Re: [asterisk-users] conferencing help

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Nhadie wrote:
> Hi Matt,
> 
> I tried
> 
> /usr/local/src/zaptel-1.2.22.1# ./zttest -v
> 
> and it just freezes at this.
> 
> Opened pseudo zap interface, measuring accuracy...
> 
> no more outputs,  when i cancelled this is what i got.
> 
> --- Results after 0 passes ---
> Best: 0.00 -- Worst: 100.00 -- Average: 100.00

Yeah that's what I thought.  Am just trying to remember what caused it
though.  Maybe Tzafrir will chime in :)

- --
Kind Regards,

Matt Riddell
Director
___

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Re: [asterisk-users] conferencing help

[Tzafrir Cohen]

On Wed, Jan 09, 2008 at 03:36:06PM +0800, Nhadie wrote:
> Hi Matt,
> 
> I tried
> 
> /usr/local/src/zaptel-1.2.22.1# ./zttest -v
> 
> and it just freezes at this.
> 
> Opened pseudo zap interface, measuring accuracy...
> 
> no more outputs,  when i cancelled this is what i got.
> 
> --- Results after 0 passes ---
> Best: 0.00 -- Worst: 100.00 -- Average: 100.00
> 
> does that mean my zaptel is bad?

Well, yes.

Is ztdummy loaded?

  cat /proc/zaptel/*

What kernel version do you use? What version of Zaptel? What Linux
distribution?

-- 
   Tzafrir Cohen
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Re: [asterisk-users] Set CDR userfield in a realtime dialplan

[Benchev]

On Wednesday 09 January 2008 09:54:59 Yves Räber wrote:
> Hello,
>
> I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have
> some trouble with the CDR userfield that is not changed when using the
> SET command in the realtime dialplan.
> In my dialplan (extensions.conf, the file) I'm setting the userfield
> like this :
>
> exten => s,n,Set(CDR(userfield)="X")
>
> Later, my dialplan switches to the realtime part and this is an extract
> for what is inside :
> ===
> id | context | exten |  priority | app | appdata
> ===
> 12 |  script | s |  n| SET | CDR(userfield)="Y"
> ===
>
> I can show that the command is executed :
> -- Executing Set("SIP/siemens1-081ca290", "CDR(userfield) = Y")
>
> But in my CDR, the old value is saved (X in this case).
Into a database the line exten => s,n,Set(CDR(userfield)="X")
should be enterd as:
context | exten |  priority | app | appdata
means in your case:
your_context| s|n|Set(CDR(userfield)|X

Boyko

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Re: [asterisk-users] Set CDR userfield in a realtime dialplan

[Andrea Cristofanini]

Hello
I'm running the same with no-problem.

CDR(userfield)=INCOMING

We use also a quite nice patch from Matt Riddell
http://bugs.digium.com/view.php?id=9424
that allow to have extra userfield.

CDR(userfield2)=${CODEC-IN}
CDR(userfield3)=${CODEC-OUT}


and so on...

This is quite good for custom report.
/a

Benchev ha scritto:
> On Wednesday 09 January 2008 09:54:59 Yves Räber wrote:
>   
>> Hello,
>>
>> I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have
>> some trouble with the CDR userfield that is not changed when using the
>> SET command in the realtime dialplan.
>> In my dialplan (extensions.conf, the file) I'm setting the userfield
>> like this :
>>
>> exten => s,n,Set(CDR(userfield)="X")
>>
>> Later, my dialplan switches to the realtime part and this is an extract
>> for what is inside :
>> ===
>> id | context | exten |  priority | app | appdata
>> ===
>> 12 |  script | s |  n| SET | CDR(userfield)="Y"
>> ===
>>
>> I can show that the command is executed :
>> -- Executing Set("SIP/siemens1-081ca290", "CDR(userfield) = Y")
>>
>> But in my CDR, the old value is saved (X in this case).
>> 
> Into a database the line exten => s,n,Set(CDR(userfield)="X")
> should be enterd as:
> context | exten |  priority | app | appdata
> means in your case:
> your_context| s|n|Set(CDR(userfield)|X
>
> Boyko
>
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Re: [asterisk-users] [Zaptel] Checking that TDM card works?

[Vincent]

On Tue, 08 Jan 2008 13:43:50 -0500, Jared Smith <[EMAIL PROTECTED]>
wrote:
>I always find that looking at the files that are generated
>under /proc/zaptel is very enlightening as far as showing what the
>zaptel drivers are seeing.

Thanks for the tip.


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Re: [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?

[Steve Langstaff]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Johansson Olle E
> Sent: 09 January 2008 06:50
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to check if a SIP phone 
> isforwardedwithout ringing it ?
> 
> 
> 9 jan 2008 kl. 02.48 skrev Raj Jain:
> 
> > This issue of phone vendors not supporting OPTIONS according to RFC
> > 3261
> > often comes up on this list. Like Kevin Fleming said, an OPTIONS 
> > request is supposed to be responded in the same way as an INVITE. 
> > Almost all SIP phone vendors have construed OPTIONS as some 
> kind of a 
> > keep-alive request, which is wrong.
> Which we do too, by the way. In worst case, maybe Asterisk 
> has set this industry standard.
> 
> OPTIONS is far to heavy in processing on the server side to 
> be used for keep-alives. I'm  starting to see devices that 
> use it for checking capabilities - the proper way. To do this 
> properly, we will have to authenticate the OPTIONs request 
> and match it with the proper peer/ user to get the proper 
> codec settings, ACLs and such.
> 
> Since all versions of Asterisk use OPTIONs for 
> NAT-keepalives, I'm a bit hesitant to fix this. It's a catch 
> 22. I want to do it properly, but then the amount of 
> processing for each OPTIONs request that we receive is going 
> to be a bit too much. Maybe one could ask vendors to add a 
> header to the  OPTIONs packet saying "this is just a keep-alive.  
> Give me a 200 OK without any parsing and be happy, because I 
> don't care about the reply."

It looks like there are two issues rolled into one here...

I hope I'm not "teaching my grandmother to suck eggs" when telling you
this,
Olle, but as I understand it, Asterisk sending OPTIONS to devices as a
NAT
keepalive is a separate issue from devices sending OPTIONS to asterisk
as a
capabilities check...

Received OPTIONS messages should/must be handled as-per the RFCs
(so authentication, matching etc should be done).

If Asterisk wants to send an OPTIONS message just to keep a NAT binding
open then I don't think that it has any obligation to include
authentication
headers if it receives a 401/407 response -  it has received *some*
response,
and that's enough.

If Asterisk wants to send an OPTIONS message to discover peer state
(e.g.
call forward enabled) then obviously it will have to complete any
401/407
handling.

So instead of needing
"a header to the OPTIONs packet saying "this is just a keep-alive""
I think that maybe Asterisk needs to control how it uses OPTIONS,
depending on purpose.
   

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[asterisk-users] remote Snom 360: no voice passing thru

[gincantalupo]

Hi,
I have an Asterisk PBX 1.2.18 connected to a remote Snom 360 (firmware 
ver: 6.5.10), outside my LAN. If I call the Snom, I sometimes cannot 
hear the called party and the called party cannot hear me. When this 
happens, I get the following message from Asterisk CLI:

NOTICE[19164]: chan_sip.c:2012 auto_congest: Auto-congesting 
SIP/3-081da588  -- SIP/3-081da588 is circuit-busy

I searched on internet but found nothing (something similar was an old 
bug but now fixed).

Is there anybody who knows what is causing this strange (random) behaviour?

Consider that:
- there is no firewall between Asterisk and the Snom (Asterisk <-> 
router A <-> internet <-> router B <-> Snom 360)
- I have a Grandstream ATA connected to the same router where the Snom 
is connected but the ATA works good.
- I tried to change cables without success
- rtp debug shows no voice packet is sent between Asterisk and Snom

Any help is appreciated.

Thank you.


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Re: [asterisk-users] Help! channel_find_deadlocked: Avoided initial deadlock for ...

[Steve Davies]

FYI, check the changelog for 1.2.14 to 1.2.25 - IIRC, there are a
significant number of deadlock-fixing updates. There is at least one
related to the code where that error message is displayed.

Regards,
Steve

On 1/9/08, Douglas Garstang <[EMAIL PROTECTED]> wrote:
>
> Replying to myself. :)
> I just noticed the deadlock message still displayed on the console at the
> end of a normal call, so the the deadlock message is not related to the
> early CANCEL
>
>
> - Original Message 
> From: Douglas Garstang <[EMAIL PROTECTED]>
> To: asterisk-users@lists.digium.com
> Sent: Tuesday, January 8, 2008 5:31:12 PM
> Subject: [asterisk-users] Help! channel_find_deadlocked: Avoided initial
> deadlock for ...
>
>
> Hope someone can help.
>
> I have a situation where asterisk is sending a SIP CANCEL message before the
> Dial() timeout has hit. It doesn't always do it.
>
> Normally, we send an INVITE to the ITSP. They respond with a 100 Trying,
> then a 180 Ringing, or 183 Session Progress. It seems to be at this point
> that Asterisk starts the dial timer. Normally, when no more replies have
> been received by the dial timeout, Asterisk sends a CANCEL message. That's
> all fine, and when this happens, this is what appears on the console:
>
> -- Called [EMAIL PROTECTED]
> -- SIP/teleglobe-09879188 is making progress passing it to
> SIP/teleglobe-09876568
> -- Nobody picked up in 4 ms
> -- Executing PlayTones("SIP/teleglobe-09876568",
> "congestion") in new stack
>
> However, when asterisk sends the CANCEL earlier then this, this is what
> appears on the console:
>
> -- SIP/teleglobe-09879188 is making progress passing it to
> SIP/teleglobe-09876568
>   == Spawn extension (default, callback, 7) exited non-zero on
> 'SIP/teleglobe-09876568'
> Jan  9 01:16:34 WARNING[5719]: channel.c:781 channel_find_locked: Avoided
> initial deadlock for '0x97f24d8', 10 retries!
>
> Does anyone know what the deadlock message is all about? It is ocurring
> quite frequently.
> This is Asterisk 1.2.14.
>
> Thanks,
> Doug
>

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Re: [asterisk-users] Set CDR userfield in a realtime dialplan

[Yves Räber]

I answered my own question ... it was a stupid syntax mistake.

CDR(userfield) = "Y"   <--- No spaces allowed at all here

Mea culpa.


On Wed, 2008-01-09 at 08:54 +0100, Yves Räber wrote:
> Hello, 
> 
> I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have
> some trouble with the CDR userfield that is not changed when using the
> SET command in the realtime dialplan.
> 
> In my dialplan (extensions.conf, the file) I'm setting the userfield
> like this :
> 
> exten => s,n,Set(CDR(userfield)="X")
> 
> Later, my dialplan switches to the realtime part and this is an extract
> for what is inside :
> ===
> id | context | exten |  priority | app | appdata 
> ===
> 12 |  script | s |  n| SET | CDR(userfield)="Y"
> ===
> 
> I can show that the command is executed :
> -- Executing Set("SIP/siemens1-081ca290", "CDR(userfield) = Y")
> 
> But in my CDR, the old value is saved (X in this case).
> 
> Does anyone have an idea what's going on here ? Of course I'll send my
> complete config details if needed.
> 
> Thanks
> 
> Yves.
> 
> 
> 
> 
> 
> 
> 
> 
> 
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Re: [asterisk-users] [Zaptel] Checking that TDM card works?

[Tzafrir Cohen]

On Wed, Jan 09, 2008 at 10:42:52AM +0100, Vincent wrote:

> BTW, is there an order when loading modules for a TDM card? The
> OpenVox seems to need zaptel, wctdm, and wcfxo, so I just run this:
> 
> # modprobe zaptel
> # modprobe wctdm
> # modprobe wcfxo

wcfxo is not needed.

Basically all you need is:

  modprobe 

This also pulls all of its dependencies (e.g: zaptel)

  modprobe wctdm

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] [Zaptel] Checking that TDM card works?

[Vincent]

On Tue, 8 Jan 2008 20:29:20 +0200, Tzafrir Cohen
<[EMAIL PROTECTED]> wrote:
>This change is simply due to different versions of Zaptel. Zaptel >=
>1.4.6 prints "to configure" because this message is printed (and has
>always been prinetd) before the configuration is actually applied.

Good to know :-)

>In the Asterisk CLI run:
>
>  core set verbose 3
>
>And then see what happens when a call comes in.
>Basically you miss an action to do after the Verbose line.

But I wasn't getting _anything_ in the CLI. Since I was stuck, I
recompiled Zaptel and Asterisk... and lo-and-behold! I have no idea
why it solved the issue, though.

BTW, is there an order when loading modules for a TDM card? The
OpenVox seems to need zaptel, wctdm, and wcfxo, so I just run this:

# modprobe zaptel
# modprobe wctdm
# modprobe wcfxo
# lsmod

Thanks for the tips on "cat /proc/zaptel/*" and ""asterisk -rx 'zap
show channels'". I'll add them to the list of tools to investigate.


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Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?

[Benny Amorsen]

Olivier <[EMAIL PROTECTED]> writes:

> As using OPTIONS requests main benefit is to non-phone specific, what
> shall we do when most vendors do not comply with RFC ?

Write polite letters to the vendors?


/Benny



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Re: [asterisk-users] conferencing help

[Nhadie]

Hi Tzafrir,

cat /proc/zaptel/*

Span 1: ZTDUMMY/1 "ZTDUMMY/1 1"


Kernel: 2.6.18-5-686 #1 SMP
Zaptel: zaptel-1.2.20.1
OS: Debian GNU/Linux 4.0

i downgraded my zaptel from 1.2.22.1 to 1.2.20.1 but still the same.

thanks again

regards,
nhadie


Tzafrir Cohen wrote:
> On Wed, Jan 09, 2008 at 03:36:06PM +0800, Nhadie wrote:
>> Hi Matt,
>>
>> I tried
>>
>> /usr/local/src/zaptel-1.2.22.1# ./zttest -v
>>
>> and it just freezes at this.
>>
>> Opened pseudo zap interface, measuring accuracy...
>>
>> no more outputs,  when i cancelled this is what i got.
>>
>> --- Results after 0 passes ---
>> Best: 0.00 -- Worst: 100.00 -- Average: 100.00
>>
>> does that mean my zaptel is bad?
> 
> Well, yes.
> 
> Is ztdummy loaded?
> 
>   cat /proc/zaptel/*
> 
> What kernel version do you use? What version of Zaptel? What Linux
> distribution?
> 


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Re: [asterisk-users] What's the best ztdummy?

[Tzafrir Cohen]

On Tue, Jan 08, 2008 at 11:29:08AM -0800, Steve Edwards wrote:
> I have several servers using ztdummy as the timing source, some CentOS 
> 4.x, some CentOS 5.x, some Asterisk 1.2.x, some Asterisk 1.4.x.
> 
> "zap show status" differs between the servers:
> 
> ZTDUMMY/1 (source: Linux26) 1UNCONFIGUR 0  0  0
> ZTDUMMY/1 (source: RTC) 1UNCONFIGUR 0  0  0
> ZTDUMMY/1 1  UNCONFIGUR 0  0  0
> 
> Is one better than the other? What is the best timing source for ztdummy 
> and what does its "status" look like?

The "source" string to ztdummy was only added in latest versions of
ztdummy. So you can't really tell about the last one. 

Linux26 gets an interrupt on every tick of the Linux kernel. It works 
reasonably well when the kernel has HZ=1000 (was the only possible 
value before 2.6.13. And since that kernel that value is configurable, 
and the defualt is 250). Using it on newer kernels normally requires 
using a non-distro kernel.

The problem with RTC is that the kernel gives us 1024 ticks per second,
rather than the 1000 we need. On zaptel 1.2 this resulted in pretty bad
quality. In 1.4 a PLL was introduced to make the rate better, IIRC.

As of kernel 2.6.22 you can use high-resolution timers support in the
kernel, which is better. In that case you'll see the source "HRTimer".

So in short, you're mostly stuck with one supported by your kernel,
unless you want to rebuild it.

For future reference, the README file in the Zaptel sorce distribtion,
or its on-line version at http://zaptel.tzafrir.org.il/ . 

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
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Re: [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?

[Raj Jain]

Olle,

Yes, OPTIONS is too heavy for keep-alives and conflicts with its intended
usage - capability discovery without actually placing a call. The IETF seems
to be finally reaching a conclusion on how to do keep-alives in a
lightweight fashion. These are described in the SIP-outbound draft:

http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-11.txt

Basically, the idea is to use STUN for SIP/UDP and a CRLF packet for
SIP/TCP.

--
Raj


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Johansson Olle E
> Sent: Wednesday, January 09, 2008 1:50 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to check if a SIP phone 
> isforwardedwithout ringing it ?
> 
> 
> 9 jan 2008 kl. 02.48 skrev Raj Jain:
> 
> > This issue of phone vendors not supporting OPTIONS according to RFC
> > 3261
> > often comes up on this list. Like Kevin Fleming said, an OPTIONS 
> > request is supposed to be responded in the same way as an INVITE. 
> > Almost all SIP phone vendors have construed OPTIONS as some 
> kind of a 
> > keep-alive request, which is wrong.
> Which we do too, by the way. In worst case, maybe Asterisk 
> has set this industry standard.
> 
> OPTIONS is far to heavy in processing on the server side to 
> be used for keep-alives. I'm  starting to see devices that 
> use it for checking capabilities - the proper way. To do this 
> properly, we will have to authenticate the OPTIONs request 
> and match it with the proper peer/ user to get the proper 
> codec settings, ACLs and such.
> 
> Since all versions of Asterisk use OPTIONs for 
> NAT-keepalives, I'm a bit hesitant to fix this. It's a catch 
> 22. I want to do it properly, but then the amount of 
> processing for each OPTIONs request that we receive is going 
> to be a bit too much. Maybe one could ask vendors to add a 
> header to the  OPTIONs packet saying "this is just a keep-alive.  
> Give me a 200 OK without any parsing and be happy, because I 
> don't care about the reply."
> 
> Linksys has a setting and use NOTIFY for Keep-alives, which 
> also is a poor solution, but at least something we can just 
> give an error response to without a lot of processing. There 
> was a proposal for PING, but it never got anywhere.
> 
> /O
> 
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Re: [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?

[Raj Jain]

There is something called as answer-mode in SIP. The idea is to allow the
UAC to request the UAS to auto-answer the call. At least in theory, this
could be used to check the status of the phone without ringing it. This is
obviously not an ideal replacement of OPTIONS. Also, this is a new spec so
I'm not sure how many phone vendors support it yet:

http://www.ietf.org/internet-drafts/draft-ietf-sip-answermode-06.txt 
 
--
Raj




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Wednesday, January 09, 2008 1:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to check if a SIP phone
isforwardedwithout ringing it ?


As using OPTIONS requests main benefit is to non-phone specific,
what shall we do when most vendors do not comply with RFC ?


2008/1/9, Raj Jain <[EMAIL PROTECTED] >: 

This issue of phone vendors not supporting OPTIONS according
to RFC 3261
often comes up on this list. Like Kevin Fleming said, an
OPTIONS request is
supposed to be responded in the same way as an INVITE.
Almost all SIP phone
vendors have construed OPTIONS as some kind of a keep-alive
request, which 
is wrong.

Can we ask the phone vendors to play by the book?

--
Raj




From: [EMAIL PROTECTED]
 
[mailto:[EMAIL PROTECTED] On Behalf
Of Olivier
Sent: Tuesday, January 08, 2008 7:50 AM
To: Asterisk Users Mailing List - Non-Commercial
Discussion 
Subject: Re: [asterisk-users] How to check if a SIP
phone is
forwardedwithout ringing it ?


2008/1/7, Kevin P. Fleming <[EMAIL PROTECTED]>: 

Olivier wrote:

> Is there way for an Asterisk server to
check if a sip
phone is forwarded
> without bothering phone's user ?

No. 

> I was thinking of some Alert-Info option
that would let
the phone reply
> with a 302 Moved Temporarily or 182 Queued
message and not
let the phone
> ring or display anything on its screen. 

According to the SIP RFC, a SIP endpoint is
supposed to
respond to an
OPTIONS message the same way that it would
respond to an
INVITE message
with the identical destination, but I've
never seen a phone 
respond to
an OPTIONS message with anything but '200
OK', even when a
redirect
(forward) is in place.


So, the alternative option is to play with html and
use phone 
embedded html server to get this redirection data.

Cheers



--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk
Experience" (TM) 






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Re: [asterisk-users] [Zaptel] Checking that TDM card works?

[Vincent]

On Wed, 9 Jan 2008 12:05:34 +0200, Tzafrir Cohen
<[EMAIL PROTECTED]> wrote:
>wcfxo is not needed.
>
>Basically all you need is:
>
>  modprobe 
>
>This also pulls all of its dependencies (e.g: zaptel)
>
>  modprobe wctdm

Thanks, but on AstLinux, the modules are not unloaded:

===
pbx admin # /etc/init.d/zaptel stop

pbx admin # lsmod
Module  Size  Used by
wctdm  31552  1 
wcfxo  11424  0 
zaptel188604  6 wctdm,wcfxo
hdlc   22528  1 zaptel
syncppp15300  1 hdlc
ppp_generic28692  1 zaptel
===

Why would an init script not remove modules?


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Re: [asterisk-users] [Zaptel] Checking that TDM card works?

[Tzafrir Cohen]

On Wed, Jan 09, 2008 at 12:26:46PM +0100, Vincent wrote:
> On Wed, 9 Jan 2008 12:05:34 +0200, Tzafrir Cohen
> <[EMAIL PROTECTED]> wrote:
> >wcfxo is not needed.
> >
> >Basically all you need is:
> >
> >  modprobe 
> >
> >This also pulls all of its dependencies (e.g: zaptel)
> >
> >  modprobe wctdm
> 
> Thanks, but on AstLinux, the modules are not unloaded:
> 
> ===
> pbx admin # /etc/init.d/zaptel stop
> 
> pbx admin # lsmod
> Module  Size  Used by
> wctdm  31552  1 
> wcfxo  11424  0 
> zaptel188604  6 wctdm,wcfxo
> hdlc   22528  1 zaptel
> syncppp15300  1 hdlc
> ppp_generic28692  1 zaptel
> ===
> 
> Why would an init script not remove modules?

That depends on how the script in astlinux works.
What does it do?

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Linksys SPA-9xx Audio Issues

[Chris Bagnall]

> I have found with a number of clients to who we have installed the LinkSys
> phones, that when you get the input gains to 6, that the phones have a
> tendency to pick up too much background noise. Have you experienced this
> at all?

We have a number of customers out there with SPA-942s and have also found
ourselves having to increase the input gain to 6.

We've not had issues with the background noise, but we have sometimes found
that the input gain introduces echo, even on purely IP-IP routes. It's only
intermittent and not enough to bother the majority of users, but it does
crop up from time to time.

Apart from the config interface, I wonder if there's anything preventing
Linksys from providing other options apart from -6, 0 and 6 ? I think
something like 3 or 4 would probably achieve the desired volume increase
without causing the introduction of echo.

Has anyone tried forcing non-standard figures through a provisioning system?

Regards,

Chris



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Re: [asterisk-users] [Zaptel] Checking that TDM card works?

[Darrick Hartman (lists)]

Vincent wrote:
> On Wed, 9 Jan 2008 12:05:34 +0200, Tzafrir Cohen
> <[EMAIL PROTECTED]> wrote:
>> wcfxo is not needed.
>>
>> Basically all you need is:
>>
>>  modprobe 
>>
>> This also pulls all of its dependencies (e.g: zaptel)
>>
>>  modprobe wctdm
> 
> Thanks, but on AstLinux, the modules are not unloaded:
> 
> ===
> pbx admin # /etc/init.d/zaptel stop
> 
> pbx admin # lsmod
> Module  Size  Used by
> wctdm  31552  1 
> wcfxo  11424  0 
> zaptel188604  6 wctdm,wcfxo
> hdlc   22528  1 zaptel
> syncppp15300  1 hdlc
> ppp_generic28692  1 zaptel
> ===
> 
> Why would an init script not remove modules?

Vincent,

Come on over to the astlinux mailing list (on our sourceforge page).  It 
will be easier to handle any Astlinux specific questions over there.

But look in your /etc/rc.conf file for the ZAPMODS variable.  You should 
have that variable set to:

ZAPMODS="wctdm"

Beyond that, as long as Asterisk is not running, issuing service zaptel 
stop should remove all zaptel related modules.

Darrick
-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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[asterisk-users] Newbie: confusion with the new FXO/FXS card

[Vytenis Sabaliauskas]

   Hello everyone,

I'm trying to set up a Asterisk server. I have two cards - one is an 
BeroNet BN2S0 with two ISDN lines (4 channels):

http://www.adcomtec.com/webstore/beronet_bn2s0.php?cat=90

and a Rhino R8FXX with one FXO module and two FXS:

http://www.voipsupply.com/product_info.php?products_id=2940

I would like to set up an Asterisk server with 8 phones, which will 
share the phone numbers via ISDN. Sorry if i'm not writing this very 
clearly, since my telecomunication skills are appaling (I'm used to 
linux though).

As documentation states, Rhino R8FXX can work as FXO or as FXS depending 
on the modules installed. How about my situation? At first, I would like 
to set up a simpliest thing - to make phone do anything (play WAV, echo, 
etc.) Does anyone show me the road how to do it? A link to some manual 
or such...

I tried googling, but have found no results (or didn't knew, that the 
info was good for me). This is my first Asterisk server and I'm very 
interested in bringing it up.

Thanks. Ow, and sory for laminess :)

--
V.


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Re: [asterisk-users] [Zaptel] Checking that TDM card works?

[Vincent]

On Wed, 09 Jan 2008 06:01:32 -0600, "Darrick Hartman (lists)"
<[EMAIL PROTECTED]> wrote:
>But look in your /etc/rc.conf file for the ZAPMODS variable.  You should 
>have that variable set to:
>
>ZAPMODS="wctdm"

Yes indeed:

#ZAPMODS="wctdm"

Should I add this module here, or in rc.modules?

Are we positive that wcfxo is not needed in addition to wctdm and
zaptel?

>Beyond that, as long as Asterisk is not running, issuing service zaptel 
>stop should remove all zaptel related modules.

Thanks, but it doesn't seem to unload the modules:

==
# /etc/init.d/zaptel stop

# lsmod
Module  Size  Used by
wctdm  31552  1 
wcfxo  11424  0 
binfmt_misc11784  1 
zaptel188604  6 wctdm,wcfxo
hdlc   22528  1 zaptel
syncppp15300  1 hdlc
ppp_generic28692  1 zaptel
slhc6784  1 ppp_generic

# asterisk -r
pbx*CLI> stop now
Disconnected from Asterisk server

# ps
  PID  Uid VmSize Stat Command
(snip : no trace of Asterisk)
 1327 root368 R   ps 

# /etc/init.d/zaptel stop

# lsmod
Module  Size  Used by
wctdm  31552  0 
wcfxo  11424  0 
binfmt_misc11784  1 
zaptel188604  2 wctdm,wcfxo
hdlc   22528  1 zaptel
syncppp15300  1 hdlc
ppp_generic28692  1 zaptel
slhc6784  1 ppp_generic
==

I guess the zaptel script doesn't remove them, and I need to use rmmod
manually?


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Re: [asterisk-users] How to check if a SIP phone isforwardedwithout ringing it ?

[Johansson Olle E]

9 jan 2008 kl. 10.46 skrev Steve Langstaff:

>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of
>> Johansson Olle E
>> Sent: 09 January 2008 06:50
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] How to check if a SIP phone
>> isforwardedwithout ringing it ?
>>
>>
>> 9 jan 2008 kl. 02.48 skrev Raj Jain:
>>
>>> This issue of phone vendors not supporting OPTIONS according to RFC
>>> 3261
>>> often comes up on this list. Like Kevin Fleming said, an OPTIONS
>>> request is supposed to be responded in the same way as an INVITE.
>>> Almost all SIP phone vendors have construed OPTIONS as some
>> kind of a
>>> keep-alive request, which is wrong.
>> Which we do too, by the way. In worst case, maybe Asterisk
>> has set this industry standard.
>>
>> OPTIONS is far to heavy in processing on the server side to
>> be used for keep-alives. I'm  starting to see devices that
>> use it for checking capabilities - the proper way. To do this
>> properly, we will have to authenticate the OPTIONs request
>> and match it with the proper peer/ user to get the proper
>> codec settings, ACLs and such.
>>
>> Since all versions of Asterisk use OPTIONs for
>> NAT-keepalives, I'm a bit hesitant to fix this. It's a catch
>> 22. I want to do it properly, but then the amount of
>> processing for each OPTIONs request that we receive is going
>> to be a bit too much. Maybe one could ask vendors to add a
>> header to the  OPTIONs packet saying "this is just a keep-alive.
>> Give me a 200 OK without any parsing and be happy, because I
>> don't care about the reply."
>
> It looks like there are two issues rolled into one here...
>
> I hope I'm not "teaching my grandmother to suck eggs" when telling you
> this,
> Olle, but as I understand it, Asterisk sending OPTIONS to devices as a
> NAT
> keepalive is a separate issue from devices sending OPTIONS to asterisk
> as a
> capabilities check...
>
> Received OPTIONS messages should/must be handled as-per the RFCs
> (so authentication, matching etc should be done).
>
> If Asterisk wants to send an OPTIONS message just to keep a NAT  
> binding
> open then I don't think that it has any obligation to include
> authentication
> headers if it receives a 401/407 response -  it has received *some*
> response,
> and that's enough.
>
> If Asterisk wants to send an OPTIONS message to discover peer state
> (e.g.
> call forward enabled) then obviously it will have to complete any
> 401/407
> handling.
>
> So instead of needing
>"a header to the OPTIONs packet saying "this is just a keep-alive""
> I think that maybe Asterisk needs to control how it uses OPTIONS,
> depending on purpose.

The issue here is that it requires a lot of extra processing when  
RECEIVING
the OPTION message if we want to do it right. Sending is not an issue.

If we want to handle OPTIONs correctly we need to match with the peer/ 
user
list and then set up a complete dialog with all the options from the  
peer/user and
then reply...


Where's my eggs?

/O

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Re: [asterisk-users] What's the best ztdummy?

[Thomas Stein]

On Wednesday 09 January 2008, Tzafrir Cohen wrote:

> As of kernel 2.6.22 you can use high-resolution timers support in the
> kernel, which is better. In that case you'll see the source "HRTimer".

Is this still an option with kernel 2.6.23? I didn't find that option in my 
current kernel sources.

/usr/src/linux-2.6.23.12 $ cat .config | grep CONFIG_HIGH_RES_TIMERS
/usr/src/linux-2.6.23.12 $

t.
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Re: [asterisk-users] Newbie: confusion with the new FXO/FXS card

[Glenn Cobb]

Go here

www.voip-info.org

and read alot.  Almost everything you need to know (or a link to it) can
be found through there.

Seriously, its your best starting point 

regards,
Glenn

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Vytenis Sabaliauskas
> Sent: Wednesday, January 09, 2008 7:41 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Newbie: confusion with the new FXO/FXS card
> 
>Hello everyone,
> 
> I'm trying to set up a Asterisk server. I have two cards 
> - one is an BeroNet BN2S0 with two ISDN lines (4 channels):
> 
> http://www.adcomtec.com/webstore/beronet_bn2s0.php?cat=90
> 
> and a Rhino R8FXX with one FXO module and two FXS:
> 
> http://www.voipsupply.com/product_info.php?products_id=2940
> 
> I would like to set up an Asterisk server with 8 phones, 
> which will share the phone numbers via ISDN. Sorry if i'm not 
> writing this very clearly, since my telecomunication skills 
> are appaling (I'm used to linux though).
> 
> As documentation states, Rhino R8FXX can work as FXO or as 
> FXS depending on the modules installed. How about my 
> situation? At first, I would like to set up a simpliest thing 
> - to make phone do anything (play WAV, echo,
> etc.) Does anyone show me the road how to do it? A link to 
> some manual or such...
> 
> I tried googling, but have found no results (or didn't knew, 
> that the info was good for me). This is my first Asterisk 
> server and I'm very interested in bringing it up.
> 
> Thanks. Ow, and sory for laminess :)
> 
> --
> V.
> 
> 
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Re: [asterisk-users] How to check if a SIP phone isforwardedwithoutringing it ?

[Steve Langstaff]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Johansson Olle E
> Sent: 09 January 2008 14:11
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to check if a SIP phone 
> isforwardedwithoutringing it ?
> 
> 
> 9 jan 2008 kl. 10.46 skrev Steve Langstaff:
> 
> >> -Original Message-
> >> From: [EMAIL PROTECTED]
> >> [mailto:[EMAIL PROTECTED] On Behalf Of 
> >> Johansson Olle E
> >> Sent: 09 January 2008 06:50
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] How to check if a SIP phone 
> >> isforwardedwithout ringing it ?
> >>
> >>
> >> 9 jan 2008 kl. 02.48 skrev Raj Jain:
> >>
> >>> This issue of phone vendors not supporting OPTIONS 
> according to RFC
> >>> 3261
> >>> often comes up on this list. Like Kevin Fleming said, an OPTIONS 
> >>> request is supposed to be responded in the same way as an INVITE.
> >>> Almost all SIP phone vendors have construed OPTIONS as some
> >> kind of a
> >>> keep-alive request, which is wrong.
> >> Which we do too, by the way. In worst case, maybe Asterisk has set 
> >> this industry standard.
> >>
> >> OPTIONS is far to heavy in processing on the server side 
> to be used 
> >> for keep-alives. I'm  starting to see devices that use it for 
> >> checking capabilities - the proper way. To do this 
> properly, we will 
> >> have to authenticate the OPTIONs request and match it with 
> the proper 
> >> peer/ user to get the proper codec settings, ACLs and such.
> >>
> >> Since all versions of Asterisk use OPTIONs for 
> NAT-keepalives, I'm a 
> >> bit hesitant to fix this. It's a catch 22. I want to do it 
> properly, 
> >> but then the amount of processing for each OPTIONs request that we 
> >> receive is going to be a bit too much. Maybe one could ask 
> vendors to 
> >> add a header to the  OPTIONs packet saying "this is just a 
> >> keep-alive.
> >> Give me a 200 OK without any parsing and be happy, because I don't 
> >> care about the reply."
> >
> > It looks like there are two issues rolled into one here...
> >
> > I hope I'm not "teaching my grandmother to suck eggs" when 
> telling you 
> > this, Olle, but as I understand it, Asterisk sending OPTIONS to 
> > devices as a NAT keepalive is a separate issue from devices sending 
> > OPTIONS to asterisk as a capabilities check...
> >
> > Received OPTIONS messages should/must be handled as-per the 
> RFCs (so 
> > authentication, matching etc should be done).
> >
> > If Asterisk wants to send an OPTIONS message just to keep a NAT 
> > binding open then I don't think that it has any obligation 
> to include 
> > authentication headers if it receives a 401/407 response -  it has 
> > received *some* response, and that's enough.
> >
> > If Asterisk wants to send an OPTIONS message to discover peer state 
> > (e.g.
> > call forward enabled) then obviously it will have to complete any
> > 401/407
> > handling.
> >
> > So instead of needing
> >"a header to the OPTIONs packet saying "this is just a 
> keep-alive""
> > I think that maybe Asterisk needs to control how it uses OPTIONS, 
> > depending on purpose.
> 
> The issue here is that it requires a lot of extra processing 
> when RECEIVING the OPTION message if we want to do it right. 
> Sending is not an issue.
> 
> If we want to handle OPTIONs correctly we need to match with 
> the peer/ user list and then set up a complete dialog with 
> all the options from the peer/user and then reply...

Right.

I agree that RECEIVING an OPTION message on the Asterisk server may
require a lot of extra processing.

I agree that sending an OPTION message from the Asterisk server could
well have a low processing load.

The original poster wanted to use OPTIONS sent FROM the Asterisk server
to query the phone state, so I don't think your concerns about receive
processing come into the picture.



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Re: [asterisk-users] [Zaptel] Checking that TDM card works?

[Darrick Hartman (lists)]

Vincent wrote:
> On Wed, 09 Jan 2008 06:01:32 -0600, "Darrick Hartman (lists)"
> <[EMAIL PROTECTED]> wrote:
>> But look in your /etc/rc.conf file for the ZAPMODS variable.  You should 
>> have that variable set to:
>>
>> ZAPMODS="wctdm"
> 
> Yes indeed:
> 
> #ZAPMODS="wctdm"
> 
> Should I add this module here, or in rc.modules?

Uncomment that if you expect it to work.  The module should be listed in 
ZAPMODS not in rc.modules.

> Are we positive that wcfxo is not needed in addition to wctdm and
> zaptel?

Yes we're sure.

>> Beyond that, as long as Asterisk is not running, issuing service zaptel 
>> stop should remove all zaptel related modules.
> 
> Thanks, but it doesn't seem to unload the modules:
> 
> ==
> # /etc/init.d/zaptel stop
> 
> # lsmod
> Module  Size  Used by
> wctdm  31552  1 
> wcfxo  11424  0 
> binfmt_misc11784  1 
> zaptel188604  6 wctdm,wcfxo
> hdlc   22528  1 zaptel
> syncppp15300  1 hdlc
> ppp_generic28692  1 zaptel
> slhc6784  1 ppp_generic
> 
> # asterisk -r
> pbx*CLI> stop now
> Disconnected from Asterisk server
> 
> # ps
>   PID  Uid VmSize Stat Command
> (snip : no trace of Asterisk)
>  1327 root368 R   ps 
> 
> # /etc/init.d/zaptel stop
> 
> # lsmod
> Module  Size  Used by
> wctdm  31552  0 
> wcfxo  11424  0 
> binfmt_misc11784  1 
> zaptel188604  2 wctdm,wcfxo
> hdlc   22528  1 zaptel
> syncppp15300  1 hdlc
> ppp_generic28692  1 zaptel
> slhc6784  1 ppp_generic
> ==
> 
> I guess the zaptel script doesn't remove them, and I need to use rmmod
> manually?
> 

Since you have ZAPMODS commented out, the zaptel init script doesn't 
know which modules it should be using.  I can assure you that this 
script does work properly if you have the configuration set correctly.

Vincent, all of this is really Astlinux specific and would be better 
handled on that list instead.

Darrick
-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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Re: [asterisk-users] [asterisk-dev] MixMonitor doesn't work right with SIP and Zap/Flash transfers

[Atis Lezdins]

Vinicius Fontes wrote:
> Hey guys, I don't know if this is the right place to ask this. I was  
> thinking about reporting a bug, but maybe it's better to sort out if  
> this is really a bug or just me being lame.
> 
> I want to record *every* call in my Asterisk box, so I use the  
> MixMonitor() application like this is my extensions.conf:
> 
> exten => _0X.,1,Answer()
> exten => _0X.,n,MixMonitor(${CALLERID(num)}-${STRFTIME($ 
> {EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav)
> exten => _0X.,n,Dial(IAX2/pabx-canall/${EXTEN},60,tT)
> 
> exten => _2XX,1,Answer() exten => _2XX,n,MixMonitor(${CALLERID(num)}-$ 
> {STRFTIME(${EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav)  
> exten => _2XX,n,Dial(SIP/${EXTEN},60,tT)
> 
> The scenario is as following:
> 
> 1) 201 asks operator for an external call, hangs up. The audio file is  
> stored correctly. From the CLI:
> 
> [Jan 8 16:20:19] -- Executing [EMAIL PROTECTED]:1] Answer("SIP/ 
> 201-081d8740", "") in new stack
> [Jan 8 16:20:19] -- Executing [EMAIL PROTECTED]:2] MixMonitor("SIP/ 
> 201-081d8740", "201-2008-01-08-16-20-19-200.wav") in new stack
> [Jan 8 16:20:19] -- Executing [EMAIL PROTECTED]:3] Dial("SIP/201-081d8740",  
> "SIP/200|60|tT") in new stack
> [Jan 8 16:20:19] == Begin MixMonitor Recording SIP/201-081d8740
> [Jan 8 16:20:19] -- Called 200
> [Jan 8 16:20:19] -- SIP/200-081fac90 is ringing
> [Jan 8 16:20:23] -- SIP/200-081fac90 answered SIP/201-081d8740
> [Jan 8 16:20:27] == Spawn extension (default, 200, 3) exited non-zero  
> on 'SIP/201-081d8740'
> [Jan 8 16:20:27] == End MixMonitor Recording SIP/201-081d8740
> 
> 
> 
> 
> 2) 200 dials to the PSTN. So far so good.
> 
> [Jan 8 16:20:35] -- Executing [EMAIL PROTECTED]:1] Answer("SIP/ 
> 200-081d8740", "") in new stack
> [Jan 8 16:20:35] -- Executing [EMAIL PROTECTED]:2] MixMonitor("SIP/ 
> 200-081d8740", "200-2008-01-08-16-20-35-021047020.wav") in new stack
> [Jan 8 16:20:35] -- Executing [EMAIL PROTECTED]:3] Dial("SIP/ 
> 200-081d8740", "IAX2/pabx-canall/021047020|60|tT") in new stack
> [Jan 8 16:20:35] == Begin MixMonitor Recording SIP/200-081d8740
> [Jan 8 16:20:35] -- Called pabx-canall/021047020
> [Jan 8 16:20:35] -- Call accepted by 200.248.136.140 (format alaw)
> [Jan 8 16:20:35] -- Format for call is alaw [Jan 8 16:20:35] -- IAX2/ 
> pabx-canall-16384 answered SIP/200-081d8740
> 
> 
> 
> 
> 3) Extension 200 is a Polycom SoundPoint 301 IP phone. It presses the  
> Transfer button, putting 021047020 in hold and dialing to 201 who  
> answers the call:
> 
> [Jan 8 16:20:45] -- Started music on hold, class 'default', on IAX2/ 
> pabx-canall-16384
> [Jan 8 16:20:51] -- Executing [EMAIL PROTECTED]:1] Answer("SIP/ 
> 200-081fac90", "") in new stack
> [Jan 8 16:20:51] -- Executing [EMAIL PROTECTED]:2] MixMonitor("SIP/ 
> 200-081fac90", "200-2008-01-08-16-20-51-201.wav") in new stack
> [Jan 8 16:20:51] -- Executing [EMAIL PROTECTED]:3] Dial("SIP/200-081fac90",  
> "SIP/201|60|tT") in new stack
> [Jan 8 16:20:51] -- Called 201
> [Jan 8 16:20:51] == Begin MixMonitor Recording SIP/200-081fac90
> [Jan 8 16:20:51] -- SIP/201-081edf80 is ringing
> [Jan 8 16:20:54] -- SIP/201-081edf80 answered SIP/200-081fac90
> 
> 
> 
> 
> 4) The operator says "here's your call" to 201 and presses Transfer on  
> the phone once more. The call is transferred correctly, but:
> [Jan 8 16:20:57] -- Stopped music on hold on IAX2/pabx-canall-16384
> [Jan 8 16:20:57] == Spawn extension (default, 021047020, 3) exited non- 
> zero on 'SIP/200-081d8740'
> [Jan 8 16:20:57] == End MixMonitor Recording SIP/200-081d8740
> [Jan 8 16:20:57] == End MixMonitor Recording SIP/200-081fac90
> 
> 
> Notice that all the MixMonitor processes stopped!
> 
> 
> 
> 5) 201 finally hangs up the phone:
> 
> [Jan 8 16:21:45] == Spawn extension (default, 201, 3) exited non-zero  
> on 'IAX2/pabx-canall-16384'
> [Jan 8 16:21:45] -- Hungup 'IAX2/pabx-canall-16384'
> 
> 
> 
> So, all the audio regarding the important part -- the call to the PSTN  
> itself -- is simply lost.
> 
> I noticed that if I use Asterisk's built-in transfer features (atxfer,  
> blindxfer) everything works fine. Too bad the users are so used to  
> that Transfer button. I tried it using FXS channels and the FLASH  
> button on the phone, same results.
> 
> Is there any workaround for this? I'm running these from a separate  
> box so any procediment you guys could suggest will be tried as it is  
> not in production. I'm also willing to give you any information needed.

Make sure you have "canreinvite=no" for peers, that will ensure that RTP 
is always passed trough asterisk. Now your Polycom might send audio 
directly to other phone.

Btw, asterisk-dev is for development discussions, but this is 
configuration problem. If unsure, you should write to asterisk-users 
first (cross-posted there)

Regards,
Atis

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[asterisk-users] Zaptel FXS Cards - Station Distance

[Tim Nelson]

Hello! In the near future, I'll be deploying an asterisk system that will have 
(in addition to many SIP handsets) four FXS handsets. Is there a known 
limitation to the length of cabling that can be used between a 
Digium/Openvox/Sangoma FXS card and the end station? I understand the card will 
need a molex power connector plugged in to provide extra voltage. With 
traditional phone systems, I have not seen any distance limitations. I would 
just like to verify that the FXS ports are able to provide sufficient power for 
longer runs. Thank you!!!

Tim Nelson
Systems/Network Support
Rockbochs Inc.


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[asterisk-users] Busy notification with call limiting by GROUP_COUNT()

[Peter Galiovsky]

Hello all,

I was wondering what will be the "proper" way to manage BUSY state 
notification in presence once call-limit, incominglimit and all those 
settings are gone.

I'm using GROUP_COUNT for call limiting in Asterisk 1.4.13 but I have no 
idea how to set up the settings needed for BUSY notification to work as 
I want it to.

Basically, I want to disable call waiting (this is done by the 
GROUP_COUNT() check in dialplan) but don't want to limit the number of 
outgoing calls per user at all. I want the user to be presented as busy 
if he has at least one call active, be it incoming or outgoing. How 
should I set things up to achieve this?

Thanks in advance. Best,
Peter

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Re: [asterisk-users] Zaptel FXS Cards - Station Distance

[Kevin P. Fleming]

Tim Nelson wrote:
> Hello! In the near future, I'll be deploying an asterisk system that will 
> have (in addition to many SIP handsets) four FXS handsets. Is there a known 
> limitation to the length of cabling that can be used between a 
> Digium/Openvox/Sangoma FXS card and the end station? I understand the card 
> will need a molex power connector plugged in to provide extra voltage. With 
> traditional phone systems, I have not seen any distance limitations. I would 
> just like to verify that the FXS ports are able to provide sufficient power 
> for longer runs. Thank you!!!

You don't quote any numbers here, and 'longer' is not anything we can
relate to.

However, I can say that the FXS line drivers on Digium analog modules
should be able to handle 1000' runs or so, but not more than that. This
should be equivalent to what any standard PBX with FXS ports can do,
unless it has special 'long line' drivers.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] Zaptel FXS Cards - Station Distance

[Tilghman Lesher]

On Wednesday 09 January 2008 09:15:08 Tim Nelson wrote:
> Hello! In the near future, I'll be deploying an asterisk system that will
> have (in addition to many SIP handsets) four FXS handsets. Is there a known
> limitation to the length of cabling that can be used between a
> Digium/Openvox/Sangoma FXS card and the end station? I understand the card
> will need a molex power connector plugged in to provide extra voltage. With
> traditional phone systems, I have not seen any distance limitations. I
> would just like to verify that the FXS ports are able to provide sufficient
> power for longer runs. Thank you!!!

There are limitations, even with traditional systems.  It's usually around
1,000 feet, but the basic problem is that DC signalling degrades over
distance.  Incidentally, this is why power lines are provided with AC current.
The TDM cards should be able to provide reliable signalling over the same
distance as traditional telephony lines, as they are sending with the same
power specifications.

-- 
Tilghman

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Re: [asterisk-users] Zaptel FXS Cards - Station Distance

[John Novack]

Kevin P. Fleming wrote:
> Tim Nelson wrote:
>   
>> Hello! In the near future, I'll be deploying an asterisk system that will 
>> have (in addition to many SIP handsets) four FXS handsets. Is there a known 
>> limitation to the length of cabling that can be used between a 
>> Digium/Openvox/Sangoma FXS card and the end station? I understand the card 
>> will need a molex power connector plugged in to provide extra voltage. With 
>> traditional phone systems, I have not seen any distance limitations. I would 
>> just like to verify that the FXS ports are able to provide sufficient power 
>> for longer runs. Thank you!!!
>> 
>
> You don't quote any numbers here, and 'longer' is not anything we can
> relate to.
>
> However, I can say that the FXS line drivers on Digium analog modules
> should be able to handle 1000' runs or so, but not more than that. This
> should be equivalent to what any standard PBX with FXS ports can do,
> unless it has special 'long line' drivers.
>   
In reality many PBX analog circuits using DTMF only can go quite a bit 
longer. The  1K foot limitation with pulse dial is more realistic, due 
to cable capacitance mis shaping of pulses. DTMF only is limited by the 
line resistance limiting current to 20-30 Ma required to operate the 
tone pad. More modern ( cheap Chinese ) phones often will operate with less.
With lower than normal 48VDC supplies, the distance grows ever shorter.
So, what is the distance you require?
What is the size of the wire in the cable?

Dial long line circuits ARE available, and advised along with special 
protection if the stations will be off premises.

John Novack

-- 
Dog is my co-pilot


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[asterisk-users] Broken calls

[mccoy silva]

  Hello Fellows!

  I have the following problem: When peers are in a conversation the call
broken, it not happen every time, but 60% of calls. :(
  In Asterisk log I got this message:  rtp.c: RTCP Read too short.
  I'm using a TDM2400 with Asterisk 1.4.14 and zaptel 1.4.7 (Debian Etch)

  Thank you for any help.

  Regards,

  McCoy
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Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?

[Olivier]

2008/1/9, Benny Amorsen <[EMAIL PROTECTED]>:
>
> Olivier <[EMAIL PROTECTED]> writes:
>
> > As using OPTIONS requests main benefit is to non-phone specific, what
> > shall we do when most vendors do not comply with RFC ?
>
> Write polite letters to the vendors?


To get a polite "go to hell !" in return ?  ;-)

/Benny
>
>
>
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[asterisk-users] WaitExten and Macros

[Tony Plack]

I am trying to use a WaitExten in a Macro, and I am finding that the extension 
which is pressed ends up in context of the calling context and not in the Macro.

How do you do a WaitExten in a Macro?

Tony Plack

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[asterisk-users] WaitExten and Macros

[Tony Plack]

I am trying to use a WaitExten in a Macro, and I am finding that the extension 
which is pressed ends up in context of the calling context and not in the Macro.

How do you do a WaitExten in a Macro?

Tony Plack

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Re: [asterisk-users] How to check if a SIP phone isforwardedwithoutringing it ?

[Olivier]

2008/1/9, Steve Langstaff <[EMAIL PROTECTED]>:
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Johansson Olle E
> > Sent: 09 January 2008 14:11
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] How to check if a SIP phone
> > isforwardedwithoutringing it ?
> >
> >
> > 9 jan 2008 kl. 10.46 skrev Steve Langstaff:
> >
> > >> -Original Message-
> > >> From: [EMAIL PROTECTED]
> > >> [mailto:[EMAIL PROTECTED] On Behalf Of
> > >> Johansson Olle E
> > >> Sent: 09 January 2008 06:50
> > >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> > >> Subject: Re: [asterisk-users] How to check if a SIP phone
> > >> isforwardedwithout ringing it ?
> > >>
> > >>
> > >> 9 jan 2008 kl. 02.48 skrev Raj Jain:
> > >>
> > >>> This issue of phone vendors not supporting OPTIONS
> > according to RFC
> > >>> 3261
> > >>> often comes up on this list. Like Kevin Fleming said, an OPTIONS
> > >>> request is supposed to be responded in the same way as an INVITE.
> > >>> Almost all SIP phone vendors have construed OPTIONS as some
> > >> kind of a
> > >>> keep-alive request, which is wrong.
> > >> Which we do too, by the way. In worst case, maybe Asterisk has set
> > >> this industry standard.
> > >>
> > >> OPTIONS is far to heavy in processing on the server side
> > to be used
> > >> for keep-alives. I'm  starting to see devices that use it for
> > >> checking capabilities - the proper way. To do this
> > properly, we will
> > >> have to authenticate the OPTIONs request and match it with
> > the proper
> > >> peer/ user to get the proper codec settings, ACLs and such.
> > >>
> > >> Since all versions of Asterisk use OPTIONs for
> > NAT-keepalives, I'm a
> > >> bit hesitant to fix this. It's a catch 22. I want to do it
> > properly,
> > >> but then the amount of processing for each OPTIONs request that we
> > >> receive is going to be a bit too much. Maybe one could ask
> > vendors to
> > >> add a header to the  OPTIONs packet saying "this is just a
> > >> keep-alive.
> > >> Give me a 200 OK without any parsing and be happy, because I don't
> > >> care about the reply."
> > >
> > > It looks like there are two issues rolled into one here...
> > >
> > > I hope I'm not "teaching my grandmother to suck eggs" when
> > telling you
> > > this, Olle, but as I understand it, Asterisk sending OPTIONS to
> > > devices as a NAT keepalive is a separate issue from devices sending
> > > OPTIONS to asterisk as a capabilities check...
> > >
> > > Received OPTIONS messages should/must be handled as-per the
> > RFCs (so
> > > authentication, matching etc should be done).
> > >
> > > If Asterisk wants to send an OPTIONS message just to keep a NAT
> > > binding open then I don't think that it has any obligation
> > to include
> > > authentication headers if it receives a 401/407 response -  it has
> > > received *some* response, and that's enough.
> > >
> > > If Asterisk wants to send an OPTIONS message to discover peer state
> > > (e.g.
> > > call forward enabled) then obviously it will have to complete any
> > > 401/407
> > > handling.
> > >
> > > So instead of needing
> > >"a header to the OPTIONs packet saying "this is just a
> > keep-alive""
> > > I think that maybe Asterisk needs to control how it uses OPTIONS,
> > > depending on purpose.
> >
> > The issue here is that it requires a lot of extra processing
> > when RECEIVING the OPTION message if we want to do it right.
> > Sending is not an issue.
> >
> > If we want to handle OPTIONs correctly we need to match with
> > the peer/ user list and then set up a complete dialog with
> > all the options from the peer/user and then reply...
>
> Right.
>
> I agree that RECEIVING an OPTION message on the Asterisk server may
> require a lot of extra processing.
>
> I agree that sending an OPTION message from the Asterisk server could
> well have a low processing load.
>
> The original poster wanted to use OPTIONS sent FROM the Asterisk server
> to query the phone state, so I don't think your concerns about receive
> processing come into the picture.


Yes, obviously,  there are 2 different issues  :
1. how to query phone status
2. how to keep NAT alives

A. For phone status queries, a list of phone vendors (at least one)
supporting this usage or future SIPit tests would help to motivate other
vendors to implement desired SIP OPTIONS behaviour along today's one.
Maybe an email sent by Digium to a list of SIP phone vendors and asking for
guidance would be a practical mean to get replies.

B. As keeping NAT alive is a basic need, maybe we should ask SIP PING method
authors or sponsors for advice on this (contact address in enclosed in
http://www.cornfed.com/ping.html ) ?
Either SIP PING (or something equivalent) should be ratified or SIP OPTIONS
should be "clarified".


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Re: [asterisk-users] WaitExten and Macros

[Tilghman Lesher]

On Wednesday 09 January 2008 12:09:29 Tony Plack wrote:
> I am trying to use a WaitExten in a Macro, and I am finding that the
> extension which is pressed ends up in context of the calling context and
> not in the Macro.
>
> How do you do a WaitExten in a Macro?

You cannot.

-- 
Tilghman

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[asterisk-users] Intercom & Paging with Polycoms

[Rob Schall]

I've been able to page to a specific phone (intercom type of thing), but
I'd like to have a macro or agi that pages all phones but first checks
if their on the phone. It looked like there used to be a pageall.agi
type of script on the wiki, but that link isn't valid anymore. Does
anyone have that script, or something else that would work? I would just
do SIP/1000&SIP/1001, but there are about 60 phones involved.

Rob

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Re: [asterisk-users] WaitExten and Macros

[Rob Schall]

You may just want a "Read" if you know how many numbers you're looking for.

Rob


Tony Plack wrote:
> I am trying to use a WaitExten in a Macro, and I am finding that the 
> extension which is pressed ends up in context of the calling context and not 
> in the Macro.
>
> How do you do a WaitExten in a Macro?
>
> Tony Plack
>
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Re: [asterisk-users] WaitExten and Macros

[Jared Smith]

On Wed, 2008-01-09 at 12:09 -0600, Tony Plack wrote:
> I am trying to use a WaitExten in a Macro

Bad idea... as far as I know, neither Background nor WaitExten work
correctly inside of macros.  I'd suggest you use the Read() application
instead.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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[asterisk-users] Subscriptions, Firewalls, 489 "Bad Event" and Bug 7608

[Tony Plack]

I am running 1.4 svn-r95946 and have an error 

-- Got SIP response 489 "Bad event" back from 1.2.3.4

The problem is that this error is being generated by Asterisk trying to send a 
NOTIFY to a phone behind a NAT firewall that does not exist anymore.  The phone 
was physically decommissioned 3 weeks ago and 4 restarts ago (couple versions 
up on Asterisk during that time), but is still configured in real-time 
sippeers.  We intend to bring this extension back so we have not removed it 
from sippeers, but there is no hardware today to connect using it.

The old phone:address:port was 5003:192.168.168.77:5060 and the firewall is 
showing that this address is being translated to address 
5004:192.168.168.75:5060.

So I assumed that Asterisk still had a subscription but it does not show up in 
sip show subscriptions at all.

SIP debug shows that Asterisk is generating this packet, but I cannot find out 
how to remove it.  I assume this is stuck in the astdb.

I found Bug 7608 but the version of Grandstream does not seem to make a 
difference.  Why is Asterisk keep sending out these NOTIFY messages on a 
non-subscribed phone?  I do not believe the Grandstream phones are wrong in the 
489 message, because 5003 is not that phone.

Should I have the 7608 bug reopened, should I create a new bug?  Or is there 
something I should be changing on my Asterisk server?

Here is "sip set debug 1.2.3.4":

<->
--- (0 headers 0 lines) Nat keepalive ---
Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: 
NOTIFY)
Reliably Transmitting (NAT) to 1.2.3.4:5060:
NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
v: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK0706ed06;rport
f: "asterisk" ;tag=as4612e402
t: 
m: 
i: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
o: message-summary
c: text/plain
l: 88

Messages-Waiting: no
Message-Account: sip:[EMAIL PROTECTED]
Voice-Message: 0/0 (0/0)

---
s56*CLI>
<--- SIP read from 1.2.3.4:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK0706ed06;received=1.2.3.4;rport=5060
From: "asterisk" ;tag=as4612e402
To: ;tag=as7172e0a9
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<->



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Re: [asterisk-users] WaitExten and Macros

[Tony Plack]

> You may just want a "Read" if you know how many numbers you're
> looking for.
>
> Rob
>
>

This worked, thanks!!!  Feel silly for not seeing that option, guess I was 
being lazy.

Tony Plack

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Re: [asterisk-users] [Zaptel] Checking that TDM card works?

[Vincent]

On Wed, 09 Jan 2008 08:32:40 -0600, "Darrick Hartman (lists)"
<[EMAIL PROTECTED]> wrote:
>Uncomment that if you expect it to work.  The module should be listed in 
>ZAPMODS not in rc.modules. [...] Since you have ZAPMODS commented out,
> the zaptel init script doesn't know which modules it should be using.

Thanks a lot for the tips.


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Re: [asterisk-users] WaitExten and Macros

[Tony Plack]

>> You may just want a "Read" if you know how many numbers you're
>> looking for.
>>
>> Rob
>>
>>
> This worked, thanks!!!  Feel silly for not seeing that option,
> guess I was being lazy.
>
> Tony Plack

Spoke to soon, forgot that I used Background, which is doing the same thing.  
Any reason macro context is not supported?  Just curious.

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[asterisk-users] Polycom 550 IP SoundStation Fuzzy Voice Quality

[Mike Coakley]

I'm setting up a new Asterisk system on a Dell server and I'm getting  
"fuzzy" voice between the Polycom IP SoundStation 550 and the Asterisk  
server. I've checked all of my codec settings and both the Asterisk  
and the Polycom agree on u-Law encoding. I'm using the latest release  
of the Asterisk code (1.4.17) and other software. If I call between  
phones (i.e. two Polycom's) everything is crystal clear so it doesn't  
appear to be a Polycom issue.

Here is the current information on the Polycom phone:

SoundPoint IP 550
Assembly: 2345-12500-001 Rev: A
BootBlock 2.7.0 (12500_001)
BootRom: 4.0.0.0423
SIP: v.2.2.0.0047
PolyDSP Titan Mem1 FS3 v1.7.0.0057

Here is my SIP config:

[2000]
type=friend
username=2000
password=sip-access
dtmfmode=rfc2833
[EMAIL PROTECTED]
disallow=all
allow=ulaw

Any help would be appreciated.

Thanks,

Mike


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Re: [asterisk-users] Linksys SPA-9xx Audio Issues

[Andrew Joakimsen]

Distorted and broken noise at the remote end. The odd thing is I can
never reproduce the issue but it very constant. I have set the 0.020
setting and I will continue to test with G723.

On Jan 8, 2008 7:57 PM, Daniel Cole <[EMAIL PROTECTED]> wrote:
> Can you describe the issue more please? Can the remote person not hear you at 
> all? Or is there distorted/broken voice?
>
>
> Cheers,
>
> Daniel Cole
>
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew 
> Joakimsen
> Sent: Wednesday, 9 January 2008 9:26 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Subject: [asterisk-users] Linksys SPA-9xx Audio Issues
>
> Anyone else have problems with phones like SPA-922, SPA-921, etc?
> Inbound audio is perfect but the remote end reports audio quality issues on 
> the audio the handset is sending out. It's not the network I've tried 
> asterisk 1.2, 1.4. I've used ulaw, G726, G793 & G729. Ulaw seems to be the 
> least problematic but its still an issue.
> Doesn't matter if we send the calls to the PSTN via IAX or SIP or PRI.
> I don't know it if happens all the time but about 40% of the time the remote 
> caller reports they cannot hear me.
>
>
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[asterisk-users] IAXy ringing

[Adam Moffett]

When I make calls from my IAXy I don't hear any ringing most of the time.

I've tried using the r option on the asterisk dial application to "indicate 
ringing to the calling party" but that didn't make a difference.

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Re: [asterisk-users] Busy notification with call limiting byGROUP_COUNT()

[Don Pobanz]

Peter Galiovsky wrote on Wednesday, January 09, 2008 9:39 AM 
> I want the user to be presented as busy if he has at 
> least one call active, be it incoming or outgoing. How 
> should I set things up to achieve this?

I have a very similar need. We are using call queues and would like to
have only 1 call presented to our trouble call reps at a time, but would
like to give them the ability to initiate an outgoing call on a
different line (even while on an incoming call). We are using Aastra
480i SIP phones. 

I have read that app_groupcount and setgroup may be the way to handle
this but the wiki is confusing. 

http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup

Can someone suggest an approach (preferably one currently in use)? We
are using Asterisk 1.4.17. 

Don Pobanz

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Re: [asterisk-users] WaitExten and Macros

[Tony Plack]

>>
> Spoke to soon, forgot that I used Background, which is doing the
> same thing.  Any reason macro context is not supported?  Just
> curious.
>

Okay, I figured out that I can put the macro context into the Background 
option, that works.  Bit messy in the dial plan, but it will work.

Would be nice if WaitExten supported the context like Background, then both 
solutions would end up on the same lines As is, GotoIf is my friend.

So final solution:

Use Read to get the extension at the end with GotoIf on the variable checks and 
Background with the context parameter set to the macro context.

Tony Plack

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Re: [asterisk-users] Limiting number of simultaneous calls in E1line

[Dovid B]

You can also set a busy tone in asterisk. You can send it to a context that 
keeps track of how many incoming calls there are and if there are 10 channels 
in ues then tell the 11th and on that the line is busy.
  - Original Message - 
  From: Christian Victor 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Tuesday, January 08, 2008 3:02 PM
  Subject: Re: [asterisk-users] Limiting number of simultaneous calls in E1line


I have a standard E1 line, but want to receive only 10 calls
simultaneously. I want to give engaged tone to the 11th caller 
onwards. Can I configure E1 to do this?

  Yes - that can be done on the carrier side. Lines can be configured to be 
outgoing or incoming only.

  Christian





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Re: [asterisk-users] Newbie: confusion with the new FXO/FXS card

[Andrew Stewart]

Or better yet, download the PDF of the Asterisk: The Future of Telephony 
  (aka the starfish book): 

Glenn Cobb wrote:
> Go here
> 
> www.voip-info.org
> 
> and read alot.  Almost everything you need to know (or a link to it) can
> be found through there.
> 
> Seriously, its your best starting point 
> 
> regards,
> Glenn
> 
>> -Original Message-
>> From: [EMAIL PROTECTED] 
>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>> Vytenis Sabaliauskas
>> Sent: Wednesday, January 09, 2008 7:41 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Newbie: confusion with the new FXO/FXS card
>>
>>Hello everyone,
>>
>> I'm trying to set up a Asterisk server. I have two cards 
>> - one is an BeroNet BN2S0 with two ISDN lines (4 channels):
>>
>> http://www.adcomtec.com/webstore/beronet_bn2s0.php?cat=90
>>
>> and a Rhino R8FXX with one FXO module and two FXS:
>>
>> http://www.voipsupply.com/product_info.php?products_id=2940
>>
>> I would like to set up an Asterisk server with 8 phones, 
>> which will share the phone numbers via ISDN. Sorry if i'm not 
>> writing this very clearly, since my telecomunication skills 
>> are appaling (I'm used to linux though).
>>
>> As documentation states, Rhino R8FXX can work as FXO or as 
>> FXS depending on the modules installed. How about my 
>> situation? At first, I would like to set up a simpliest thing 
>> - to make phone do anything (play WAV, echo,
>> etc.) Does anyone show me the road how to do it? A link to 
>> some manual or such...
>>
>> I tried googling, but have found no results (or didn't knew, 
>> that the info was good for me). This is my first Asterisk 
>> server and I'm very interested in bringing it up.
>>
>> Thanks. Ow, and sory for laminess :)
>>
>> --
>> V.
>>
>>
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> 
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Re: [asterisk-users] Busy notification with call limiting byGROUP_COUNT()

[Ira]

At 01:42 PM 1/9/2008, you wrote:
>I have a very similar need. We are using call queues and would like to
>have only 1 call presented to our trouble call reps at a time, but would
>like to give them the ability to initiate an outgoing call on a
>different line (even while on an incoming call). We are using Aastra
>480i SIP phones.


You could always set the Aastra to allow incoming only on line 9 so a 
second call shows up as busy and let them call out on 1-8.  That does 
it completely outside Asterisk but should work just fine.

Ira 


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Re: [asterisk-users] Busy notification with call limiting byGROUP_COUNT()

[Atis Lezdins]

On 1/9/08, Don Pobanz <[EMAIL PROTECTED]> wrote:
> Peter Galiovsky wrote on Wednesday, January 09, 2008 9:39 AM
> > I want the user to be presented as busy if he has at
> > least one call active, be it incoming or outgoing. How
> > should I set things up to achieve this?
>
> I have a very similar need. We are using call queues and would like to
> have only 1 call presented to our trouble call reps at a time, but would
> like to give them the ability to initiate an outgoing call on a
> different line (even while on an incoming call). We are using Aastra
> 480i SIP phones.
>
> I have read that app_groupcount and setgroup may be the way to handle
> this but the wiki is confusing.
>
> http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup
>
> Can someone suggest an approach (preferably one currently in use)? We
> are using Asterisk 1.4.17.

I'm currently working on porting my system to GROUP_COUNT etc.. so
what i have learned - before dial to agent from queue:

Set([EMAIL PROTECTED]);
Dial(SIP/${agent},...)

this will add result channel (if agent 222 answers) to group [EMAIL PROTECTED]

Analogically you add [EMAIL PROTECTED] and [EMAIL PROTECTED] to
outgoing calls and incoming non-queue calls.

Then - in context that will receive call from queue (with Local
channel) you check
GROUP_COUNT([EMAIL PROTECTED]) & GROUP_COUNT([EMAIL PROTECTED]) &
GROUP_COUNT([EMAIL PROTECTED])
- this will allow agent to pick up queue call only if he doesn't have
any other calls. Otherwise just execute Busy() and queue will pass
call to next agent.

You can have similar check for direct incoming calls and compare
GROUP_COUNT([EMAIL PROTECTED])+GROUP_COUNT([EMAIL 
PROTECTED])+GROUP_COUNT([EMAIL PROTECTED])<2
- to accept only two simultanous calls per agent.

And finally for outgoing calls - you just don't check any GROUP_COUNT,
just set the group - that will allow any number of outgoing calls.

Regards,
Atis

-- 
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835

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[asterisk-users] FXOTUNE update

[Matthew Fredrickson]

Hey all,


First of all, some background:

Fxotune is a utility that is used to tune the hybrid on FXO modules
For all of you with FXO modules out there, fxotune can help you adjust 
the analog and digital hybrid that is on the FXO interface and tune it 
so that it maximizes echo return loss.  This means that it will reduce 
your default echo which is received, and will help any echo cancellers 
on the line to do a better job.  If using one of the open source 
software echo cancellers, using fxotune can be the difference between 
having echo problems and not having echo problems.

This is the update:

I just committed a new version of fxotune which uses a better technique 
for measuring echo return loss.  Before, there was a simple power 
calculation which was done on the samples that would indiscriminately 
check the power of all samples received.  This works well when the line 
is silent, but if there are any sort of tones in the background or noise 
due to noisy line conditions, this calculation can yield results which 
may improve things, but are not the best results.

The new method involves using fourier analysis of the tones used in the 
test reference which is sent out.  Using fourier analysis instead of the 
power calculation, we can cut through any background noise which is not 
related to our test sequence's set of tones, producing a much more 
accurate and noise immune calculation.

If you have run fxotune before on your lines, I recommend you re-run it 
with the updated version of the utility.  As of this moment, it is not 
yet in a released version of zaptel, but if you check out either latest 
1.2 or 1.4 branches, it will be there.  If you run fxotune with the -v 
option, it will tell you what the return loss it calculates for each AC 
impedance and set of coefficient parameters in dB.

In order to use the new analysis calculations, you do not need to pass 
any sort of special parameters to fxotune, it does the new analysis 
technique by default.

Please let me know if you have any issues as well.  Thanks!

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] Intercom & Paging with Polycoms

[C F]

You can use app_page.
If you call a local channel that uses app_chanisavail first then you
should be able to call as many as  you need to. You can actualy break
it down in groups that way.

On Jan 9, 2008 1:28 PM, Rob Schall <[EMAIL PROTECTED]> wrote:
> I've been able to page to a specific phone (intercom type of thing), but
> I'd like to have a macro or agi that pages all phones but first checks
> if their on the phone. It looked like there used to be a pageall.agi
> type of script on the wiki, but that link isn't valid anymore. Does
> anyone have that script, or something else that would work? I would just
> do SIP/1000&SIP/1001, but there are about 60 phones involved.
>
> Rob
>
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[asterisk-users] Two Asterisk Boxes Playing Together

[Shane D]

Okay, here's the dal.

Me and my friend both have asterisk boxes. I want to be able to type
extension 27 on my end and get his extension 27, and he wants to be
able to type 277 on his end and get my extension . We both have
FQDN's, and would like to see about doing this either over sip or
IAX...

Any help much papreciated,
Shane

-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] Two Asterisk Boxes Playing Together

[Alex Balashov]

You can do it over SIP or IAX2.  What you would do is set up a SIP or IAX2 
trunk between the two Asterisk boxes (type=peer) and then define 
extensions in your dial plan such that those extensions are routed over
the trunk.

In sip.conf, define your trunk.  I'm going to use SIP in this example,
but you can use IAX2 if you like.  Do this on both machines:

[SomeTrunkName]

type=peer
insecure=very
nat=no
canreinvite=no
host=the_remote_ip_addr
context=dialplan_context_for_calls_to_land_into


On your Asterisk box:

exten => 27,1,Dial(SIP/[EMAIL PROTECTED])

On his:

exten => 277,1,Dial(SIP/[EMAIL PROTECTED])

Also, make sure to create [dialplan_context_for_calls_to_land_into]
in your dial plan on both sides to route calls coming over the trunk
inward into the system, i.e. map  to your phone.

Hope that helps,

-- Alex

On Wed, 9 Jan 2008, Shane D wrote:

> Okay, here's the dal.
>
> Me and my friend both have asterisk boxes. I want to be able to type
> extension 27 on my end and get his extension 27, and he wants to be
> able to type 277 on his end and get my extension . We both have
> FQDN's, and would like to see about doing this either over sip or
> IAX...
>
> Any help much papreciated,
> Shane
>
> -- 
> -Shane
> Blog: http://blind-geek.com/blog/
> CoOwner: http://sjtechzone.com
> AIM: inhaddict
> Skype: chatter8712
>
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--
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] Two Asterisk Boxes Playing Together

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Shane D wrote:
> Okay, here's the dal.
> 
> Me and my friend both have asterisk boxes. I want to be able to type
> extension 27 on my end and get his extension 27, and he wants to be
> able to type 277 on his end and get my extension . We both have
> FQDN's, and would like to see about doing this either over sip or
> IAX...

http://www.voip-info.org/wiki-Asterisk+-+dual+servers

- --
Kind Regards,

Matt Riddell
Director
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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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Re: [asterisk-users] Two Asterisk Boxes Playing Together

[Rob Hillis]

Google is your friend.

http://www.google.com.au/search?hl=en&client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial&hs=jR6&q=asterisk+iax+two+servers&btnG=Search&meta=

Shane D wrote:
> Okay, here's the dal.
>
> Me and my friend both have asterisk boxes. I want to be able to type
> extension 27 on my end and get his extension 27, and he wants to be
> able to type 277 on his end and get my extension . We both have
> FQDN's, and would like to see about doing this either over sip or
> IAX...
>
> Any help much papreciated,
> Shane
>
>   


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Re: [asterisk-users] x100p wcfxo hangup on outgoing calss

[Jonathan GF]

Hi Miguel,

i'm in Spain like you. For a normal operational system inmediate should be
set to no. Busycount and Busydetect can be improved performance with
busypattern.
The pattern should be shown in the CLI, just take a look.

In Spain we use Kewlstart. If you card allows it try use Julian J. Menendez
patch for AnswerOnPolarity... If your provider allows it also, you won't
haver further issues, just tune the echo with fxotune.

Disable echotraining. Not needed. Disable fax detection. Will work, let's
see ;)

Regards,

Jonathan GF



On Jan 4, 2008 11:18 PM, Miguel A Felipe Rodríguez <[EMAIL PROTECTED]>
wrote:

> I have changed the signalling of the x100p to a fxsls, now i can make
> outgoing calls, but now I have another problem, cant detect hangup. I
> post the zapara.conf and the zaptel.conf so if any has idea of waht to
> change y have tested changing busydetect, busypattern, callprogress,
> etc.. but no whet results.
>
> zaptel.conf
> fxsls=1
> loadzone= es
> defaultzone = es
>
> zapata.conf
> [channels]
> language=es
> context=incoming
> switchtype=national
> signalling=fxs_ls
> usecallerid=yes
> cidsignalling=dtmf
> cidstart=polarity
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=no
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echotraining=yes
> rxgain=2.0
> txgain=1.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=yes
> callerid=asreceived
> busydetect=yes
> busycount=6
> callprogress=no
> faxdetect=incoming
> channel => 1
>
>
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[asterisk-users] OT: Traffic Shaping

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

Does anyone know of a cheap (very cheap) dual port traffic shaping box
(i.e. sub $100) that can be configured for IAX/SIP?

- --
Kind Regards,

Matt Riddell
Director
___

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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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=i190
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Re: [asterisk-users] Two lines for outgoing calls

[Jonathan GF]

Dominik,

apart from the good responses, please get rid of the 't' in the options of
dial or you will be allowing the called party to transfer the call while you
are paying.

Regards,

Jonathan GF



On Dec 26, 2007 3:32 PM, Dominik Zalewski <[EMAIL PROTECTED]> wrote:

> Dear All,
>
> I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel
> 2.6.18.
>
> I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm
> using below context for dialing out.
>
> [outbound-local]
> exten => _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
> exten => _9X,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
> exten => _9ZXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
> exten => _9ZXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr)
> exten => _90900X,1,Dial(Zap/g1/${EXTEN:1},30m,tTr)
>
> When Zap/1 is busy and I try to call, it will use Zap/2 which is fine
> but there is something wrong cause I hear one ring and then a weird
> sound like a noise or something and then hangup. I have to reload zaptel
> modules and then everything works fine for a while.
>
>   -- Executing [EMAIL PROTECTED]:1] Dial("SIP/200-08221590",
> "Zap/g1/150|30|tTr") in new stack
>-- Called g1/150
>-- Zap/2-1 answered SIP/200-08221590
>-- Hungup 'Zap/2-1'
>
> I even thought that second fxo module is broken so I changed it. No
> results.
>
> Any ideas?
>
> Thank you in advance,
>
> Dominik
>
>
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Re: [asterisk-users] OT: Traffic Shaping

[Erik Anderson]

On Jan 9, 2008 8:33 PM, Matt Riddell <[EMAIL PROTECTED]> wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi,
>
> Does anyone know of a cheap (very cheap) dual port traffic shaping box
> (i.e. sub $100) that can be configured for IAX/SIP?

Pick up a Linksys WRT54GL and install dd-wrt on it. That will traffic
shape any type of traffic you want.  I have installed several of these
around the country and they work great for prioritizing VoIP traffic.

-Erik

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[asterisk-users] Asterisk 1.4 and ISDN-BRI support

[Jaap Winius]

Hi list,

Has anyone been able to get ISDN-BRI support to work reliably on  
Asterisk 1.4? If so, I'd love to know how you did it (hardware,  
distro, kernel, modules, versions, config files).

I've tried to get it to work on a Debian etch system with an HFC-PCI  
card and the zaptel package (v1.4.7, also from xorcom.com), but with  
no luck: all three channels that are created when the zaphfc or  
vzaphfc module loads always change to an 'in use' state as soon as  
Asterisk starts up and so can't be used. I've received the exact same  
results on two different systems.

Thanks,

Jaap

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Re: [asterisk-users] OT: Traffic Shaping

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Erik Anderson wrote:
> On Jan 9, 2008 8:33 PM, Matt Riddell <[EMAIL PROTECTED]> wrote:
>> -BEGIN PGP SIGNED MESSAGE-
>> Hash: SHA1
>>
>> Hi,
>>
>> Does anyone know of a cheap (very cheap) dual port traffic shaping box
>> (i.e. sub $100) that can be configured for IAX/SIP?
> 
> Pick up a Linksys WRT54GL and install dd-wrt on it. That will traffic
> shape any type of traffic you want.  I have installed several of these
> around the country and they work great for prioritizing VoIP traffic.

Heh yeah that's what I was thinking of doing.  What's the traffic
shaping like?  Can I specify max bandwidth etc or use hfsc shaping?

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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Re: [asterisk-users] Asterisk 1.4 and ISDN-BRI support

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Jaap Winius wrote:
> Hi list,
> 
> Has anyone been able to get ISDN-BRI support to work reliably on  
> Asterisk 1.4? If so, I'd love to know how you did it (hardware,  
> distro, kernel, modules, versions, config files).
> 
> I've tried to get it to work on a Debian etch system with an HFC-PCI  
> card and the zaptel package (v1.4.7, also from xorcom.com), but with  
> no luck: all three channels that are created when the zaphfc or  
> vzaphfc module loads always change to an 'in use' state as soon as  
> Asterisk starts up and so can't be used. I've received the exact same  
> results on two different systems.

I have the Digium b410p card installed and kind of working at a few
sites.  There is not much in the way of support, the hardware echo
canceller doesn't work and mISDN seems a little immature.

I've had a million times more success with analogue and PRI cards, and
the new FXOtune from mattf looks like it should be really nice!

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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y8Ibn0xw4ZM5wWR1ImVhy3A=
=e0Ec
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Re: [asterisk-users] OT: Traffic Shaping

[Erik Anderson]

On Jan 9, 2008 9:40 PM, Matt Riddell <[EMAIL PROTECTED]> wrote:
>
> Heh yeah that's what I was thinking of doing.  What's the traffic
> shaping like?  Can I specify max bandwidth etc or use hfsc shaping?

DD-WRT will do both HTB and HFSC shaping, though I've only ever used HTB.

Here's the dd-wrt wiki page on its QoS implementation:

http://www.dd-wrt.com/wiki/index.php/Quality_of_Service

Looks like they don't recommend HFSC currently due to some lag issues.
That might have been fixed, though, in the more recent firmware
builds.

-Erik

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[asterisk-users] IEEE 802.1x capable sip phones

[Jeronimo Romero]

Does anyone know if sip phones from any of the major IP phone vendors
support 802.1x authentication? Any feedback would be greatly
appreciated. 

 

Thanks in advance. 

 

==
Jeronimo Romero
EUS Networks
Email: [EMAIL PROTECTED]
Cell: 917-332-7238
Office: 212-624-5943
Web: www.euscorp.com
== 

 

 

 

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Re: [asterisk-users] Polycom 550 IP SoundStation Fuzzy Voice Quality

[Doug Lytle]

Mike Coakley wrote:
> I'm setting up a new Asterisk system on a Dell server and I'm getting  
> "fuzzy" voice between the Polycom IP SoundStation 550 and the Asterisk  
>   

Probably related to this bug:

http://bugs.digium.com/view.php?id=11243

Doug


-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."



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Re: [asterisk-users] IEEE 802.1x capable sip phones

[Kev S]

Im pretty sure the Cisco Unified IP Phones 7900 Series phones support 
this, Dont quote me on it but its worth checking out

Kev


Jeronimo Romero wrote:
>
> Does anyone know if sip phones from any of the major IP phone vendors 
> support 802.1x authentication? Any feedback would be greatly appreciated.
>
>  
>
> Thanks in advance.
>
>  
>
> ==
> Jeronimo Romero
> EUS Networks
> Email: [EMAIL PROTECTED] 
> Cell: 917-332-7238
> Office: 212-624-5943
> Web: www.euscorp.com 
> ==
>
>  
>
>  
>
>  
>
> 
>
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-- 
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Re: [asterisk-users] OT: Traffic Shaping

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Erik Anderson wrote:
> On Jan 9, 2008 9:40 PM, Matt Riddell <[EMAIL PROTECTED]> wrote:
>> Heh yeah that's what I was thinking of doing.  What's the traffic
>> shaping like?  Can I specify max bandwidth etc or use hfsc shaping?
> 
> DD-WRT will do both HTB and HFSC shaping, though I've only ever used HTB.

Sweet. We had been using HTB but upgraded all our CPE to use HFSC when
AstLinux did and found it great.

> Here's the dd-wrt wiki page on its QoS implementation:
> 
> http://www.dd-wrt.com/wiki/index.php/Quality_of_Service

Cool, looks good, although I would have thought Default Bandwidth Level
would have been pretty self explanatory if it works as expected.

I guess I'll have to try it out and update the wiki if it does.

> Looks like they don't recommend HFSC currently due to some lag issues.
> That might have been fixed, though, in the more recent firmware
> builds.

Will try both out and see how they go.

Thanks for the pointers.

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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Re: [asterisk-users] IEEE 802.1x capable sip phones

[Robert Moskowitz]

Jeronimo Romero wrote:
>
> Does anyone know if sip phones from any of the major IP phone vendors 
> support 802.1x authentication? Any feedback would be greatly appreciated.
>
This is so unlikely.  I worked on 802.1X and 802.11i.  There is just too 
much overhead there.  No way to meet the ITU 50ms disruption requirement.

Plus it is a lot of code.  Wait until 802.11r and/or 11s get done to get 
any real secure roaming.  Rather implement SRTP.
>
>  
>
> Thanks in advance.
>
>  
>
> ==
> Jeronimo Romero
> EUS Networks
> Email: [EMAIL PROTECTED] 
> Cell: 917-332-7238
> Office: 212-624-5943
> Web: www.euscorp.com 
> ==
>
>  
>
>  
>
>  
>
> 
>
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Re: [asterisk-users] IEEE 802.1x capable sip phones

[Jeronimo Romero]

I called Cisco and they are so far the only vendor that offers it. 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Moskowitz
Sent: Wednesday, January 09, 2008 11:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IEEE 802.1x capable sip phones

Jeronimo Romero wrote:
>
> Does anyone know if sip phones from any of the major IP phone vendors 
> support 802.1x authentication? Any feedback would be greatly
appreciated.
>
This is so unlikely.  I worked on 802.1X and 802.11i.  There is just too

much overhead there.  No way to meet the ITU 50ms disruption
requirement.

Plus it is a lot of code.  Wait until 802.11r and/or 11s get done to get

any real secure roaming.  Rather implement SRTP.
>
>  
>
> Thanks in advance.
>
>  
>
> ==
> Jeronimo Romero
> EUS Networks
> Email: [EMAIL PROTECTED] 
> Cell: 917-332-7238
> Office: 212-624-5943
> Web: www.euscorp.com 
> ==
>
>  
>
>  
>
>  
>
>

>
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[asterisk-users] forward call intended for another domain

[Randall Smith]

I'm new here.  For a registered SIP 'friend' that dials an address not 
handled by my server (say [EMAIL PROTECTED]), how do I get Asterisk to 
forward that call to ekiga.net?

Thanks.

Randall


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Re: [asterisk-users] Asterisk 1.4 and ISDN-BRI support

[stoffell]

> Has anyone been able to get ISDN-BRI support to work reliably on
> Asterisk 1.4? If so, I'd love to know how you did it (hardware,
> distro, kernel, modules, versions, config files).

Maybe your best bet is using bristuff, the bristuff-0.4.0 series are
tests for asterisk 1.4, I haven't tested them out yet.  (
http://junghanns.net/downloads/ )

mISDN however is the other option..

cheers

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Re: [asterisk-users] Which IP Phone is really the best?

[stoffell]

I'm in Europe (yeah, that does matter when choosing a good phone!) ..

Some of my (and my customers') favorites:

- Polycom (pretty much all of them)
- Thomson ST2030
- Siemens Gigaset C450 IP dect (for wireless phones)

cheers,
stoffell

---
http://www.electromarket.be

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Re: [asterisk-users] Which IP Phone is really the best?

[randulo]

On Aug 31, 2007 7:11 PM, William Herrera <[EMAIL PROTECTED]> wrote:
> Out of all the IP Phones out there, which one is the best and why?

My experiences are with Polycom (ip500) and the Linksys/Cisco SPA94?.
I like both but they are different. The best suggestion on this thread
was to pick 4 and show them to the cust. Last time I was at a VoIP
convention, there was a stand with about 50 models of phones to try
from all the main manufacturers.

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Re: [asterisk-users] HFC-S zap channels always busy

[Tzafrir Cohen]

On Sat, Jan 05, 2008 at 04:27:25PM +0100, Patrick wrote:
> 
> On Sat, 2008-01-05 at 01:18 +0100, Jaap Winius wrote:
> > Quoting Michiel van Baak <[EMAIL PROTECTED]>:
> > 
> > >> I don't know about NL but in the UK, multiple ISDN2e lines have to be
> > >> configured as bri_cpe_ptp not bri_cpe_ptmp. Have you tried this mode?
> > >
> > > It's the same here in .nl
> > 
> > Interesting, but I would think this to be unnecessary in my case,  
> > since I have only one ISDN-BRI line. It's just that for some reason  
> > the software keeps loading both the zaphfc and vzaphfc modules
> 
> Why not remove or disable the module you don't need just to make sure it
> is not interfering?

To be more specific: I get the wierd impression that either both modules 
somehow get interrupts from the two cards, or each module handles a
different card. This hsouldn't happen.

So try blacklisting one of them:

  echo 'blacklist vzaphfc' >/etc/modprobe.d/blacklist_hfc

/etc/init.d/asterisk zaptel-fix

Or maybe blacklist zaphfc instead.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Asterisk 1.4 and ISDN-BRI support

[Tzafrir Cohen]

On Thu, Jan 10, 2008 at 07:27:12AM +0100, stoffell wrote:
> > Has anyone been able to get ISDN-BRI support to work reliably on
> > Asterisk 1.4? If so, I'd love to know how you did it (hardware,
> > distro, kernel, modules, versions, config files).
> 
> Maybe your best bet is using bristuff, the bristuff-0.4.0 series are
> tests for asterisk 1.4, I haven't tested them out yet.  (
> http://junghanns.net/downloads/ )

BTW:

[EMAIL PROTECTED]:~/Proj/Packs/bristuff/bristuff-0.4.0-test6$ cat 
patches/zaphfc/series
100-local_zap.diff
101-newzaptel.diff
102-florz.diff

-- 
   Tzafrir Cohen
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+972-50-7952406   mailto:[EMAIL PROTECTED]
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Re: [asterisk-users] Asterisk 1.4 and ISDN-BRI support

[IT-Connect]

stoffell schrieb:

Has anyone been able to get ISDN-BRI support to work reliably on
Asterisk 1.4? If so, I'd love to know how you did it (hardware,
distro, kernel, modules, versions, config files).



Maybe your best bet is using bristuff, the bristuff-0.4.0 series are
tests for asterisk 1.4, I haven't tested them out yet.  (
http://junghanns.net/downloads/ )

mISDN however is the other option..

cheers

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I've run Asterisk 1.4.17 with mISDN 1.7 on a Suse 10.0 with 
Kernel-Version 2.6.23.13. But there are any issues

with newer Kernel-Versions. You have to patch the mISDN packet.
If you're interested, you can get a description from me.

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Re: [asterisk-users] How to check if a SIP phone isforwardedwithoutringing it ?

[Benny Amorsen]

"Steve Langstaff" <[EMAIL PROTECTED]> writes:

> I agree that sending an OPTION message from the Asterisk server could
> well have a low processing load.
>
> The original poster wanted to use OPTIONS sent FROM the Asterisk server
> to query the phone state, so I don't think your concerns about receive
> processing come into the picture.

If the asterisk community is going to ask the phone manufacturers to
comply to the RFC, it looks rather silly when asterisk itself doesn't.

Not that I have a good solution.


/Benny



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Re: [asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?

[Benny Amorsen]

Olivier <[EMAIL PROTECTED]> writes:

> To get a polite "go to hell !" in return ?  ;-)

I think the vendors will be nicer than that. Asterisk has a good bit
of the VoIP PBX market.


/Benny


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