hacked by someone
Hi,
While I haven't personally used any of their equipment yet, Brix is
supposed to have good h/w and software for measuring a MOS score:
http://www.brixnet.com/products/BrixCall.shtml
http://www.voiptroubleshooter.com/basics/mosr.html
-- James
Hello Fellow Users,
I am looking for a way -
Thanks for all the help. I have been in this biz for several years using
windows machines and analog dialers. I need to get on top of learning
enhanced networking, linux systems and firewalls.
Lots of goof information - Much appreciated!
Mark Adams
_
From: [EMAIL
HI,
I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair
command over Asterisk up till now and have run it in different scenarios
such as Call Center Solution, PBX solution.
There is a requirement to use Asterisk only as Voice Recording solution
in following manner:
Joe,
I'm unable to find an incident in your name in Digium Support's tracking
systems. Can you email me off-list with a case or reference number, or give me
contact information so one of our technicians can work with you to address your
echo issue?
A week-long wait is not typical of Digium
On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote:
HI,
I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair
command over Asterisk up till now and have run it in different scenarios
such as Call Center Solution, PBX solution.
There is a
Look for the DeadAGi command.
Thanks
l.
On Thu, 12 Jun 2008 13:41:14 +0200, voip crazy [EMAIL PROTECTED] wrote:
Which is the way to run an AGI after hangup a call?
The problem I have is when the call dies the AGI dies too
I try the Dial command g option, but it does not work for me
Any
Hello everybody
I have a problem using the 'qualify' option with iax2:
- snip --
Jun 12 16:11:14 VERBOSE[22657] logger.c: Tx-Frame Retry[000] -- OSeqno: 000
ISeqno: 000 Type: IAX Subclass: POKE
Jun 12 16:11:14 VERBOSE[22657] logger.c:
Thanks Steve,
How I can use it Asterisk as Man In The Middle. Since we have to keep
our Native PBX intact and functioning but only thing it doesn't handle
is Voice Recording. I thought if I can get some Channel Variable or some
system generated event regarding OFF-HOOK and ON-HOOK condition
You have to run DeadAGI, in h .
Regards
Andrea Cristofanini
voip crazy ha scritto:
Which is the way to run an AGI after hangup a call?
The problem I have is when the call dies the AGI dies too
I try the Dial command g option, but it does not work for me
Any clue will be welcomed.
Thanks
Jay R. Ashworth wrote:
On Thu, Jun 12, 2008 at 08:02:24AM -0500, Tilghman Lesher wrote:
One of the most frequent security issues comes not in the form of a
software flaw, but simply in people choosing easy-to-guess passwords
on the root account. There are two suggestions I have to reduce
Hi Julian,
[...]
What can you do with it? Well, a direct usage of this application is
to make an easy to use GoogleTalk voice gateway out of Asterisk. Here
is an example (assuming the asterisk-xmpp account is configured) :
context gtalk-in {
s = {
NoOp(Caller id :
Yes it all makes sense, I left it all open so sip traffic could pass. My
experience has only been with analog gateways which well no one would wasn't
to break into or do any of these things too.
Thanks for the sonicwall tip, that was what I was about to buy.
Mark Adams
-Original
Hi Philippe,
thanks for the replies. It all seems sensible.
Now, for a request ;)
How difficult would it be to have a JabberReceive Event *initiate* a
channel ?
This could be done by specifying a [EMAIL PROTECTED] in
jabber.conf
So, when a message is received by asterisk, a call is
You will need exactly two times the number of ports that your legacy
system has. Asterisk takes the call on _.,1,DAHDI, starts monitor and
dials out the second DAHDI port to your legacy system.
It is about ten lines in extensions.conf.
Thanks,
Steve T
On Thu, Jun 12, 2008 at 12:01 PM, Syed
Hi Sema,
In article [EMAIL PROTECTED],
Sema Arca [EMAIL PROTECTED] wrote:
Thanks a lot for the tips. I have turned on the logging and saw them in the
console. However, this applies only for startup, when I try to register a
user, which I cannot succeed, there is no logging done.
Check your
So is this Digium taking over Pulver's' void? Same color scheme and fonts.
Thanks,
Steve
-- Forwarded message --
From: Digium [EMAIL PROTECTED]
Date: Thu, Jun 12, 2008 at 10:34 AM
Subject: Complimentary Subscription to VoIP Industry Publication
To: [EMAIL PROTECTED]
If you
Thanks for your answers, DeadAGI was the solution.
Thanks again.
Voipcrazy
2008/6/12 Andrea Cristofanini [EMAIL PROTECTED]:
You have to run DeadAGI, in h .
Regards
Andrea Cristofanini
voip crazy ha scritto:
Which is the way to run an AGI after hangup a call?
The problem I have is when
Fail.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi,
I want to add some custom functions in voicemail.
For example user can switch SMS on/off or the voicemail global on/off.
Whats best way to do this?
modify app_voicemail.c or or do everything in dialplan?
or any other solutions (Asterisk 1.2.X please)
best regards
Thomas
Hi list,
I have some on hold activities that I would like to implement and I'm just
wondering if there is a way to do it. Here's what I'm thinking:
While a caller is on hold, they could have the option to do things like
retrieve weather, news, play blackjack( ala tellme), etc. These are all
The Asterisk.org development team has released Asterisk version 1.4.21.
This release is a regular bug fix release for the 1.4 series of
Asterisk. For a full list of changes, see the ChangeLog included in the
release.
* http://svn.digium.com/view/asterisk/tags/1.4.20/ChangeLog?view=markup
Dovid Bender wrote:
Hi List,
I was thinking the other day that even with T.38 there are still some
issues with faxing. I was thinking of a protocol that instead of just
sending down the fax tones an ATA or VOIP fax machine would get the
entire fax convert it into some sort of image and
Hi,
I need to know if the following features are available on asterisk
and their quality
-SMS
-Call control, budgeting and monitoring
-Video conferencing
-support for 500 extensions
-fax
-audio and video conferencing
and
1. Call accounting showing calls made
2.
Hello
I have asterisk with a sip connection with an external provider, getting the
calls from a toll free number.
Asterisk collects information from the callers, using also an script on windows
with the dial plan.
Debuging the script, I noticed that randomly asterisk losses some of the
Hi Forum,
nbsp;
While integrating a Nokia E61i cell phone into my Asterisk installation, I have
encountered an issue that I have pin-pointed to the phone's SIP protocol.
nbsp;
Upon the arrival of an incoming call, the dialplan set the variable
CALLERID(name)
to the caller's name, then it dials
Atis Lezdins wrote:
On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
I have setup an asterisk system which:
recieves incoming sip calls
ask the caller the number they want to dial, and then dial that number
after the caller is done talking and callee
yeah something like that. is it possible to set asterisk to make 10
calls per second?
On Thu, Jun 12, 2008 at 8:23 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
I know you can limit the total calls in any given time, for example,
you say I would like to have 10 SIP calls established as maximum.
snip
BUT if the two legs hangup, you have to use DEADAGI on the h extension..
Quick note, he doesn't necessarily have to use DeadAGI unless it's an
AGI being called. He just has to make sure he defines the h extension in
that context and set up the same executions as the post-dial
On Thu, Jun 12, 2008 at 9:14 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
I have setup an asterisk system which:
recieves incoming sip calls
ask the caller the number they want to dial, and
On Thu, Jun 12, 2008 at 9:16 PM, Mark Quitoriano
[EMAIL PROTECTED] wrote:
yeah something like that. is it possible to set asterisk to make 10
calls per second?
On Thu, Jun 12, 2008 at 8:23 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
I know you can limit the total calls in any given time, for
Well, as I said, you can tell Asterisk to accept until 10 SIP calls,
for example, at ANY TIME (I don't understand why per second, I mean,
if the 10 calls are established in the same second, they are acepted,
and so they are if they are established in the same milisecond, while
the max concurrent
Atis Lezdins wrote:
On Thu, Jun 12, 2008 at 9:14 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
I have setup an asterisk system which:
recieves incoming sip calls
ask the
On Thu, Jun 12, 2008 at 09:53:53AM -0400, Jay R. Ashworth wrote:
Additionally, you should install a brute-force-attack blocker:
http://www.la-samhna.de/library/brutessh.html
This is effectively another service listening. It is also a method for
an attacker to lock you out of the system.
I am trying to work in the console, figuring why it exits, but about 75%
is always taken up with
Really destroying SIP dialog '' Method: OPTIONS
Can anyone point me where I can stop this without turning down the
debugging/verbose on the entire console.
Hello list- I'm having an extremely odd issue with an installation of mine. The
system is running * 1.2.12.1 and currently handles around 100 handsets. With
the exception of a few Grandstream DTA's, all devices are Polycom 320, 430, or
601's. After a recent power outage, I'm having an extremely
Hello,
I'm wondering about following DUNDI setup
Suppose we have 2 Asterisks: astA and astB with DUNDI peering active
between them
and 2 SIP endpoints: sipA registered with astA and sipB regsitered
on astB
All this is on the same LAN
now sipA call an number which corresponds to [EMAIL
Hi guys, here I attach a call log made by Mariano (full log enabled in
logger.conf, set debug 100, set verbose 100 on asterisk console)...
1) first call, is one from pstn (EWSD) to our asterisk box.
http://rafb.net/p/HCoodb28.html
2) 2nd call, is one from a sip user registered at asterisk,
We're busily churning away at creating the Astricon
(http://www.astricon.net/) talk track this year, and it's been
delayed by a problem that we've never had in years past: too many
high-quality talk submissions. Not a bad problem to have, but still
a problem.
We have four tracks on the
On Thu, Jun 12, 2008 at 3:23 AM, OCG Technical Support [EMAIL PROTECTED]
wrote:
I'm looking at building up a standard asterisk system fanless/no moving
parts. I found a cheap solid state disk (Transcend TS32GSSD25S-M), but it
is SLOW...25mb/sec read 8mb/sec write.
I'm developing an asterisk
Hilary Miller hilfmil at gmail.com writes:
Something that I can put on our internal company website to replace
our hardware IP phones.
I see many web 2.0 startups offering browser based clients for their
own service, but I can't seem to find anything that I can use with my
own PBX. Do I
Hello,
I would recommend that if you do add another tech track that you spend
a great deal of effort trying to make sure that sessions that would
appeal to similar audiances are not done at the same time. This has
happened a few times in past Astricons and it's always a tough choice
for attendees
Lol - I was checking the analytics for my www.collins.net.pr/blog site
this afternoon and saw a funny search engine referral term - check out
the search words in the output below.
Funny how my site comes up first for that particular combination of
words. I hope he/she found what they were
On Thu, Jun 12, 2008 at 7:57 PM, Matt Florell [EMAIL PROTECTED] wrote:
Hello,
snip
To this end, I might suggest even video-recording the presentations to
be replayed at night during the conference(or possibly on the web) so
attendees can see what they missed if they were unable to sit in on a
but still i
gt; get the wrong time.
gt;
gt; My system time (currently) Thu Jun 12 15:12:11 GST 2008
gt;
gt; on asterisk i use EPOCH to look at the time,
gt; NoOp(SIP/105101-00857e60, DATE: 20080612-081147)
gt;
gt; i would really appreciate any help. TIA
gt;
gt; ron
gt;
gt; --- On *Thu, 6/12/08
i'm using 64-bit Ubuntu Server Edition 8.04
I just use GMT+0, but i'm on Singapore whcih should be at GMT+8, but
if
i use GMT+8 the system does not give the correct time.
You should actually be using Asia/Singapure rather than guess.
i'm not using ntp, coz when i do i also don't get
Hi Steve, thanks for your response...
I will try it this saturday and I'll let you know...
Best regards
On Wed, Jun 11, 2008 at 7:11 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Tue, Jun 10, 2008 at 1:40 PM, Raúl Gómez C. [EMAIL PROTECTED]
wrote:
Hi list,
I'm having trouble with calls
Any suggestions ?
Available options for the two settings similiar to the one identified are as
follows:
admin set send-dnis-type-of-number?
send-dnis-type-of-number:
Type of Number to be sent in called party IE in the setup message to
pstn. For ISDN signaling. To be used on egress gateway for
Basically, you run the phone lines into the asterisk box, then out of
the Asterisk system into the PABX.
This works reasonably well, and gives you the option to migrate to a
full asterisk setup in the future.
PaulH
Syed Nasruddin wrote:
Thanks Steve,
How I can use it Asterisk as Man In
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