Re: [asterisk-users] partial ChanSpy

[nik600]

i'm taking a look to app_chanspy.c what do you intend for trunk? the last cvs?

can i download the last cvs and then write a patch for the actual 1.2
branch stable?

thanks

On 8/3/07, James FitzGibbon <[EMAIL PROTECTED]> wrote:
> On 8/3/07, nik600 <[EMAIL PROTECTED]> wrote:
>
> > is it possible to spy (not record, spy) partially on a channel?
> >
> > for exaple, i'd like to listen only the input or the output voice.
> >
>
> trunk has added an 'o' option to ChanSpy:
>
>  "o - Only listen to audio coming from this channel.\n"
>
> You might be able to achieve what you want by alternately spying on either
> side of the bridged call using 'o' both times.
>
> I'm not sure if this would be portable back into 1.4 though or if you'll
> have to wait for 1.6 to be released.
>
> --
> j.
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-- 
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser

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Re: [asterisk-users] Unicall and Private CID

[Steve Underwood]

Carlos Chavez wrote:
>   It seems the problem with Unicall and Nextel is also present in
> Asterisk 1.2 and not only in 1.4.  I decided to downgrade from 1.4.9 to
> 1.2.23 so the customer could have CID and calls from Nextel but today he
> told me that they cannot receive any calls from Nextel, they get a busy
> tone every time.  I downloaded the following from softswitch:
>
> http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.3.tgz
> http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre11/libmfcr2-0.0.3.tar.gz
> http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre11/libsupertone-0.0.2.tar.gz
> http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre11/libunicall-0.0.3.tar.gz
> http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre11/asterisk-1.2.x/chan_unicall.c
> http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre11/asterisk-1.2.x/channels_Makefile.patch
>
>   The patch file fails in three places but I patched by hand.  All other
> calls come in and out, only calls from Private CID (like Nextel) get a
> busy tone all the time.  Could it be that this is something that got
> broken on more recent versions of libmfcr2? I have other systems
> installed over two years ago that do not have this problem.
>   
Use spandsp-0.0.2 with unicall-0.0.3

Steve


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[asterisk-users] VoiceMail Call Limit Messages

[GNUbie]

Hello all,

I was wondering if it's possible that when forwarded to a VoiceMail and
before the caller "can leave a message after the beep", the voice info will
inform the caller up to how many seconds or minutes can a caller stay to
leave the voicemail message.

Please advice.

Thank you in advance.

GNUbie
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[asterisk-users] IAX bat phone.

[Michael Munger]

Is there a way to setup an IAX bat phone (immediate=yes) or is this a
privilege only reserved for ZAP channels?

 

-Michael

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Re: [asterisk-users] Polycom 320 - Can it actually be configured?

[Michael Munger]

I find the absolute best way to provision phones is to use a TFTP server
or apache with .cfg files. It works every time. Just punching in the
numbers from the phone is such a pain, sometimes Polycoms seem to
"forget."

I have written a Windows app (GUI) that builds the files for you. If you
need it, let me know.

Yours,

Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug
Sent: Wednesday, August 01, 2007 10:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 320 - Can it actually be
configured?

At 16:49 8/1/2007, Douglas Garstang wrote:
 >Don't know about the 320, but we provisioned the 301's. They're
 >provisioning is basically the same as the 501's and 601's. What
problems
 >are you having?

Have no problems with 501s or 601s or 430s.

I punch in the provisioning server IP, but
the phone doesn't save it.  Usually, a phone
will prompt to save the config, but this
one doesn't.

 >
 >> -Original Message-
 >> From: [EMAIL PROTECTED]
[mailto:asterisk-users-
 >> [EMAIL PROTECTED] On Behalf Of Doug
 >> Sent: Wednesday, August 01, 2007 2:41 PM
 >> To: asterisk-users@lists.digium.com
 >> Subject: [asterisk-users] Polycom 320 - Can it actually be
configured?
 >>
 >> Just got one of these.  Horrible to program.
 >> Trying to key in the FTP server.  Won't even
 >> remember the info after rebooting.
 >>
 >> Anybody know the proper way to beat on this
 >> stupid beast so it will work?
 >>
 >>
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[asterisk-users] Handling message for SAPI/TEI=0/0 Repeated Quickly

[Steve Totaro]

Hello List,

I just turned up a T1 with Qwest.  All the channels came up and inbound 
and outbound work and sound great. 

The only issue I have is this message "Handling message for 
SAPI/TEI=0/0" repeated on the console a couple times a second.

I tried googling it but only found links to code in libpri.  I am 
running Asterisk SVN-trunk-r76015M.  I am going to upgrade to the latest 
trunk but does anyone know why this is happening or what it means?  It 
is not a warning or an error.

Thanks,
Steve

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[asterisk-users] queue beep

[Todd H]

Hi -
I have a queue setup with 4 agents.  The people working the queue  
tell me that when a new call goes into the queue, both the agent and  
the caller hear a tone.  These are static agents - could it be  
ringing the line again, even though the agent is on a call?  We are  
using GXP2000's with only 1 line programmed in.
thanks
Todd


queue.conf
> [601]
> announce-frequency=0
> monitor-format=wav
> monitor-join=yes
> music=default
> queue-callswaiting=
> queue-thankyou=queue-thankyou
> queue-thereare=
> queue-youarenext=
> retry=5
> rtone=0
> strategy=ringall
> timeout=15
> wrapuptime=1
> leavewhenempty=no
> eventwhencalled=no
> joinempty=Yes
> context=
> maxlen=0
> announce-holdtime=no
> eventmemberstatus=no
> member=Local/[EMAIL PROTECTED]/n,0
> member=Local/[EMAIL PROTECTED]/n,0
> member=Local/[EMAIL PROTECTED]/n,0
> member=Local/[EMAIL PROTECTED]/n,0
> member=Local/[EMAIL PROTECTED]/n,0
>

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Re: [asterisk-users] Unicall and Private CID

[Moises Silva]

I would not call that properly a fix. We need to know why is failing
in newer spandsp versions in the first place. Can you make a diff and
post it?

On 8/3/07, Carlos Chavez <[EMAIL PROTECTED]> wrote:
> On Fri, 2007-08-03 at 00:23 -0300, Luis Antonio Prata Barbosa wrote:
> > Hi Carlos,
> >
> > I suggest you download spandsp-0.0.3pre22.
> > (http://www.neuwald.biz/files/spandsp-0.0.3pre22.gz)
> >
> > I don´t know why , spandsp after that uses digits 1,2..8,9,A,B,C,D,E,F
> > instead of 1,2,..,9,0,A,B,C,D,E. So, do you get "F" digits that are
> > incompatible with mfcr2 .
> >
> Thank you.  I got an older set of files I had on another server (pre6)
> and now everything is working.  The customer now gets CID and calls from
> Nextel.
>
> This is probably the way to fix Unicall on 1.4 since it uses a newer
> version of spandsp and has the exact same problem.
>
> >
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
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>
>


-- 
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";

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Re: [asterisk-users] AGI and exec Playback

[Kevin Smith]

I'm not sure of a way to do it through AGI, but I know you could make 
the script take the recording, use sox to convert it to the file format 
you need, then maybe use like a Flash media player to control the 
playback of the sound file. It is a bit clunky but it was just one of 
the ideas (the better ones) that came to mind when I was reading this.

On our system, I created the option to call your extension with the call 
and play it back using ControlPlayback, or it converts it to a simple 
file format (such as wave, or mp3) and you can then download it and use 
a media player and do what you want with it.  Otherwise I'm not sure 
what you can or cannot control with AGI in reference to playing sound 
files.

Hopes this gives you a few ideas,
Kevin

Atis wrote:
> Hello,
>
> I'm looking for a way to play sound file, and control the playback
> trough web interface. Is it possible to use AGI to play a sound file
> and then by receiving some event stop playing it, and play another
> file. The catch is that i want to seek to 1st minute, 5th minute, etc
> - so regular ControlPlayback with intervals wouldn't fit  - i have to
> use sox to create different file and then jump to it.
>
> Also - i have read that in asterisk 1.4. there is SendDTMF trough AMI
> - is it possible to use that for ControlPlayback? Here i would want
> regular Forward/Backward buttons on web :)
>
> Regards,
> Atis
>
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Re: [asterisk-users] Measuring Jitter in Asterisk

[Douglas Garstang]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of John Todd
> Sent: Friday, August 03, 2007 2:12 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
> 
> At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:
> >
> >How can I objectively measure jitter in Asterisk on a SIP channel?
> >
> >I don't just want to turn the new 1.4 jitter buffer on. I want to
> >measure jitter.
> >
> >Thanks,
> >Doug.
> 
> You could look at the txjitter and rxjitter values (and other values)
> stored in the CHANNEL() function, and those values you're looking for
> were previously known as RTPAUDIOQOS.  Or is this not sufficient?

Are txjitter and rxjitter working reliably? These calls are going to be
placed from AMI and bridged together. Do you think the variables would
be correctly set for each leg of the call?

Doug.

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Re: [asterisk-users] Measuring Jitter in Asterisk

[Douglas Garstang]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of John Todd
> Sent: Friday, August 03, 2007 2:12 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
> 
> At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:
> >
> >How can I objectively measure jitter in Asterisk on a SIP channel?
> >
> >I don't just want to turn the new 1.4 jitter buffer on. I want to
> >measure jitter.
> >
> >Thanks,
> >Doug.
> 
> You could look at the txjitter and rxjitter values (and other values)
> stored in the CHANNEL() function, and those values you're looking for
> were previously known as RTPAUDIOQOS.  Or is this not sufficient?

Thanks John. Missed those... they're not documented... not even in 'show
function CHANNEL'.

Doug.

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Re: [asterisk-users] Macro and Arguments

[bilal ghayyad]

Dear James;

Thanks a lot for your kindly help and reply.

Here is the question: what is the CONSOLE variable
that is related to the channel? What it means and to
what it indicates this variable?

So when we say that Console/dsp, what does that mean?

Because console term is related to the console used
with the asterisk command that we connect to it using:

/usr/sbin/asterisk -r

So what is the diffrenece between this and the console
variable related to the channel?

Regards
Bilal

> At the extensions.conf file, at [demo] context,
there
> is a line:
>
> exten => 1234,n,Macro(stdexten, 1234,
> ${GLOBAL(CONSOLE)})
>
> In this line, I understand that it calls the macro
> name stdexten [macro-stdexten] but about the other
> variables, do we consider 1234 is ARG1 and the
> ${GLOBAL(CONSOLE)} is the ARG2? This is important to
> distinguish the arguments inside the macro.


Correct.

>From the other side, why it used ${GLOBAL(CONSOLE)}
to
> retreive the variable and did not write it directly
> ${CONSOLE} as already CONSOLE is configured in the
> [global] or what is the storey :) - ?


Using ${CONSOLE} relies on magic - it looks for a
channel variable, and
 when
one is not found, it falls back to the global var.  If
CONSOLE happened
 to
be defined on the channel, it would be returned
instead of the global
 var.

Using the GLOBAL dialplan function lets you get at
global variables
 whether
or not direct access to them is occluded by a
like-named local or
 channel
variable.

-- 
j.



   

Pinpoint customers who are looking for what you sell. 
http://searchmarketing.yahoo.com/

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Re: [asterisk-users] Measuring Jitter in Asterisk

[Douglas Garstang]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Alex Balashov
> Sent: Friday, August 03, 2007 2:12 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
> 
> On Fri, 3 Aug 2007, Jared Smith wrote:
> 
> > If the provider sends RTCP packets, you could simply watch for those
and
> > write the data to a database.  (I think modern versions of Asterisk
even
> > allow you to get to the data from the dialplan, and possibly from
the
> > Manager Interface.)  That at least gives you some per-call
statistics.
> 
>If you want to go that route, just yank those packets out of a
> constantly running tcpdump process with the right filters, and then
> process them with a script and load that data into a DB.

Alex, ok... so if I wanted to measure jitter to an ITSP I could run
tcpdump to it, and parse the output. According to
http://wiki.wireshark.org/RTP_statistics, I'd have to compare the
timestamp in each RTP packet with the timestamp shown by tcpdump. Looks
kinda complicated.


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Re: [asterisk-users] Unicall and Private CID

[Carlos Chavez]

On Fri, 2007-08-03 at 00:23 -0300, Luis Antonio Prata Barbosa wrote: 
> Hi Carlos, 
>  
> I suggest you download spandsp-0.0.3pre22.
> (http://www.neuwald.biz/files/spandsp-0.0.3pre22.gz)
>  
> I don´t know why , spandsp after that uses digits 1,2..8,9,A,B,C,D,E,F
> instead of 1,2,..,9,0,A,B,C,D,E. So, do you get "F" digits that are
> incompatible with mfcr2 .
>  
Thank you.  I got an older set of files I had on another server (pre6)
and now everything is working.  The customer now gets CID and calls from
Nextel.

This is probably the way to fix Unicall on 1.4 since it uses a newer
version of spandsp and has the exact same problem.

> 
-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Measuring Jitter in Asterisk

[John Todd]

At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:
>
>How can I objectively measure jitter in Asterisk on a SIP channel?
>
>I don't just want to turn the new 1.4 jitter buffer on. I want to 
>measure jitter.
>
>Thanks,
>Doug.

You could look at the txjitter and rxjitter values (and other values) 
stored in the CHANNEL() function, and those values you're looking for 
were previously known as RTPAUDIOQOS.  Or is this not sufficient?

I opened a request ticket to allow viewing of arbitrary CHANNEL() 
data on any active channel, but to my knowledge it has not been 
implemented.  The RTP source of media has however been impelemented 
in the CHANNEL() structure.  It may be possible to use chan_local to 
ascertain media data on the "other" leg of a call, but I have not 
experimented with that.

http://bugs.digium.com/view.php?id=9620

JT

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Re: [asterisk-users] TE220B

[Jared Smith]

On Thu, 2007-08-02 at 09:32 -0400, Remi Quezada wrote:
> Has anyone ever had any problem with the TE220B card with it showing up 
> as four ports instead of two.  I RMA'd the first one with the retailer 
> (Digium tech advice), but I just got another brand new card and it is 
> coming up as four ports again.  The card identifier is showing 0420 when 
> I do lspci.  Has this happened to anyone and if so is there a fix?

Please contact Digium support directly about this issue.  There were a
number of these cards that were incorrectly programmed, and they I'm
sure they'd be happy to send you a replacement card.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Measuring Jitter in Asterisk

[Alex Balashov]

On Fri, 3 Aug 2007, Jared Smith wrote:

> If the provider sends RTCP packets, you could simply watch for those and
> write the data to a database.  (I think modern versions of Asterisk even
> allow you to get to the data from the dialplan, and possibly from the
> Manager Interface.)  That at least gives you some per-call statistics.

   If you want to go that route, just yank those packets out of a 
constantly running tcpdump process with the right filters, and then
process them with a script and load that data into a DB.

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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Re: [asterisk-users] Measuring Jitter in Asterisk

[Jared Smith]

On Fri, 2007-08-03 at 13:38 -0700, Douglas Garstang wrote:
> What I do want to do is record QoS data to every single ITSP in real
> time. 

If the provider sends RTCP packets, you could simply watch for those and
write the data to a database.  (I think modern versions of Asterisk even
allow you to get to the data from the dialplan, and possibly from the
Manager Interface.)  That at least gives you some per-call statistics.  

Beyond that, there's not a whole lot you can do with Asterisk itself,
unless someone gets around to writing a Call Quality Detail Record
module for Asterisk that would log the call quality stats on a
call-by-call basis.

Another option might be Packet Island's VoIPCare for Asterisk[1].  It
sounds like a nice solution, but I haven't tried it, so I can't say
whether or not it would work for you in your particular circumstances.

[1] http://www.packetisland.com/page-voipcare-for-asterisk.html


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] partial ChanSpy

[James FitzGibbon]

On 8/3/07, nik600 <[EMAIL PROTECTED]> wrote:

is it possible to spy (not record, spy) partially on a channel?
>
> for exaple, i'd like to listen only the input or the output voice.
>

trunk has added an 'o' option to ChanSpy:

"o - Only listen to audio coming from this channel.\n"

You might be able to achieve what you want by alternately spying on either
side of the bridged call using 'o' both times.

I'm not sure if this would be portable back into 1.4 though or if you'll
have to wait for 1.6 to be released.

-- 
j.
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Re: [asterisk-users] DIALSTATUS not set

[Julian Lyndon-Smith]

Oh, for god's sake.

how stupid is I am feeling :)

My brain cell is feeling very ashamed.

Julian.

James FitzGibbon wrote:
> On 8/3/07, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
>> why if I call the Busy or Congestion extensions, the DIALSTATUS and
>> HANGUPCAUSE variables are not set ?
>>
>> If I call (say) extension 1234 all things are set ok.
> 
> 
> I think you've answered your own question there.  The only asterisk
> application that sets DIALSTATUS is Dial().  If you grep the source, you'll
> see that the value is retrieved by some other modules (chan_sip, chan_iax,
> etc.), but only Dial() sets the value of the variable.
> 
> I assume when you say "when I call the Busy extension" you mean something
> like a SIP user whose context is "outgoing" doing an INVITE to "
> [EMAIL PROTECTED]".  If so, you're bridging a SIP call leg to an asterisk
> application, so Dial() isn't invoked and DIALSTATUS isn't set.
> 
> It might work if you did an invite to an extension that used Dial() to call
> a Local channel (e.g. Local/[EMAIL PROTECTED]), but I'm not sure how 
> DIALSTATUS
> would interact with the /n option on the local channel.
> 
> 
> 
> 
> 
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Re: [asterisk-users] Measuring Jitter in Asterisk

[Douglas Garstang]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jared Smith
> Sent: Friday, August 03, 2007 1:22 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
> 
> On Fri, 2007-08-03 at 12:31 -0700, Douglas Garstang wrote:
> > How can I objectively measure jitter in Asterisk on a SIP channel?
> 
> > I don't just want to turn the new 1.4 jitter buffer on. I want to
> > measure jitter.
> 
> You can use Wireshark (formerly Ethereal) to analyze the RTP stream
> after it's been captured.  You can either use Wireshark itself to do
the
> network capture, or you can capture the traffic with tcpdump and then
> pull the file into Wireshark at a later time.

Jared, that won't do. I don't want to run the wireshark GUI, and I don't
wan't to run it on every single Asterisk box, connecting back to a local
X server running on my desktop. I also don't want to capture the RTP
data, and store it somewhere for later analysis. I'm looking at a
situation here with millions of subscribers and dozens of ITSP's.

What I do want to do is record QoS data to every single ITSP in real
time. I can then lease cost route based not just on route cost, but also
on historical QoS data. Whatever tool is used to collect the QoS data
has to stick it somewhere, such as MySQL, and then when I route a call,
I will have to query that data from MySQL.

> 
> Inside Wireshark, go to Statistics, RTP, Show All Streams, and then
> select a stream and hit the "Analyze" button.

I'm trying to avoid post-eyeballing the data.


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Re: [asterisk-users] Measuring Jitter in Asterisk

[Douglas Garstang]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Alex Balashov
> Sent: Friday, August 03, 2007 1:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
> 
> On Fri, 3 Aug 2007, Douglas Garstang wrote:
> 
> > If it COULD, you could leave a tshark process running, constantly
> > measuring jitter in real time. You'd run one for each ITSP you use,
and
> > voila, you have real time jitter metrics on a provider by provider
> > basis.
> 
>There are various command-line SIP performance test tools (sipp?)
that
> can do this too, I think.

I don't think you could do this with SIPP 

> 
>Also, it may be possible to modify Wireshark's plugin to
periodically
> invoke its jitter analysis function automatically and export the
results
> to some retrievable location.  The most difficult problem would be
> giving it a particular data stream to home in on as a VoIP call;  the
> easiest thing there would be to nail up your own periodic tests from
> a SIP UAC with definable IP endpoint locations and constantly run it
> with that filter.
> 
>Hackjobs aside, this sort of thing is essentially what products
like
> Brix do, as well as check in with SRTP stats.

Ok, maybe I should call them. But, as I said, if all their product does
is measure QoS and then give you pretty graphs to eyeball, it isn't much
use.

I need something that can measure jitter, latency etc in real time and
then stick the results somewhere, such as in MySQL. I can then choose
ITSP's based not just on route cost, but on a combination of route cost
and historical QoS data.

Doug.


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Re: [asterisk-users] Measuring Jitter in Asterisk

[Alex Balashov]

On Fri, 3 Aug 2007, Douglas Garstang wrote:

> If it COULD, you could leave a tshark process running, constantly 
> measuring jitter in real time. You'd run one for each ITSP you use, and 
> voila, you have real time jitter metrics on a provider by provider 
> basis.

   There are various command-line SIP performance test tools (sipp?) that
can do this too, I think.

   Also, it may be possible to modify Wireshark's plugin to periodically
invoke its jitter analysis function automatically and export the results
to some retrievable location.  The most difficult problem would be
giving it a particular data stream to home in on as a VoIP call;  the
easiest thing there would be to nail up your own periodic tests from
a SIP UAC with definable IP endpoint locations and constantly run it
with that filter.

   Hackjobs aside, this sort of thing is essentially what products like
Brix do, as well as check in with SRTP stats.

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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Re: [asterisk-users] Measuring Jitter in Asterisk

[Jared Smith]

On Fri, 2007-08-03 at 12:31 -0700, Douglas Garstang wrote:
> How can I objectively measure jitter in Asterisk on a SIP channel?

> I don’t just want to turn the new 1.4 jitter buffer on. I want to
> measure jitter.

You can use Wireshark (formerly Ethereal) to analyze the RTP stream
after it's been captured.  You can either use Wireshark itself to do the
network capture, or you can capture the traffic with tcpdump and then
pull the file into Wireshark at a later time.

Inside Wireshark, go to Statistics, RTP, Show All Streams, and then
select a stream and hit the "Analyze" button.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] partial ChanSpy

[Jaswinder Singh]

Chanspy() app allows spying live channel but you will get 2 way voice  in it
. I dont think any other app allows to spy on one side of call .

On 03/08/07, nik600 <[EMAIL PROTECTED]> wrote:
>
> Hi
>
> is it possible to spy (not record, spy) partially on a channel?
>
> for exaple, i'd like to listen only the input or the output voice.
>
> is it possible?
> thanks
>
>
> --
> /*/
> nik600
> https://sourceforge.net/projects/ccmanager
> https://sourceforge.net/projects/nikstresser
>
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Re: [asterisk-users] Measuring Jitter in Asterisk

[Douglas Garstang]

Did a little research.

 

Wireshark can graph jitter measurement. That's cool, but pretty useless.

 

Now, what would be REALLY cool, was if tshark, the command line tool,
could measure jitter. It looks like it lacks this feature.

 

If it COULD, you could leave a tshark process running, constantly
measuring jitter in real time. You'd run one for each ITSP you use, and
voila, you have real time jitter metrics on a provider by provider
basis.

 

But... tshark doesn't' support this. Arrgh!

 

Doug.

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Friday, August 03, 2007 12:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Measuring Jitter in Asterisk

 

How can I objectively measure jitter in Asterisk on a SIP channel?

 

I don't just want to turn the new 1.4 jitter buffer on. I want to
measure jitter.

 

Thanks,

Doug.

 

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Re: [asterisk-users] DIALSTATUS not set

[Jared Smith]

On Fri, 2007-08-03 at 19:58 +0100, Julian Lyndon-Smith wrote:
> why if I call the Busy or Congestion extensions, the DIALSTATUS and 
> HANGUPCAUSE variables are not set ?

The DIALSTATUS channel variable is set when you call the Dial()
application.  If you don't call the Dial() application (like if you
called the Congestion extension directly in your example), then it won't
be set.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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[asterisk-users] Measuring Jitter in Asterisk

[Douglas Garstang]

How can I objectively measure jitter in Asterisk on a SIP channel?

 

I don't just want to turn the new 1.4 jitter buffer on. I want to
measure jitter.

 

Thanks,

Doug.

 

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Re: [asterisk-users] DIALSTATUS not set

[James FitzGibbon]

On 8/3/07, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
>
> why if I call the Busy or Congestion extensions, the DIALSTATUS and
> HANGUPCAUSE variables are not set ?
>
> If I call (say) extension 1234 all things are set ok.


I think you've answered your own question there.  The only asterisk
application that sets DIALSTATUS is Dial().  If you grep the source, you'll
see that the value is retrieved by some other modules (chan_sip, chan_iax,
etc.), but only Dial() sets the value of the variable.

I assume when you say "when I call the Busy extension" you mean something
like a SIP user whose context is "outgoing" doing an INVITE to "
[EMAIL PROTECTED]".  If so, you're bridging a SIP call leg to an asterisk
application, so Dial() isn't invoked and DIALSTATUS isn't set.

It might work if you did an invite to an extension that used Dial() to call
a Local channel (e.g. Local/[EMAIL PROTECTED]), but I'm not sure how DIALSTATUS
would interact with the /n option on the local channel.

-- 
j.
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Re: [asterisk-users] Teliax Quality of Service

[Anthony Francis]

Haudy Kazemi wrote:
> On Aug 2 2007, John Meksavan wrote:
>
>   
>> Asterisk Users,
>>
>>  I recently ran into some problems with the quality of service with 
>> Teliax.
>>  This occurred on August 1, 2007 with a dropped outbound call, audio 
>> quality isse on the callee side- not hearing me well on callee side, and 
>> sending DTMF tones (configured for RFC2833). Am I the only Teliax 
>> customer having this problem?
>>
>>  It seems like when I am ready to go live with my Asterisk PBX System, I 
>> run into quality of service issues with the SIP provider.  Who should I go 
>> with that would guarantee me quality service just like an analog line?
>> 
>
> VoIP is susceptible to packet delivery problems anywhere between your PBX 
> and your SIP provider's PRI lines/termination point. If you have direct SIP 
> PBX to SIP PBX calls, then your problems can be anywhere on the Internet 
> path between the sites. The only workaround that I know of is having your 
> ISP be your SIP provider, so that your SIP packets only cross your ISP's 
> own network to its termination point, and do not cross the public Internet. 
> This way QoS can work from your office to your ISP's office to make sure 
> you maintain reliability.
>
> I have not personally used iTEL-ip's 'iTEL Voice Service', but others have 
> said, as do their own notes that their network QoS is effective at 
> maintaining call quality. When I contacted them, their pricing for a 'QoS 
> private IP backbone for voice and data' was $618/month for a full 1.5mbps 
> T1. Then SIP trunks (#11-24) were anywhere from $10-12 per month depending 
> on contract length. Per minute rates were $.03.
>
> When I ran the numbers, it appeared that a regular full T1 + a regular full 
> PRI would be only slightly more. A major tradeoff comes in the physical 
> location flexibility you get with SIP over traditional phone lines in the 
> case you need to move an office (although physically moving the phones to a 
> non iTEL-ip data line would mean you're not getting their Qos).
>
> iTEL-ip's 'iTEL Voice Service' 
> http://www.itelconnect.com/default.aspx?type=t§ion=iTEL-ipVoiceService&selection=16
>
> http://wiki.pbxnsip.com/index.php/ITEL-ip
>
> -hk
>
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>   
On the original problem of missed DTMF set dtmfmode=info in your sip.conf.

Anthony

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[asterisk-users] DIALSTATUS not set

[Julian Lyndon-Smith]

I'm trying to write a dialplan that will allow me to "stress" test it. I 
want to be able to dial an extension, or pretend that the extension is 
busy or out of order (so that I can see what to do)

given the dialplan snippet:

[outbound]

exten => _X.,1,NoOp(${TEST})
exten => _X.,n,Dial(SIP/${EXTEN})

exten => Busy,1,Busy(2)
exten => Busy,n,Hangup()

exten => Congestion,1,Congestion(2)
exten => Congestion,n,Hangup()

exten => NoAnswer,1,Wait(10)
exten => NoAnswer,n,Hangup()

exten => h,1,NoOp(X)
exten => h,n,NoOp(${DIALSTATUS}:${HANGUPCAUSE})

why if I call the Busy or Congestion extensions, the DIALSTATUS and 
HANGUPCAUSE variables are not set ?

If I call the NoAnswer extension, DIALSTATUS is blank and hangupcause is 
16. I presume that this is correct ?

If I call (say) extension 1234 all things are set ok.

Julian.




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[asterisk-users] Time Limit on Call or Conference Room?

[JR Richardson]

Hi All,

I recently had an incident where a conf bridge was left open due to
improper disconnection.  I've read about the meetme options and marked
callers closing the bridge when they exit.  This is OK for meetme, but
I'm really interested in a call timer that can be set on inbound and
outbound calls within the dial plan, per call.

I have another customer who wants to offer free calls, for 5-10
minutes with auto disconnect.

Can anyone point me int he right direction?

Thanks.

JR
-- 
JR Richardson
Engineering for the Masses

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Re: [asterisk-users] Time Limit on Call or Conference Room?

[Alex Balashov]

On Fri, 3 Aug 2007, JR Richardson wrote:

> Can anyone point me int he right direction?

   At the risk of coming off in a gratuitiously self-aggrandising manner
quoting myself:

   http://lists.digium.com/pipermail/asterisk-users/2007-May/188438.html

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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[asterisk-users] Several doubts on Asterisk as an UAC

[Filipe Brandenburger]

Hi,

I'm new to Asterisk and I've been trying to configure it to talk to
several SIP providers (such as FWD). I found that, although there are
some "recipes" on how to do it, there are few documents that really
explain *why* the settings are used, and overall I found very little
documentation on sip.conf.

I've been using this page as a reference:
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
It's very good. However, there's no explanation on "register" command,
for instance. There are also some other things that are not completely
clear.

That's why I wanted to ask lots of questions about it, and hopefully
I'll get some good answers here.

I'll focus only on having Asterisk being a SIP *client* (I believe this
is called UAC [User Agent Client] in the SIP world, right?) connecting
to a SIP provider (such as FWD). I'm using asterisk 1.4.6, so I would
like to talk *only* about configuration on asterisk 1.4.x.



* SIP channels on outgoing calls

If on my sip.conf I have a section [myprovider], it always creates a new
SIP channel "SIP/myprovider", right? If I want to use it on
extensions.conf to call extension 464646 there, I can use:
Dial(SIP/myprovider/464646)
or:
Dial(SIP/[EMAIL PROTECTED])
Is that right?


If I don't want to keep the section on sip.conf, and myprovider's host
is "hostname.myprovider.com", and I login with user "myuser" and secret
"mypasswd", I can also use this information directly on the "Dial" like
this:
Dial(SIP/myuser:[EMAIL PROTECTED]/464646)
Is that right?


Do I always have to authenticate to make outgoing calls? For instance,
could I do just this?
Dial(SIP/hostname.myprovider.com/464646)
Would it work for some outgoing numbers but not others? I believe with a
provider that allows me to do outgoing PSTN calls it wouldn't work, but
with free providers it might... someone has some info on this?



* Difference between type=user and type=peer

I saw this:
http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer

But I still didn't understand completely how it works...

First of all, the configuration type=... will affect only *incoming*
calls, right?

If I got it right, when Asterisk receives a call it will try to match to
all the type=peer sections by matching the "Host:" field of the INVITE
packet to the host=... configurations of type=peer sections. Right?

If that fails, it will ask for an authenticated INVITE, then it will
take the username/secret from the authenticated INVITE packet, and then
will look for type=user sections and match with username=... fields.

Now, what does happen for type=friend sections? Can they have both
host=... and username=... fields and they'll match on both?

Could someone really explain what is the algorithm (or at least the
idea) of how the call is matched when it's being received?



* context= in [default] and on outgoing calls

Ok, so if I have [myprovider] section with context=something. When I do
an outgoing call by using "Dial(SIP/myprovider/464646)", does
context=... affect anything?

As I understand it, it only affects incoming calls, but I might be
wrong.

Another thing, the setting of context=... on [default] section will
affect all [provider] sections without context=..., right? What if I
don't specify any context on [default], what would be the default
context? What if there's no context or an invalid context on a section,
what would happen to incoming calls that match that section?



* What exactly are "fromuser" and "fromdomain"?

As I get it, they're used only in outgoing calls, right?

As I understand, they change some fields on the INVITE packet sent to
the provider. Right?

Now, why do I need them?

If I don't specify them, what is the behaviour? What's the default value
for these options?



* The use of "register => ..."

When I use "register", the *only* thing it does is that Asterisk will
send periodic REGISTER packets to the provider, right? This is useful
for the provider to know to which IP it should direct calls that it
receives for my extension.

When I use "register => ...", does it create a "SIP/..." channel? I
think it doesn't, but if you use the same syntax, which is
"SIP/myuser:[EMAIL PROTECTED]/${EXTEN}" it will create a
channel "on-the-fly", am I right?

When I use "register => ...", if after that I use
"SIP/hostname.myprovider.com/${EXTEN}", will it use the same
username/secret I used for registering?

Is it possible to register without a secret? Does it make sense?

Do I always have to put "register => ..." commands on the [default]
section? Does it make sense to put them on other sections? What would be
the side effects of that?



* The use of "insecure"

If I got this right, I should use "insecure" when I want specific tasks
to be done without authentication.

For example, if I'm registered to a provider and someone wants to call
me, the provider's host will send me an INVITE packet, right?

If I don't use "insecure", the Asterisk will answer with an error code
that

Re: [asterisk-users] How to use stun server?

[Miguel]

Well, there is a valid reason for embedding the IP and port inside SIP.
First, SIP can be proxied multiple times (a chain, the proxy of your
provider talks to other proxy and so on until you reach the other
person provider's proxy), this is the way it was designed and still
used, so if you detect the IP address using layer 3 means, you may be
connecting to a proxy, no the endpoint. The SIP conversation is always
relayed via the proxy chain (never directly), and the proxies
add/delete things as the proxy passes thru them.
The ip and port embedded inside SIP tell the endpoints where is the
real other endpoint.
Second, SIP as it's name indicates Session Iniciation Protocol, is not
a media protocol, in that case it would be called something like
"voice carry protocol"; SIP only carries the signaling to
create/destroy media sessions, capabilities negociation, etc (not only
voice, but multple video, games, messaging, presence,...), but doesn't
carry the media. Also SIP normally is implemented in TCP and this is
planly BAD for media streams, because of the delays and retries that
tcp incurrs to deliver a ordered and complete stream of data.
So the media streams can't be carried over tcp because of it's
time-delay sensibility. So? you carry media sessions over UDP. So, if
a packet doesn't arrive? sorry it's lost forever. a packet arrives
late? simple, drop it, but doesn't stops the delivery of the future
packets, as if you did it over tcp. The RTP packets go end to end
directly, not via the proxy chain, using a shorter route, and usually
carrying TOS flags and are applied QoS.
Adobe flash video streaming is a example of doing strange things; to
be compatible with web an browsers, they did an implementation of a
RTP protocol over TCP (the server has to explicitly watch receipts
timestamps, and only transmit the most up to date data, skiping
delayed data), usually works fine, but when things go wrong, they go
VERY wrong and the video suffers until you can clean the bottleneck.
Also there is no obligation of using RTP as a real time media
companion to SIP, you could use SRTP, ZRTP, or other propietary
prococol; for other things like games, the only games in town are
propietary media protocols.
This works wonderfully in the plain old internet, but in the NAT'ed
paranoid internet of our days, the clients put their private address
in the SIP fields, and the other end can only laugh because the
address is unreachable. There is the usefulness of STUN and TURN, to
detect you NAT type and external ip address and port so the incoming
RTP stream can reach the endpoint via a NAT provided "hole" (ip
address and port that forwards to the endpoint [internal] ip address
and port). Also an administrator can put a proxy in the network's
boundary that rewrites SIP requests and eliminate the use of STUN.
But problems don't end there, for certain types of NAT, firewalls and
network setups, there is NO incomming way. This is the point where
things become really difficult.

That is the use of STUN. You should read a little more to understand
the basics of the prococols, more if you work for an ITSP.


> Message: 1
> Date: Fri, 3 Aug 2007 20:24:17 +0500
> From: "Rizwan Hisham" <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] How to use stun server?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Message-ID:
> <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I'm sure there was a perfectly good reason for encoding the devices IP
> address inside the SIP data when they invented it, but right now, I can't
> think why
> one thing i still dont understnd. if the device we are using is a computer,
> and we r running a softphone on it. and side by side we are also surfing the
> net. then why is it so that web content is coming into the computer without
> any problem but rtp data is not. i think both the web application and
> softphone are using computer's local ip address in their requests. So whats
> the reason for this?
>
> I understand how stun works but thanx for explaining it in so simple and
> concise way.
>
> One other question which has been bothering me is:
> If the client phone is behind nat, that means there is NATTING going on
> between public internet and local net. Then why do we need stun? NATTING
> should handle the problem itself as it does for other applications running
> on the same computer where softphone is also running.
>
>
> On 8/2/07, Gordon Henderson <[EMAIL PROTECTED]> wrote:
> >
> >
> > On Thu, 2 Aug 2007, Rizwan Hisham wrote:
> >
> > > hi again.well i have been trying to know what is the relationship
> > > between asterisk and stun. what i mean is, i understand that a client
> > > requests stun server to know whether its behind a nat or not. if its
> > not,
> > > then its ok. if it is behind nat, then what? Now client knows what kind
> > of
> > > nat it is behind, what is the roll of asterisk in it. asterisk already
> > knows
> > > client's public ip w

Re: [asterisk-users] Macro and Arguments

[James FitzGibbon]

On 8/3/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:
>
> At the extensions.conf file, at [demo] context, there
> is a line:
>
> exten => 1234,n,Macro(stdexten, 1234,
> ${GLOBAL(CONSOLE)})
>
> In this line, I understand that it calls the macro
> name stdexten [macro-stdexten] but about the other
> variables, do we consider 1234 is ARG1 and the
> ${GLOBAL(CONSOLE)} is the ARG2? This is important to
> distinguish the arguments inside the macro.


Correct.

>From the other side, why it used ${GLOBAL(CONSOLE)} to
> retreive the variable and did not write it directly
> ${CONSOLE} as already CONSOLE is configured in the
> [global] or what is the storey :) - ?


Using ${CONSOLE} relies on magic - it looks for a channel variable, and when
one is not found, it falls back to the global var.  If CONSOLE happened to
be defined on the channel, it would be returned instead of the global var.

Using the GLOBAL dialplan function lets you get at global variables whether
or not direct access to them is occluded by a like-named local or channel
variable.

-- 
j.
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[asterisk-users] Macro and Arguments

[bilal ghayyad]

Hi List;

At the extensions.conf file, at [demo] context, there
is a line:

exten => 1234,n,Macro(stdexten, 1234,
${GLOBAL(CONSOLE)})

In this line, I understand that it calls the macro
name stdexten [macro-stdexten] but about the other
variables, do we consider 1234 is ARG1 and the
${GLOBAL(CONSOLE)} is the ARG2? This is important to
distinguish the arguments inside the macro.

>From the other side, why it used ${GLOBAL(CONSOLE)} to
retreive the variable and did not write it directly
${CONSOLE} as already CONSOLE is configured in the
[global] or what is the storey :) - ?

Regards,
--
Bilal Ghayad


   

Sick sense of humor? Visit Yahoo! TV's 
Comedy with an Edge to see what's on, when. 
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[asterisk-users] CONSOLE=Console/dsp

[bilal ghayyad]

Hi List;

In the extensions.conf file, at the [global] context,
there is a variable configured as:

CONSOLE=Console/dsp

What does it mean that? What dsp mean and it is
shortcut for what?

How can I use the core to get some data about such
thing ambiguous for me?

Regards,
--
Bilal Ghayad


   

Be a better Globetrotter. Get better travel answers from someone who knows. 
Yahoo! Answers - Check it out.
http://answers.yahoo.com/dir/?link=list&sid=396545469

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[asterisk-users] "Asterisk can be attacked using buffer overflow."

[Doug]

Forbes.com - Magazine Article



LAS VEGAS - Internet Security VoIP Vandals

Andy Greenberg, 08.02.07, 12:32 AM ET

Internet telephone services like Skype and Vonage are
starting to look less like digital gimmicks and more
like the next generation of voice communication.
They're cheaper than traditional phone services and
increasingly fast and reliable. But they may also be
far more hackable.

Security professionals at the Black Hat conference in
Las Vegas spent Wednesday outlining the exploitable
vulnerabilities in voice over Internet protocol
technology, or VoIP. In a series of presentations,
they demonstrated ways in which cybercriminals can
eavesdrop on VoIP calls, steal data from Internet
telephony devices, intercept credit card numbers from
VoIP connections and shut connections down altogether.

"VoIP is about convergence. The idea is that you save
money and resources and time," said Barrie Dempster, a
senior security consultant at Next Generation Security
Software who made a presentation at the conference.
"But convergent systems give you more avenues of
attack, more ways in. It's not a secure environment."

Because VoIP connects telephone calls via the
Internet, it shares the Internet's weaknesses,
Dempster argued. Those include vulnerability to denial
of service attacks, which overload servers with
thousands of simultaneous requests for data, as well
as basic hacking tactics like guessing the password of
users who fail to change default settings.

Peter Thermos, chief technology officer of Palindrome
Technologies, proved the point onstage: He played
snippets of conversations recorded by snooping on VoIP
calls, exploiting vulnerability in a common element in
VoIP communications known as media gateway control
protocol. "Using this weakness in MGCP, you can do
anything like reroute or tear down connections," He
said. "But eavesdropping is especially scary."

Thermos also described an exploitable hole in ZRTP,
one species of the VoIP language real-time transfer
protocol: ZRTP encrypts all transmitted sounds, but
not the numbers translated from tones. That means
hackers can listen for credit card information
communicated from touchtone phones.

Though the attacks on display were new, VoIP isn't:
Internet telephony has existed since the early '90s.
But Dempster says its increasing adoption hasn't led
to the patching of old bugs. In his presentation, he
described how Asterisk, an open-source VoIP
application, can be attacked using what he said was an
"extremely basic" method known as a buffer overflow.
"We point these problems out," he said, "But the
lessons aren't being taken."

New mobile devices are also drawing attention to VoIP
problems. Krishna Kurapati, founder and chief
technology officer of Sipera Systems, demonstrated
vulnerabilities of several Wi-Fi devices at
Wednesday's presentations, crashing a Blackberry and a
D-Link phone onstage by hacking their wireless
Internet connections. He also simulated the theft of
private data via VoIP from a laptop.

And VoIP attacks aren't just happening in onstage
demonstrations; businesses are increasingly being hit.
Several companies in the last year have been victims
of "toll fraud," a scheme in which hackers break into
a company's VoIP network and sell thousands of dollars
worth of long-distance minutes.

Eric Winsborrow of Sipera Systems says that the wave
of threats has been brought on by VoIP's new
popularity in the business world as well as the
technology's growing connection to the Internet at
large, instead of smaller networks. He also points to
plans at Microsoft to introduce VoIP applications into
upcoming software as a sign that the technology's
security issues are reaching a tipping point.

"There's a perfect storm of more openness and
mobility, more mainstream adoption, and new entrants
into the industry," he says. "The table stakes are
getting much bigger."


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Re: [asterisk-users] How to use stun server?

[Victor Toofic]

El Fri, Aug 03 de 2007 a las 20:24 +0500, Rizwan Hisham comentaba:
> I'm sure there was a perfectly good reason for encoding the devices IP
> address inside the SIP data when they invented it, but right now, I can't
> think why
> one thing i still dont understnd. if the device we are using is a computer,
> and we r running a softphone on it. and side by side we are also surfing the
> net. then why is it so that web content is coming into the computer without
> any problem but rtp data is not. i think both the web application and
> softphone are using computer's local ip address in their requests. So whats
> the reason for this?

Simple, web content uses TCP while RTP uses UDP to carry the data. In TCP
your computer needs to establish a connection with the remote side before
each one can send any data, the device which is doing the NAT realizes that
and creates a bridge between your computer's IP/Port and the remote site
IP/Port.

In the case of SIP/RTP over UDP is different. Your softphone sends the
signaling over a UDP port, the remote site receives the data and responses
back to the IP/Port it recived it from (the IP/Port of the NATing device),
the device which is doing the NAT knows that you have recently send data
over that IP/Port and routes it back to you. Thats why SIP signaling can work
fine even behind a NAT (nat=yes).

RTP flow is also different. Your softphone specify it wants to receive RTP
in a IP/Port (private IP/Port), when the remote site wants to send you RTP
data it cannot be routed because that address is private, it cannot send
the data to the address of the NATing device because the port this device
is using for your outgoing RTP is different than the port you specified.
So the RTP that is destinated to you gets lost.

> 
> I understand how stun works but thanx for explaining it in so simple and
> concise way.
> 
> One other question which has been bothering me is:
> If the client phone is behind nat, that means there is NATTING going on
> between public internet and local net. Then why do we need stun? NATTING
> should handle the problem itself as it does for other applications running
> on the same computer where softphone is also running.

NATting can, in someway, handle the problem when you originate the call, but
it cannot do it when someone wants to reach you later. The SIP header
"Contact" is used for this, when someone wants to reach you it uses the
address you specified in that header, so it must be a public IP address
which you obtained from the STUN server or another mean.


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Re: [asterisk-users] Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box

[Steve Totaro]

I just tried to call in after creating an account.

After the call connects, enter the show id: 22622# and your_PIN#

I dial in and am asked for the podcast id, I enter 22622# and am told 
that my passcode is not correct. I also tried just entering my passcode 
but got the same error message.

What am I doing wrong?

Thanks,
Steve

randulo wrote:
> Hi folks,
>
> The August 3 edition of our Friday conference call and podcast kicks
> off in just over and hour. I know the list isn't delivering properly
> but if a few people  get this it'll be better than none.
>
> We are going to be talking today about TDM inside and outside the box.
> I own some antiiquated X100P FXO and a couple of TDM400p with the FXS
> modules. This is how our company's litle pbx talks to two incoming
> POTS lines and three regular phones connected to it. It also has a
> long list of IAX and SIP providers connecting it to the rest of the
> world. I am currently in the US so I use one of my 800 numbers to take
> control of the asterisk box in Paris and make "local" calls in France
> for a few pennies a minute. We also can send and receive SMS and of
> course receive vmail via email.
>
> But enough about me. What are you doing about connecting? And more to
> today's point, what ATA are you using to connect without opening the
> box and installing hardware?
>
> Digium makes the IAXy, Sipura (whatever the name is today) has several
> SIP models,  Grandstream as well. What else is out there and
> how well do they work?
>
> Join us:
>
>  http://AsteriskUsersConference.org
>
> As Matt said somewhere, this conference is like a forum. It's a chance
> for you to give back some of the valuable information and experience
> into the community without writing a line of code. I've been using
> asterisk for a few years and while I don't write code for it, I've
> experimented a lot with lots of hardware and a long list of providers.
> I've had time to learn a lot about the real world of all this stuff
> and I'm willing to share what I know. How about you?
>
>
>
> On 7/29/07, randulo <[EMAIL PROTECTED]> wrote:
>   
>> Hi,
>>
>> I am going to be on the road for the next few days and with the
>> variable delay on the mailing list, I am posting this now, 4 days
>> before the conference. If you haven't yet listened or participated,
>> please consider doing it. We have a great kernel of people at all
>> levels of expertise and ideas and questions can be kicked around
>> immediately (well, there's a few milliseconds lag).
>>
>> This Friday we'll be talking about TDM solutions including ATA that do
>> IAX and SIP without opening the box and installing a card. Your
>> experience in this area would be appreciated. If you sell these
>> solutions come over and "pimp" them.
>>
>> You can find us here:
>>
>>  http://AsteriskUsersConference.org
>>
>> At this site there are three main conference pages, how to listen or
>> participate, a player page for the archived recordings and a page with
>> the extension for a SIP connection to the conference bridge. There are
>> also two links to other pages, a related blog and AsteriskTV which
>> will be getting more and better content and more formats due to the
>> issue of Flash not being compatible with 64-bit systems. I'm working
>> on this now and hope to have that done by mid September. If anyone
>> knows how to convert mp3 to oog on a FreeBSD system, let me know. The
>> video issues are going to be more complicated so if you have
>> suggestions, please post them or email them to me.
>>
>> Thanks to the numerous people who have been supportive of these
>> efforts including Mark Spencer and the guys at Digium.
>>
>> randy
>>
>> 
>
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>
>   


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Re: [asterisk-users] AGI SAY TIME

[ram]

On 8/2/07, Nitesh Divecha <[EMAIL PROTECTED]> wrote:
>
> Hello all,
>
> Can anyone help me with SAY TIME.
> Every time I ask to say time, it gives me wrong time.
> I want the system to say time, what ever I give to say.
> Is it possible?



Try to Sync with NTP

so the time will not change

ram
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Re: [asterisk-users] Slightly OT: SNOM & PoE

[Andrew Latham]

Correct, and they will if the wire attenuation varies a lot.  A phone
250ft away will need more power than one 20ft away...



On 8/3/07, Mark J Elkins <[EMAIL PROTECTED]> wrote:
> Anthony Cennami wrote:
> > Hello All,
> >
> > I apologize for the slightly off-topic question, but I'm sure that the
> > people best acquainted with the issue would be hanging around here.
> >
> > Question is, what are people using today to deploy PoE, and more
> > importantly, PoE to SNOM phones?
> We are running SNOM360's off a Planet POE Switch (FGSW-2620PVS)
>
> This gives us 24 x 10/100 with PoE on all (but the GigE) ports (Power
> over the data lines - not on unused copper pairs) and 2 GigE ports
> (Copper + SPF). We then plug the Asterisk machine into one of the GigE's
> and the other GigE into the rest of the network.
> This works well with a centralised wiring closet - now all the phones
> are on their own switch - making interference more un-lightly.  The
> Switch allows you to look at the power consumption - which technically
> means you can judge when there could be a "power problem". Oh - pretty
> well priced too.. we paid about US $680 for the switch - which makes it
> 'low end' cost wise. Currently has no SNMP   :-(   - but has a Linux
> friendly web front-end including setup of QoS, Access control and PoE
> power consumption.
>
>
>   POE Ports Status
>
> PortEnable  Power[mW]   Current [mA]
> PortEnable  Power[mW]   Current [mA]
> 1   Enable  9   46
> 2   Disable 0   0
> 3   Disable 0   0
> 4   Disable 0   0
> 5   Disable 0   0
> 6   Disable 0   0
> 7   Disable 0   0
> 8   Disable 0   0
> 9   Disable 0   0
> 10  Disable 0   0
> 11  Disable 0   0
> 12  Disable 0   0
> 13  Enable  426 8
> 14  Disable 0   0
> 15  Enable  233346
> 16  Disable 0   0
> 17  Disable 0   0
> 18  Disable 0   0
> 19  Enable  220645
> 20  Enable  499 48
> 21  Disable 0   0
> 22  Disable 0   0
> 23  Disable 0   0
> 24  Disable 0   0
>
>
> OK - so not many ports are active - but its interesting to see that each
> port seems to be consuming different levels of power...
>
> --
>   .  . ___. .__  Posix Systems - Sth Africa
>  /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, SCO ACE, Cisco 
> CCIE
> / |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496
>
>
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 LATHAMA (lay-th-ham-eh)
 [EMAIL PROTECTED]
 [EMAIL PROTECTED]
*/

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[asterisk-users] Asterisk, ISDN AVM C4 and Terrible noise

[Alainn]

Hello -

This problem has dogged me for weeks now.

I have installed the latest Asterisk (1.4.9), libpri 1.4.1, zaptel 1.4.4 -
and the TRUNK version of chan-capi.  also the HEAD version -

all compiled well under CentOs 5.0 - and it works with IAX soft phones properly.
 However, when I call from an ISDN phone through chan_capi to asterisk -

I hear no PlayBack
MusiconHold is violent data stream noise
I can hear what is said into the phone in the softphone
but in ISDN I hear noise, or nothing.

dialing in and out - no problem.

I have tried many things - the 1.2 branch of Asterisk and all else, the HEAD
version of chan_capi, stable - all of them.  but still the problem remains.
I have also tried several versions of the C4 firmware - no difference.

This card was working on a PII 2 CPU machine and a much earlier version of
asterisk and chan-capi.

any ideas?

My new machine is a Dual core Pentium 4300 w/ 4gb RAM.  Very fast, stable, but
it can not go into production until this is fixed!

thank you -



Álainn
*

"The cheese-mites asked how the cheese got there,
And warmly debated the matter;
The orthodox said that it came from the air,
And the heretics said from the platter."   Anon.

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Re: [asterisk-users] How to use stun server?

[Rizwan Hisham]

I'm sure there was a perfectly good reason for encoding the devices IP
address inside the SIP data when they invented it, but right now, I can't
think why
one thing i still dont understnd. if the device we are using is a computer,
and we r running a softphone on it. and side by side we are also surfing the
net. then why is it so that web content is coming into the computer without
any problem but rtp data is not. i think both the web application and
softphone are using computer's local ip address in their requests. So whats
the reason for this?

I understand how stun works but thanx for explaining it in so simple and
concise way.

One other question which has been bothering me is:
If the client phone is behind nat, that means there is NATTING going on
between public internet and local net. Then why do we need stun? NATTING
should handle the problem itself as it does for other applications running
on the same computer where softphone is also running.


On 8/2/07, Gordon Henderson <[EMAIL PROTECTED]> wrote:
>
>
> On Thu, 2 Aug 2007, Rizwan Hisham wrote:
>
> > hi again.well i have been trying to know what is the relationship
> > between asterisk and stun. what i mean is, i understand that a client
> > requests stun server to know whether its behind a nat or not. if its
> not,
> > then its ok. if it is behind nat, then what? Now client knows what kind
> of
> > nat it is behind, what is the roll of asterisk in it. asterisk already
> knows
> > client's public ip whether its behind nat or not, if the client is
> > registered. So how does stun simplify things if there are nat problems.
>
> There is no relationship between asterisk and STUN.
>
> > After requesting stun server and recieving the required information from
> > stun server.what happens next?
> > I hope im clear in stating my problem.
>
> I'm not a STUN/SIP protocol gury by any means, but this is my
> understanding (and it might be a bit simplistic)
>
> When something communicates with something else using SIP, the sending
> device (eg phone) puts it's own IP address inside the SIP data packet.
> That IP address is the IP address of the device - it doesn't know anything
> about anything else, just the IP address it has. This would work well if
> NAT hadn't been invented, unfortunately it was.
>
>
> The listening side (eg. asterisk), extracts this IP address and uses it to
> send data back.
>
> So if the originating device is behind NAT, and it's on (eg) 192.168.0.42
> then the other end, gets that IP address and tries to send data back to
> it.
>
> Which, as 192.168.0.42 is on a private network, it can't do.
>
> Oops.
>
> So the original device uses a STUN server to poke a few bytes over the
> interweb and the STUN server replys back with some information - such as
> the real external IP address and port numbers it's using.
>
> The STUN server is a tiny bit of software running on a host somewhere with
> a real IP address (or 2!) and is (or can be) quite independant of the
> asterisk server.
>
> Original device can then put those values returned from the STUN server
> inside the SIP data packets (rather than it's 'real' natted IP address)
> and send them off to the other end, which can then use them to send the
> replys back to.
>
> The device should only need to access the STUN server once in it's life,
> but devices periodically check, just in-case things have changed. They do
> not relay data through the STUN server.
>
> So that's for device to asterisk box.
>
> Asterisk boxes are supposed to be directly connected to the Internet with
> no NAT and a real live IP address. (or at least that's the best possible
> way to do it!)
>
> If they aren't ... Then the first thing you need to do is arrange
> port-forwarding from the firewall to the asterisk box. You'll need to
> forward the ports you need - eg. for SIP it might be 5060-5069 and for RTP
> it might be 1-2.
>
> But the asterisk server still needs to know what it's real external IP
> address is so it can put that in the SIP packets rather than it's own
> NATted address, and as asterisk can't use a STUN server, you need to
> explicitly tell it - this is in the sip.conf file and looks like:
>
>nat=yes
>localnet=192.168.2.0/24
>externip=1.2.3.4
>
> So now the asterisk server knows that anything that originates from the
> local network doesn't need to be translated, but anything going out needs
> to have the SIP data re-written with the real external IP address.
>
>
> Now (AIUI) some SIP proxys can look inside SIP data packets and see that
> the IP address given by the device is not the same as the IP address that
> the packet came from and adjust things accordingly.. Asterisk, not being a
> SIP proxy doesn't do this, so if your phone is talking to a server via a
> proxy, then you may not need to tell the phone about a STUN server. The
> people running the asterisk+SIP proxy will tell you if this is the case.
>
> I'm sure there was a perfectly good reason for encoding the device

Re: [asterisk-users] Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box

[randulo]

Hi folks,

The August 3 edition of our Friday conference call and podcast kicks
off in just over and hour. I know the list isn't delivering properly
but if a few people  get this it'll be better than none.

We are going to be talking today about TDM inside and outside the box.
I own some antiiquated X100P FXO and a couple of TDM400p with the FXS
modules. This is how our company's litle pbx talks to two incoming
POTS lines and three regular phones connected to it. It also has a
long list of IAX and SIP providers connecting it to the rest of the
world. I am currently in the US so I use one of my 800 numbers to take
control of the asterisk box in Paris and make "local" calls in France
for a few pennies a minute. We also can send and receive SMS and of
course receive vmail via email.

But enough about me. What are you doing about connecting? And more to
today's point, what ATA are you using to connect without opening the
box and installing hardware?

Digium makes the IAXy, Sipura (whatever the name is today) has several
SIP models,  Grandstream as well. What else is out there and
how well do they work?

Join us:

 http://AsteriskUsersConference.org

As Matt said somewhere, this conference is like a forum. It's a chance
for you to give back some of the valuable information and experience
into the community without writing a line of code. I've been using
asterisk for a few years and while I don't write code for it, I've
experimented a lot with lots of hardware and a long list of providers.
I've had time to learn a lot about the real world of all this stuff
and I'm willing to share what I know. How about you?



On 7/29/07, randulo <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I am going to be on the road for the next few days and with the
> variable delay on the mailing list, I am posting this now, 4 days
> before the conference. If you haven't yet listened or participated,
> please consider doing it. We have a great kernel of people at all
> levels of expertise and ideas and questions can be kicked around
> immediately (well, there's a few milliseconds lag).
>
> This Friday we'll be talking about TDM solutions including ATA that do
> IAX and SIP without opening the box and installing a card. Your
> experience in this area would be appreciated. If you sell these
> solutions come over and "pimp" them.
>
> You can find us here:
>
>  http://AsteriskUsersConference.org
>
> At this site there are three main conference pages, how to listen or
> participate, a player page for the archived recordings and a page with
> the extension for a SIP connection to the conference bridge. There are
> also two links to other pages, a related blog and AsteriskTV which
> will be getting more and better content and more formats due to the
> issue of Flash not being compatible with 64-bit systems. I'm working
> on this now and hope to have that done by mid September. If anyone
> knows how to convert mp3 to oog on a FreeBSD system, let me know. The
> video issues are going to be more complicated so if you have
> suggestions, please post them or email them to me.
>
> Thanks to the numerous people who have been supportive of these
> efforts including Mark Spencer and the guys at Digium.
>
> randy
>

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Re: [asterisk-users] H.323

[Alessandro Russo]

Hi,
I'm using H323 in asterisk 1.4.9
work well

On 8/3/07, yonoko molomo <[EMAIL PROTECTED]> wrote:
>
> Hi,
> I have used h323, oh323 and ooh323.
> My experience is that ooh323 does not work properly, i dont recommend it.
> I dont know why, but the sound is bad, with sound breaks. I also need
> to put some wait (2) functions after the answer( ) or playback( )
> functions, it think that asterisk takes some time to stablish the
> ooh323 channel (maybe it is due to other reason, i dont know exactly)
> but during this time no sound is played, so the first seconds of
> conversation or playback are cutted. ooh323 did not work for me at
> all.
>
> oh332 worked fine in asterisk 1.2 (i did not try in 1.4, i guess it is
> fine).
>
> h323 works fine in asterisk 1.4. it is the one i am using now, and i
> have no problems with it.
>
> bye now
>
> 2007/8/2, Rurouni Alucard <[EMAIL PROTECTED]>:
> > Hi there,
> >
> > I have use the H.323 module that comes with asterisk-addons and i
> > consider it (so far) VERY stable for my needs.
> > Im talking about 10,000 minutes at month , + or - , and never had a
> > crash or something bad about it.
> >
> > Personally, i recommend it,
> >
> >
> > --
> > J. P.
> > rakh at slackware-es dot com
> >
> > bilal ghayyad wrote:
> > > Hi List;
> > >
> > > Did any one tried the H.323 module? How much it is
> > > stable and work fine?
> > >
> > > Regards,
> > > 
> > > ITS
> > > IP Telephony and Contact Center Engineer
> > > Eng. Bilal Ghayad
> > > Mobile: 00965 9849460
> > >
> > >
> > >
> > >
> Ready
> for the edge of your seat?
> > > Check out tonight's top picks on Yahoo! TV.
> > > http://tv.yahoo.com/
> > >
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> > >
> >
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Re: [asterisk-users] Unicall and Private CID

[Moises Silva]

Carlos,

If you are interested we can meet us via MSN someday to debug the
problem. I don't know if that's possible though, since it seems is
your production server.

Moy

On 8/2/07, Luis Antonio Prata Barbosa <[EMAIL PROTECTED]> wrote:
> Hi Carlos,
>
> I suggest you download spandsp-0.0.3pre22.
> (http://www.neuwald.biz/files/spandsp-0.0.3pre22.gz)
>
> I don´t know why , spandsp after that uses digits 1,2..8,9,A,B,C,D,E,F
> instead of 1,2,..,9,0,A,B,C,D,E. So, do you get "F" digits that are
> incompatible with mfcr2 .
>
> Luis A P Barbosa.
>
> 2007/8/2, Carlos Chavez <[EMAIL PROTECTED]>:
> >
> >Here is a log with level 255 when a Nextel phone tries to call in:
> >
> > Aug  2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1  <- 0001  [1/   1/Idle  /Idle ]
> > Aug  2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 Detected
> > Aug  2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 Making a new call with CRN 32769
> > Aug  2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 1101  ->  [2/   2/Idle  /Idle ]
> > Aug  2 15:38:18 WARNING[32670]: chan_unicall.c:2644 handle_uc_event:
> > Unicall/1 event Detected
> > Aug  2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1  <- 3 on  [2/   2/Seize ack /Seize ack]
> > Aug  2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 1 on  ->  [2/   2/Seize ack /Seize ack]
> > Aug  2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1  <- 3 off [2/   2/Group A   /DNIS request ]
> > Aug  2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 1 off ->  [2/   2/Group A   /DNIS request ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1  <- 7 on  [2/   2/Group A   /DNIS request ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 1 on  ->  [2/   2/Group A   /DNIS request ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1  <- 7 off [2/   2/Group A   /DNIS request ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 1 off ->  [2/   2/Group A   /DNIS request ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1  <- 2 on  [2/   2/Group A   /DNIS request ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 1 on  ->  [2/   2/Group A   /DNIS request ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1  <- 2 off [2/   2/Group A   /DNIS request ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 1 off ->  [2/   2/Group A   /DNIS request ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1  <- 8 on  [2/   2/Group A   /DNIS request ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 6 on  ->  [2/   2/Group A   /DNIS request ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1  <- 8 off [2/   2/Group C   /Category req ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 6 off ->  [2/   2/Group C   /Category req ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1  <- 2 on  [2/   2/Group C   /Category req ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 1 on  ->  [2/   2/Group C   /Category req ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1  <- 2 off [2/   2/Group C   /ANI request  ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 1 off ->  [2/   2/Group C   /ANI request  ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1  <- F on  [2/   2/Group C   /ANI request  ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 R2 prot. err. [2/   2/Group C   /ANI request  ]
> > cause 32772 - Unexpected MF6 signal
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCall/1 1001  ->  [1/   1/Idle  /Idle ]
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:2644 handle_uc_event:
> > Unicall/1 event Protocol failure
> >-- Unicall/1 protocol error. Cause 32772
> > Aug  2 15:38:19 WARNING[32670]: chan_unicall.c:627 unicall_report:
> > MFC/R2 UniCa

Re: [asterisk-users] PRI/T1 data rate...

[Steve Totaro]

Andrew Joakimsen wrote:
>
>
> On 8/2/07, *Forrest W. Christian* <[EMAIL PROTECTED]  PROTECTED]>> wrote:
>
> Steve Totaro wrote:
>
> >I knew someone would have an explanation that makes sense.   I have
> >NEVER done anything but PRI from the Telco.  Wouldn't the question of
> >signaling and switchtype negate the need to ask for data rate?
> >
> >
> Yes.  But these are probably telco ordering droids, meaning that all
> they know is that they have to fill in the blanks.
>
> I recently ordered a LD PRI from a carrier.  I wanted PRI, switchtype
> either 5ESS or preferrably National.  The order got kicked because I
> didn't specify whether or not I wanted E&M and which type of e&m
> (immediate, wink, etc) I wanted.  I seem to recall a couple of other
> totally non-relevant questions that I had to specify as well...   Or,
> more specifically, convince the droid which was checking the order for
> completeness that they weren't needed.
>
> -forrest
>
>

I love fighting with these people trying to convince them that ANI and 
CallerID are not the same thing.  Even more senior tech do not seem to 
understand this.

When Global Crossing first rolled out it's IP backbone (delivered as a 
TDM DS3 PRI).  The only work around I could get them to do was sent 
*ani*dnis*, lol.  I had to whip up some dialplan magic to work with 
that.  Only problem was that about 30% of the time they would send 
missing info such as **dnis*  or *ani**.

I told them that we would not pay the bill for calls like that they are 
toll frees and we had no way of knowing if the billing was correct, and 
in the case of *ani**, the calls were useless since we had no way to 
route the call correctly.  We also could not reconcile bills coming from 
transfer partners due to missing ANI.

Finally, after sending CDR logs from asterisk every day for a month, 
they found the "bug" in their system. 

I swear, these guys make more work for us than what is in the job 
description.

Thanks,
Steve

Thanks,
Steve


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Re: [asterisk-users] pri "call by call" trunking?

[Erik Anderson]

On 8/2/07, Don Kelly <[EMAIL PROTECTED]> wrote:
> Hi, Erik,
>
> Never heard of call-by-call trunking.
>
> Are you in Minnesota? What carrier are you using?

Yes I am...this is for one of our branch offices, though, outside of Boston, MA.

-Erik

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Re: [asterisk-users] Difference between WaitExten and TIMEOUT (response)

[Baji Panchumarti]

  On 8/3/07, bilal ghayyad  wrote:

> Hi List;
>
> What is the difference between WaitExten function and
> TIMEOUT (response)? As I see that both are used to
> determine the allowed time to enter the digits, any
> one can advise?

 core show function TIMEOUT

 for different timeout parameters, I haven't used WaitExten

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[asterisk-users] partial ChanSpy

[nik600]

Hi

is it possible to spy (not record, spy) partially on a channel?

for exaple, i'd like to listen only the input or the output voice.

is it possible?
thanks


-- 
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser

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[asterisk-users] Difference between WaitExten and TIMEOUT (response)

[bilal ghayyad]

Hi List;

What is the difference between WaitExten function and
TIMEOUT (response)? As I see that both are used to
determine the allowed time to enter the digits, any
one can advise?

Regards
Bilal


  

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Re: [asterisk-users] Knowing zap channel status

[Tzafrir Cohen]

On Fri, Aug 03, 2007 at 11:16:06AM +0100, Julian Lyndon-Smith wrote:
> I'm trying to write a zap monitor program to visually display the status 
> of each channel. It's working well -:
> 
> However, one thing that I am still struggling with is knowing the status 
> of the zap channels when the program starts.
> 
> Zap show channels only seems to show an extension on an inbound call. I 
> don't know which channels are in use for outbound calls.
> 
> Anyone got an inspiration ?

On the Zaptel level you have the ztdiag program. But its "output" is
dumped to the kernel logs.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
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[asterisk-users] Knowing zap channel status

[Julian Lyndon-Smith]

I'm trying to write a zap monitor program to visually display the status 
of each channel. It's working well -:

However, one thing that I am still struggling with is knowing the status 
of the zap channels when the program starts.

Zap show channels only seems to show an extension on an inbound call. I 
don't know which channels are in use for outbound calls.

Anyone got an inspiration ?

Thanks

Julian

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Re: [asterisk-users] Teliax Quality of Service

[Haudy Kazemi]

On Aug 2 2007, John Meksavan wrote:

>Asterisk Users,
>
>  I recently ran into some problems with the quality of service with 
> Teliax.
>  This occurred on August 1, 2007 with a dropped outbound call, audio 
> quality isse on the callee side- not hearing me well on callee side, and 
> sending DTMF tones (configured for RFC2833). Am I the only Teliax 
> customer having this problem?
>
>  It seems like when I am ready to go live with my Asterisk PBX System, I 
>run into quality of service issues with the SIP provider.  Who should I go 
>with that would guarantee me quality service just like an analog line?

VoIP is susceptible to packet delivery problems anywhere between your PBX 
and your SIP provider's PRI lines/termination point. If you have direct SIP 
PBX to SIP PBX calls, then your problems can be anywhere on the Internet 
path between the sites. The only workaround that I know of is having your 
ISP be your SIP provider, so that your SIP packets only cross your ISP's 
own network to its termination point, and do not cross the public Internet. 
This way QoS can work from your office to your ISP's office to make sure 
you maintain reliability.

I have not personally used iTEL-ip's 'iTEL Voice Service', but others have 
said, as do their own notes that their network QoS is effective at 
maintaining call quality. When I contacted them, their pricing for a 'QoS 
private IP backbone for voice and data' was $618/month for a full 1.5mbps 
T1. Then SIP trunks (#11-24) were anywhere from $10-12 per month depending 
on contract length. Per minute rates were $.03.

When I ran the numbers, it appeared that a regular full T1 + a regular full 
PRI would be only slightly more. A major tradeoff comes in the physical 
location flexibility you get with SIP over traditional phone lines in the 
case you need to move an office (although physically moving the phones to a 
non iTEL-ip data line would mean you're not getting their Qos).

iTEL-ip's 'iTEL Voice Service' 
http://www.itelconnect.com/default.aspx?type=t§ion=iTEL-ipVoiceService&selection=16

http://wiki.pbxnsip.com/index.php/ITEL-ip

-hk

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[asterisk-users] SIP-6.0 software for Siemens

[Hans Witvliet]

Hi all,

I got two Siemens Optipoints 410 digital phones for evaluation.
It was preconfigured with a sip-4.1 firmware.
Somehow someone managed to wipe out the firmware.
>From the SIP-docu (on siemens website, not with the phone) it stated
that the lan cable MUST be connected to the phone BEFORE plugging in the
power-cable. (um, well so be it...)

Question, is there anybody around here that has the SIP-6.0 firmware for
the Siemens optipoint?
Neither my local supplier nor the contactperson on the siemens-wiki site
seems to respond (Probably on holiday)
On the Siemens site there are lengthy tutorials to to install firmware,
what all the new features and bugfixes are, but no point to howto obtain
the latest version

HtH, Hans
-- 
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[asterisk-users] B410P echo cancellation

[Stefano Arata]

We own an Asterisk box with the following configuration:

Linux Debian 4.0
Kernel 2.6.18
Asterisk 1.2.23
Zaptel 1.2.19

We installed a Digium B410P card, and we are using 3 ports in te_ptp mode
(it is connected to BRI adapters on Italian ISDN lines, provided by Telecom
Italia; in Cisco language, they should be "isdn switch-type basic-net3").
Configuration strictly complies the documentation/manual.
Phone models used are:
- GrandStream BudgeTone 200;
- GrandStream GXP 2000;
- Thomson ST2030.

All the system is up and working.
However, users often hear an annoying noise during conversations.
Such noise appears suddenly, has a short duration, and happens in particular
when local users talk while people at the other end are silent.
Expecially it happens when remote person stops talking and we (local users,
on Asterisk) start to talk.

Note that people at the remote end do not hear any noise at all:
on their, remote side, communication is felt as perfect.

We run many tests, and we got that this problem is in some way bound to the
echo cancellation. 
If we disable it (echocancel=no in misdn.conf), this annoying non-continuous
noise disappears. But in that case we hear our echo.
If we set a lower rx gain (rxgain=-2 in misdn.conf), noise is is lower and
less annoying, but it does not disappear. And rx gain cannot be set too much
low...

Are there any fixes to this issue?
How can rx gain be fine-tuned in an optimal way?

We know that ztmonitor can be used with zaptel interfaces; but it does not
work with our Digium B410P card.

Are there any ways to avoid such annoying non-continuous noise?

Thank you.
We are available, if you need more information.
Regards,
 Stefano Arata



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Re: [asterisk-users] Slightly OT: SNOM & PoE

[Mark J Elkins]

Anthony Cennami wrote:
> Hello All,
>
> I apologize for the slightly off-topic question, but I'm sure that the
> people best acquainted with the issue would be hanging around here.
>
> Question is, what are people using today to deploy PoE, and more
> importantly, PoE to SNOM phones?
We are running SNOM360's off a Planet POE Switch (FGSW-2620PVS)

This gives us 24 x 10/100 with PoE on all (but the GigE) ports (Power
over the data lines - not on unused copper pairs) and 2 GigE ports
(Copper + SPF). We then plug the Asterisk machine into one of the GigE's
and the other GigE into the rest of the network.
This works well with a centralised wiring closet - now all the phones
are on their own switch - making interference more un-lightly.  The
Switch allows you to look at the power consumption - which technically
means you can judge when there could be a "power problem". Oh - pretty
well priced too.. we paid about US $680 for the switch - which makes it
'low end' cost wise. Currently has no SNMP   :-(   - but has a Linux
friendly web front-end including setup of QoS, Access control and PoE
power consumption.


  POE Ports Status

PortEnable  Power[mW]   Current [mA]
PortEnable  Power[mW]   Current [mA]
1   Enable  9   46  
2   Disable 0   0
3   Disable 0   0   
4   Disable 0   0
5   Disable 0   0   
6   Disable 0   0
7   Disable 0   0   
8   Disable 0   0
9   Disable 0   0   
10  Disable 0   0
11  Disable 0   0   
12  Disable 0   0
13  Enable  426 8   
14  Disable 0   0
15  Enable  233346  
16  Disable 0   0
17  Disable 0   0   
18  Disable 0   0
19  Enable  220645  
20  Enable  499 48
21  Disable 0   0   
22  Disable 0   0
23  Disable 0   0   
24  Disable 0   0


OK - so not many ports are active - but its interesting to see that each
port seems to be consuming different levels of power...

-- 
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Re: [asterisk-users] multiple pbxes, multiple domains, same user ids?

[Anselm Martin Hoffmeister]

Am Mittwoch, den 01.08.2007, 16:32 +0530 schrieb Benjamin Jacob:
> Hello good ppl,
> A couple of questions for multiple pbxes
> 1. Is it possible to support multiple pbxes in one Asterisk box(using 
> contexts, etc.)?
> 2. Can we use the "domain" field in sip.conf to specify the different 
> domains for sip users, having one domain for each pbx?
> 
> I just tried registering two xlites, with different domain names (with 
> the same specified in sip.conf). But, Asterisk maintains the 
> registration of the latest registree!! thats really sad for me .
> 
> Any work around for this one(multiple pbx)?
> I would be zapped and amazed if multiple pbx isn't possible in Asterisk.
> 
> Help anyone?

If "multiple domains" means you want to register SIP phones with the
usernames "sip501" at domain1 and "sip501" at domain2, that in my
experience will not work out this way, because for registered users only
the peer name is relevant (corrections welcome, but it seems like that
to me).

What you could do of course is name the peers reasonably:

customera-501, customerb-501

On the first thought, this is not as elegant, but on the other hand, if
the phone displays the username, it is better than displaying "sip-501".

You would need to have some magic to distinguish between your "domains"
in the dialplan. There is a "static" way of doing it (by setting the
context=blah in the sip peers) or a dynamic way, by giving them all into
the same context, and then do some Asterisk DB magic to make out which
internal partner to reach if "581" is dialled, or which trunk line to
use, or whom to bill calls to. This is absolutely possible, without the
customers noticing.

If you want to support incoming SIP as in sip:[EMAIL PROTECTED], for
different domains, you can specify that in sip.conf. In my experience
(again, I am ready to learn there are better ways) the best working
thing is having a separate domain name for registrations (to get things
easily separated), like "register.yourcompany.domain", with a line
domain=register.yourcompany.domain
and for all further domains have separate contexts, like
"domain-examplecom" and "domain-exampleorg", looking like

domain=example.com,domain-examplecom
domain=example.org,domain-exampleorg

and in extensions.conf, you could go like

[domain-examplecom]
exten => secretary,1,Dial(SIP/customera-505)
exten => bigboss,1,Dial(SIP/customera-500)

[domain-exampleorg]
exten => secretary,1,Dial(SIP/customerb-555)
exten => sales,1,Dial(SIP/customerb-514&SIP/customerb-519)

HTH
Anselm


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Re: [asterisk-users] Asterisk DTMF Tones

[Keshav K.]

I have used Asterisk 1.2 and 1.4 with ATAs and PAP2. There is no issue in that.

For that confrim to your service provider that whihc they accepts, invand or rfc

Keshav

John Meksavan <[EMAIL PROTECTED]> wrote: Asterisk Users,

  I am running Asterisk 1.2.13 on Debian Linux 2.6.18-4-amd64 and having 
problems with DTMF Tones.  I have sip service from Teliax and configure to 
use rfc2833 for dtmfmode.  The problem occurs, when I am using Linksys PAP2T 
phone adapter with a regular analog phone.

Is this an issue with Asterisk? Or the Linksys PAP2Any insights would be 
greatly appreciated.


Best Regards,
John

_
http://liveearth.msn.com


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Regards,
Kesh
" Lets change the future...lets change the world."

   
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Re: [asterisk-users] H.323

[yonoko molomo]

Hi,
I have used h323, oh323 and ooh323.
My experience is that ooh323 does not work properly, i dont recommend it.
I dont know why, but the sound is bad, with sound breaks. I also need
to put some wait (2) functions after the answer( ) or playback( )
functions, it think that asterisk takes some time to stablish the
ooh323 channel (maybe it is due to other reason, i dont know exactly)
but during this time no sound is played, so the first seconds of
conversation or playback are cutted. ooh323 did not work for me at
all.

oh332 worked fine in asterisk 1.2 (i did not try in 1.4, i guess it is fine).

h323 works fine in asterisk 1.4. it is the one i am using now, and i
have no problems with it.

bye now

2007/8/2, Rurouni Alucard <[EMAIL PROTECTED]>:
> Hi there,
>
> I have use the H.323 module that comes with asterisk-addons and i
> consider it (so far) VERY stable for my needs.
> Im talking about 10,000 minutes at month , + or - , and never had a
> crash or something bad about it.
>
> Personally, i recommend it,
>
>
> --
> J. P.
> rakh at slackware-es dot com
>
> bilal ghayyad wrote:
> > Hi List;
> >
> > Did any one tried the H.323 module? How much it is
> > stable and work fine?
> >
> > Regards,
> > 
> > ITS
> > IP Telephony and Contact Center Engineer
> > Eng. Bilal Ghayad
> > Mobile: 00965 9849460
> >
> >
> >
> > Ready
> >  for the edge of your seat?
> > Check out tonight's top picks on Yahoo! TV.
> > http://tv.yahoo.com/
> >
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Re: [asterisk-users] MySQL + Realtime + SIP Registration

[Ove Aursand]

Try to add this line in extconfig.conf:
sippeers => mysql,asterisk,sip_users

Regards,
Ove

Mark Greene wrote:
I have read and followed as much as I can find but I am
missing something. What I want to do is get as much as I can running
from mysql and keep the *.conf files for static things. So I have setup
a SIP users/peers table in a mysql database and I have populated it
with a few peers. I have configured asterisk addons and from the
asterisk CLI I am able to search the sip users / peers tables using the
"realtime load" command. This is after i added "sipusers =>
mysql,asterisk,sip_users" to my extconfig.conf file. However I don't
know what to do to get asterisk to look at that table when a request to
register comes from a sip peer. I understand that sipusers and sip peer
are contradictory but they are all defined as peers.
  
  
  
  

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