RE: [Asterisk-Users] Asterisk on amd SERVER

[MBIT Technologies]

You shouldn't really have any problems with i386 version on the AMD. When
centos moved to 4.3 the x86_64 bit version was a mess with trying to install
packages. 


Regards


Mark Brooker
T: 02 4959 8670
M: 0415 846 865
F: 02 9882 0947
E: [EMAIL PROTECTED]
W: http://www.mbit.com.au


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kanishka
Somaratne
Sent: Friday, 5 May 2006 4:17 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk on amd SERVER

Hi
I am going to install asterisk on an AMD server, did any one had problems 
installing it on an AMD server ?

Regards
Kani 

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk on amd SERVER

[Kanishka Somaratne]

Hi
I am going to install asterisk on an AMD server, did any one had problems 
installing it on an AMD server ?


Regards
Kani 


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.

[Asterisk]

FIXED.
Found and
fixed the problem.
Teach me to
cut code from AAH 2.8
Issue was..
exten =>
s,1,Set(__SIPADDHEADER=Call-Info: answer-after=0)   THIS DOES NOW WORK.
exten =>
s,1,Set(__SIPADDHEADER=Call-Info:\;answer-after=0)  THIS DOES.
 
Strange what
a type can do.  Also strange why the other worked on some other
handsets, but not yours..
 
YAY!
 
James



Hadley Rich wrote:

  On Thursday 04 May 2006 20:53, Asterisk wrote:
  
  
The handsets do not work with the SIP flag to make them AUTO-ANSWER. (As
documented is should)
Ie, you cannot use them with intercom or Page features.

  
  
Works fine here;

SIPAddHeader(Call-Info:\;answer-after=0)

hads

  




___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

[Steve Underwood]

Scott Gifford wrote:


Colin Anderson <[EMAIL PROTECTED]> writes:

 


Why not capture the faxes (in or out) in tiff format, instead of audio
format?  Setup your asterisk box to relay faxes!
 


I think in this case the impact on the client would be much greater if you
can show them a recreation of the image from the raw data; you could always
claim that a TIFF file was altered (which it can be, trivially) but it's
pretty much impossible to change the raw audio to your ends unless you are
in a Tom Clancy novel. 
   



Why is this hard to fake at all?  You send a different fax to your
system, and replace the Asterisk audio file with the one from the
altered fax.  Additionally, the client has no realistic way of
verifying the correctness of your audio-to-fax translation tool; it
could just as easily output a TIFF file completely different from the
one that was actually faxed.
 

In most cases, forensic analysis of the audio from another machine would 
easily show it was a fake. It would lack tell-tale fingerprints of the 
true path, unless it was done with extreme care. Certainly using exactly 
the same model of FAX machine that sent the genuine FAX would be a must. 
Not just for the vendor information it sends, but for the fine details 
in how its modems behave. To pass of the altered fax as being from the 
original sender would require careful control of the DSP.


Steve

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Is FWD down ???

[Joseph]

I'm not getting registration from:  iax2.fwdnet.net

-- 
#Joseph
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SPA941 et al LED indications

[David Zanetti]

Hi all.

The SPA941 and friends have pretty multicoloured LEDs, but there doesn't
appear to be any support for SUBSCRIBE/NOTIFY as * as implemented for
extension hinting.

Has anyone managed to get the phone to support this?

Thanks!

-- 
David Zanetti <[EMAIL PROTECTED]>
Team Leader, Systems Administration
Catalyst IT Limited
+64-4-8032233 +64-21-402260


signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Switchboard solutions, interactions with handset

[Nicolás Gudiño]

On 5/4/06, Arnar Birgisson <[EMAIL PROTECTED]> wrote:

Hi there,

I'm looking into developing an in-house switchboard application. Does
anyone here know of a way to control a hard-phone from such an
application.

For example, the attendant forwards a call with another one in queue.
Once the first call has been forwarded (by keyboard shortcuts or
dragging-n-dropping) - she presses a button (on the computer) to
answer the waiting call.

Now, if the switchboard application embeds a soft-phone, I can figure
out how to do this. But suppose the attendant is using a hard-phone
(since it's more reliable) with a headset - can she do the above
things without having to press any of the phones buttons?

Wouldn't this require the application to somehow control if the phone
is off-hook or on-hook? Is there some other way I'm not seeing and/or
has someone here implemented similar stuff?

Could I possibly keep an open channel in Asterisk to the attendants
phone, and bridge that with whatever channel requested by the
switchboard application? I have found some mention of this, bridging
channels, in the mailing list archives, but not in the AMI
documentation. Is this maybe something that's still only on the svn
trunk?


I have done something similar using a modified Flash Operator Panel
and a phone with autoanswer capabilities (polycom 501), while the
operator is using a headset. Then you can use standard manager actions
to redirect calls to the operator. Regards,

--
Nicolás Gudiño
Buenos Aires - Argentina
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Tom Engleward]

--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
wrote:
> >> If Asterisk has a DNS lookup failure it will
> never
> >> retry that lookup.
> > "Never" meaning until the next "reload" command is
> > issued, or until the next "restart" command is
> issued,
> > or until the next time the OS reboots, or until
> the
> > next time asterisk and its config files are
> deleted
> > and reinstalled?
> 
> Correct.  Perhaps "never automatically" would be a
> better choice of words.
Wait, that was a multiple choice question: which
option is the least-drastic sufficiently drastic
option to force asterisk to retry?
If a mere "reload" is sufficient, then the periodic
test-and-reload which I suggested (which Andrew
Kohlsmith correctly said is just a band-aid) would at
least be an effective band-aid for the short term,
until the actual cause of the DNS lookup failure is
found and fixed. A proper fix is preferable, but a
band-aid is better than nothing. Of course if using IP
address instead of hostname successfully avoids the
problem, then that's a better band-aid, but only is
practical if the host in question is not subject to
having its IP address changed.



__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.

[Tom Vile]

works for me as well.

On 5/4/06, kevin ling <[EMAIL PROTECTED]> wrote:

Hi,

But it's seems the auto-answer function work on my spa-941. Have you upgrade
to the latest firmware version?

Regards,
kevin

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Friday, May 05, 2006 9:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.



Asterisk wrote:

> Hello all,
> I want to report a BUG with the Linksys SPA94X so it is general
> knowledge and that we can all make noise about it so it will get fixed
> sooner..
>
> The handsets do not work with the SIP flag to make them AUTO-ANSWER.
> (As documented is should)
> Ie, you cannot use them with intercom or Page features.
>
> This works with the Sipura841 fine.  So linksys broke it.  Um..
> interesting is it not, considering it works with there SPA9000 unit...
> sounds a bit fishy to me..
>
> So any Linksys owners using Asterisk, do pass on some discontentment,
> and Email linksys tech support at [EMAIL PROTECTED]
> 
> And tell them you have this issue..
>
>
> James
>
Curious, as I tried to get this to work with the 841, and though the phone
does auto answer, the called or paged party hears dial tone as well as the
page, just as if one went off hook by pressing the speaker button. The Pager
does NOT hear dial tone.
I sent support some information, but so far no help. They asked for more and
I have yet to get back to them

Curious, very curious.

>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Eric \"ManxPower\" Wieling]

Tom Engleward wrote:

--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
wrote:

If Asterisk has a DNS lookup failure it will never
retry that lookup.

"Never" meaning until the next "reload" command is
issued, or until the next "restart" command is issued,
or until the next time the OS reboots, or until the
next time asterisk and its config files are deleted
and reinstalled?


Correct.  Perhaps "never automatically" would be a better choice of words.

--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.

[kevin ling]

Hi,

But it's seems the auto-answer function work on my spa-941. Have you upgrade
to the latest firmware version?

Regards,
kevin 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Friday, May 05, 2006 9:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.



Asterisk wrote:

> Hello all,
> I want to report a BUG with the Linksys SPA94X so it is general 
> knowledge and that we can all make noise about it so it will get fixed 
> sooner..
>
> The handsets do not work with the SIP flag to make them AUTO-ANSWER. 
> (As documented is should)
> Ie, you cannot use them with intercom or Page features.
>
> This works with the Sipura841 fine.  So linksys broke it.  Um.. 
> interesting is it not, considering it works with there SPA9000 unit...  
> sounds a bit fishy to me..
>
> So any Linksys owners using Asterisk, do pass on some discontentment, 
> and Email linksys tech support at [EMAIL PROTECTED]
> 
> And tell them you have this issue..
>
>
> James
>
Curious, as I tried to get this to work with the 841, and though the phone
does auto answer, the called or paged party hears dial tone as well as the
page, just as if one went off hook by pressing the speaker button. The Pager
does NOT hear dial tone.
I sent support some information, but so far no help. They asked for more and
I have yet to get back to them

Curious, very curious.

>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.

[John Novack]

Asterisk wrote:


Hello all,
I want to report a BUG with the Linksys SPA94X so it is general 
knowledge and that we can all make noise about it so it will get fixed 
sooner..


The handsets do not work with the SIP flag to make them AUTO-ANSWER. 
(As documented is should)

Ie, you cannot use them with intercom or Page features.

This works with the Sipura841 fine.  So linksys broke it.  Um.. 
interesting is it not, considering it works with there SPA9000 
unit...  sounds a bit fishy to me..


So any Linksys owners using Asterisk, do pass on some discontentment, 
and Email linksys tech support at
[EMAIL PROTECTED] 


And tell them you have this issue..


James

Curious, as I tried to get this to work with the 841, and though the 
phone does auto answer, the called or paged party hears dial tone as 
well as the page, just as if one went off hook by pressing the speaker 
button. The Pager does NOT hear dial tone.
I sent support some information, but so far no help. They asked for more 
and I have yet to get back to them


Curious, very curious.



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREA CHABLE?

[Eric \"ManxPower\" Wieling]

Colin Anderson wrote:
That is not a good metric for call completion.  Shitty quality will not be 
counted in that metric...


Ah, true dat. However, if quality was crappy believe me my users would let
me know. They are salespeople and wholly intolerant of anything that keeps
them from yipping on the phone. I also run exactly the same rig at my house
backhauled to our main Asterisk box to force me to eat my own dog food, and
Snom > SIP > * > IAX2 > * > PRI sounds perfect pretty much 100% of the time.
The only time I get chop is when I use my cordless to the TDM400: Cordless >
TDM400 > * > IAX2 > * > PRI. Salespeople keep asking for cordless phones,
but I keep telling them forget it. Cordless to an FXS to IAX over the
Internet is just begging for trouble. 


Why not cordless to a channel bank to Asterisk?

Of course, you can only have so many cordless phones in one area.

--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Tom Engleward]

--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
wrote:
> If Asterisk has a DNS lookup failure it will never
> retry that lookup.
"Never" meaning until the next "reload" command is
issued, or until the next "restart" command is issued,
or until the next time the OS reboots, or until the
next time asterisk and its config files are deleted
and reinstalled?


__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: SV: [Asterisk-Users] Polycom 501 - Disable DND feature?

[Derek Listmail Acct]

You can disable the DND button completly.  I think that will get you what
you want.

I don't have a 500/501 handy to find out which button it is, but you can
check in Menu -> Status -> Diagnostics -> Test Hardware -> Keypad
Diagnostics.

It's button 9 on my 600 and this disabled it:



--Derek



> Well, yes and no. I tested that before and it causes a silent ring instead
> of a call rejection. I actually want to disable the entire feature. So the
> phone always rings unless you're actually on the phone.
>
> Thanks for the reply though!
>
> Regards,
> Jan
>
> 
>
> Från: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] För Jerry Jones
> Skickat: den 4 maj 2006 15:00
> Till: Asterisk Users Mailing List - Non-Commercial Discussion
> Ämne: Re: [Asterisk-Users] Polycom 501 - Disable DND feature?
>
>
> Attribute Values Default Interpretation
> call.rejectBusyOnDnd 0, 1 1 If set to 1, reject all incoming calls with
> the reason "busy" if do-not-disturb is
> enabled.
>
> Have not used, but looks like it may ignore the key if this is 0
>
> Let us know...
>
>
> On May 4, 2006, at 2:22 AM, <[EMAIL PROTECTED]> <[EMAIL PROTECTED]>
> wrote:
>
>
>   Hi,
>
>   Is it possible to disable the DND feature on a Polycom 501?
>
>   Regards,
>   Jan
>   ___
>   --Bandwidth and Colocation provided by Easynews.com --
>
>   Asterisk-Users mailing list
>   To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] FW: NuFone Update: DIDs

[Paul]

Steve Prior wrote:

> Rich Adamson wrote:
>
>>
>> At least outbound calls still work, even though they changed IP
>> addresses (and probably colo locations).
>>
>>
>
> Maybe not so much now.  I just got a disconnect notice from nufone
> which states that I have a positive balance in my account, but still
> need to add money to bring it up above zero for my account to be
> re-enabled.  Somebody just broke the billing code I think...
>
> Steve

I got one and it was sent to the address I originally signed up with. I
cahnged my contact address a few months ago and when I login it clearly
shows the newer address in my profile.

This is why I always insist on having a test system when I am coding. I
am just not comfortable experimenting with production systems. I'm such
a coward!

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Eric \"ManxPower\" Wieling]

Tom Engleward wrote:

--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
wrote:

Are you specifying the remote Asterisk box by IP or
by hostname.  If by 
hostname, then specify it by IP.  Asterisk's DNS

lookup support has issues.

In the trunk peer details in AMP I'd set "host=" to a
hostname. I've switched it to IP address, and will see
whether that makes a difference. Should I switch from
hostname to IP address in the register string too?


If Asterisk has a DNS lookup failure it will never retry that lookup.


--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREA CHABLE?

[Colin Anderson]

>That is not a good metric for call completion.  Shitty quality will not be 
>counted in that metric...

Ah, true dat. However, if quality was crappy believe me my users would let
me know. They are salespeople and wholly intolerant of anything that keeps
them from yipping on the phone. I also run exactly the same rig at my house
backhauled to our main Asterisk box to force me to eat my own dog food, and
Snom > SIP > * > IAX2 > * > PRI sounds perfect pretty much 100% of the time.
The only time I get chop is when I use my cordless to the TDM400: Cordless >
TDM400 > * > IAX2 > * > PRI. Salespeople keep asking for cordless phones,
but I keep telling them forget it. Cordless to an FXS to IAX over the
Internet is just begging for trouble. 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] asterisk can't find address host. Problem in chan_sip.c

[makevuy]

hello

I have an asterisk server with a public IP address and a nat address
like alias. I have 20 sip clients with private IP address. I don't Know
why, sometimes, when I try to call between 2 phones, y see the next
menssage in the astersk console:

 Can't find address for host 'XX'

What could happen??

Thaks for all and regards

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] FW: NuFone Update: DIDs

[Steve Prior]

Rich Adamson wrote:



At least outbound calls still work, even though they changed IP 
addresses (and probably colo locations).





Maybe not so much now.  I just got a disconnect notice from nufone
which states that I have a positive balance in my account, but still
need to add money to bring it up above zero for my account to be
re-enabled.  Somebody just broke the billing code I think...

Steve
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Fwd: meetme conference latency degrades...

[Colin Anderson]

Yes I believe this is a 1.0.9 bug unfortunately I can't find a reference to
it in Mantis except for this: 

http://bugs.digium.com/view.php?id=5971

I run 1.0.9 and I do all of my MeetMe's through the PRI's. This is stupid,
because we are using 2 X # of users / PRI channels but MeetMe runs solid. I
just had a 20 person, 30 channel 3 hour marathon MeetMe conference couple
weeks ago (I wasn't in it, but a bunch of company suits and some external
mucky-mucks were) and the feedback I got was how clear everything was. 

hth

-Original Message-
From: Michael George [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 04, 2006 4:14 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Fwd: meetme conference latency degrades...


I haven't seen this appear on the list, so I thought I would resend
it...

Sorry for the repost if it did appear before...

- Forwarded message from Michael George  -

Date: Wed, 3 May 2006 21:48:09 -0400
From: Michael George 
Subject: meetme conference latency degrades...
To: asterisk-users@lists.digium.com

We have recently started making more frequent use of the meetme
conference of our * system.

We are using v1.0.8 with a 2.6.11 kernel on our system.

We generally have 4 callers in it: two with the gsm codec and 2 with g729.
Initially, the conference works fine and there is little latency.  After
about 15min., though, the latency is very noticable and by 25min it's
unbearable.

If we all leave the conference and return, the latency is unnoticable
again.

The load on the box is minimal, and only our meetme is running most of
the time.  Checking system load with top shows 0.1 or less.

We have no digium hardware and use ztdummy for our timing device.
zttest yields results generally in the area of 99.96%, but about 3-4%
will be as low as 95%.

In much smaller systems with Digium hardware, the accuracy is never
below 99.98% and is often 100%.

Is this apparent inaccuracy of the ztdummy timer likely the cause of the
increasing latency in our meetme conference? 

Is there any way to improve it?

Thank you, in advance, for any help.

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREA CHABLE?

[Andrew Kohlsmith]

On Thursday 04 May 2006 18:04, Colin Anderson wrote:
> SELECT count( calldate )
> FROM `cdr`
> WHERE calldate
> BETWEEN '2006-01-01 00:00:01' AND '2006-05-04 15:51:00' AND channel
> LIKE '%IAX2%' AND duration < 1 OR calldate
> BETWEEN '2006-01-01 00:00:01' AND '2006-05-04 15:51:00' AND dstchannel
> LIKE '%IAX2%' AND duration < 1
>
> count(calldate):
> 100

That is not a good metric for call completion.  Shitty quality will not be 
counted in that metric...

but otherwise, yes.  wow.  :-)

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Fwd: meetme conference latency degrades...

[Michael George]

I haven't seen this appear on the list, so I thought I would resend
it...

Sorry for the repost if it did appear before...

- Forwarded message from Michael George  -

Date: Wed, 3 May 2006 21:48:09 -0400
From: Michael George 
Subject: meetme conference latency degrades...
To: asterisk-users@lists.digium.com

We have recently started making more frequent use of the meetme
conference of our * system.

We are using v1.0.8 with a 2.6.11 kernel on our system.

We generally have 4 callers in it: two with the gsm codec and 2 with g729.
Initially, the conference works fine and there is little latency.  After
about 15min., though, the latency is very noticable and by 25min it's
unbearable.

If we all leave the conference and return, the latency is unnoticable
again.

The load on the box is minimal, and only our meetme is running most of
the time.  Checking system load with top shows 0.1 or less.

We have no digium hardware and use ztdummy for our timing device.
zttest yields results generally in the area of 99.96%, but about 3-4%
will be as low as 95%.

In much smaller systems with Digium hardware, the accuracy is never
below 99.98% and is often 100%.

Is this apparent inaccuracy of the ztdummy timer likely the cause of the
increasing latency in our meetme conference? 

Is there any way to improve it?

Thank you, in advance, for any help.

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Tool for Polycom configurations

[Bruce Reeves]

Thanks for all the help! This should go alot quicker then by hand.On 5/4/06, Chad Osmond <[EMAIL PROTECTED]
> wrote:




You can use my script, based on Chris Mason's script, to do most of what 
you want, you can feed it your MAC's and Extensions and it will create the 
phones. 
 
Be warned, it's not pretty, my perl book was in storage so I did a lot of 
kludging. Feel fee to update. 
 
http://holburn.com/poly/poly-add-phone.pl

 
 
Chad



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] On Behalf Of Bruce 
ReevesSent: May 4, 2006 2:45 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Tool for 
Polycom configurations
I am getting read to roll out close to 100 polycom phones and 
wondered if any one knows of a program to take a list of MAC addresses, 
extensions, and names and generate the configuration files?

___--Bandwidth and Colocation provided by Easynews.com
 --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   
http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREA CHABLE?

[Colin Anderson]

>Since January I've passed over 37000 calls through these boxes.  

SELECT count( calldate ) 
FROM `cdr` 
WHERE calldate
BETWEEN '2006-01-01 00:00:01' AND '2006-05-04 15:51:00' AND channel
LIKE '%IAX2%' OR calldate
BETWEEN '2006-01-01 00:00:01' AND '2006-05-04 15:51:00' AND dstchannel
LIKE '%IAX2%' 

count(calldate):
335,761

SELECT count( calldate ) 
FROM `cdr` 
WHERE calldate
BETWEEN '2006-01-01 00:00:01' AND '2006-05-04 15:51:00' AND channel
LIKE '%IAX2%' AND duration < 1 OR calldate
BETWEEN '2006-01-01 00:00:01' AND '2006-05-04 15:51:00' AND dstchannel
LIKE '%IAX2%' AND duration < 1

count(calldate):
100

(100/335761)*100=0.029783089757297601567781844824146

(we'll round up, 0.0298)

100 - 0.0298 = 99.9702% call completion rate. *very* close to five nines.

AFAIC, IAX2 friggin *rocks*. Implementation is *everything* , though. 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Tom Engleward]

--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
wrote:
> Are you specifying the remote Asterisk box by IP or
> by hostname.  If by 
> hostname, then specify it by IP.  Asterisk's DNS
> lookup support has issues.
In the trunk peer details in AMP I'd set "host=" to a
hostname. I've switched it to IP address, and will see
whether that makes a difference. Should I switch from
hostname to IP address in the register string too?

> 2) What is your qualify= set to.  Set it to "yes"
> (2000), or don't set 
> it at all.  Also look at the qualify smoothing
> options in iax.conf.sample.
I don't have qualify set. I'd previously tried setting
it in an attempt to solve the unreliability problem,
at which point asterisk promptly stopped receiving
calls and refused to work again even after a reload,
so I unset qualify, did a reload, and asterisk started
receiving calls again. I recall seeing on various list
archives (maybe this list; don't remember) that other
people had had problems with qualify and iax2 in the
past too, so I gave up on qualify.
Hoping that the hostname vs. IP address issue is the
problem, now that I've switched to IP address, if I
get asterisk to consistently receive calls for 48
hours without needing a reload, I'll post a "success"
message to this list.


__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Andrew Kohlsmith]

On Thursday 04 May 2006 17:24, Tom Engleward wrote:
> You said I should help rather than whine. Am I not
> helping by announcing that I, too, am experiencing a
> problem which somebody else has mentioned, thus
> providing verification that the problem isn't just an
> isolated quirk caused by somebody's particular
> incorrect configuration?

That was helpful.  The rant definitely was not.

> Besides all this, I still think my suggestion is valid
> that consideration be given to adding a cron job to
> asterisk to periodically place an incoming call and
> issue a "reload" if the call fails. It's standard in
> the software industry to have such watchdog timers on
> mission-critical software.

I disagree -- I design mission-critical hardware and software for a living 
(industrial motion controllers: soft starters, variable-frequency drives, 
etc.) -- YES there are watchdog timers there for various things but if one 
ever trips off the system is torn apart to discover WHY and make sure it does 
not happen again if it was something avoidable.  Restarting a PBX is not 
something I would ever consider acceptable.  It's like that "wctdm is going 
squirrely, I just reboot it once a night to keep it from happenning" -- no 
that's not acceptable.  We need the problem labbed up and testing done to 
determine the root cause and eliminate it, not do the old Windows trick of 
"reboot and see if that fixes it" -- when you reboot (or restart in this 
case) you waste time and flush away any data you could have gathered.

My suggestion for you, if possible, is to turn on ethereal or tcpdump on one 
of the afflicted machines and let it run during the quiet times, or during 
the times it's most likely to happen (if you can stand the size of the data 
dumps) and see if you can see one box simply not replying to IAX2 PING 
messages.  

If you can't handle the data dump, add some debugging messages to both boxes' 
chan_iax2.c that prints a timestamp along with "SENDING PING" "RECEIVED PING" 
and "SENDING PONG" -- maybe it's chan_iax2, maybe it's the kernel dropping 
the packet (doubtful) and maybe it's the network dropping the packet (again 
doubtful on a LAN)...  but without data it's impossible to tell and 
impossible to fix.

> If you're inclined to respond "ok, then just add this
> feature yourself," please say so only if you happen to
> agree that such a feature would actually be a good
> idea. In other words, I'm including a question in this

No I do not believe a nightly reboot is EVER a good answer, and it's never 
ever a solution.  It's a band-aid.  It's like taking tylenol every time you 
have a headache, and never trying to figure out why you have a headache every 
single day.

> message: "would it be a good idea to add a watchdog to
> reload iax2 when it fails?"

IMO, no, for the reasons above.

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 hostbecomes UNREACHABLE?

[Vahan Yerkanian]

On Thu, 4 May 2006 23:29:37 +0200, Louis-David Mitterrand <[EMAIL PROTECTED]> 
wrote:
> On Thu, May 04, 2006 at 10:31:17PM +0500, Vahan Yerkanian wrote:
>> Same, here, two asterisk 1.2.7.1 boxes connected to the same switch...
>> Over a week I see at least one case of one of the boxes becoming
>> unavailable for the other... simple iax2 reload fixes the problem.
> 
> Is SIP between two asterisk boxes more reliable? Has someone tried it?

Yes it is.


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREA CHABLE?

[Colin Anderson]

> Is anybody on this list actually using iax2 for
> anything mission-critical?

Yes. 2K inbound / outbound calls a day to 30 remote locations, aggregated to
2 PRI's tied together with IAX2. All with IP address specified rather than
hostname. All with Asterisk 1.0.9. All with 99.9% completion rate, and it
would be 99.999% if we weren't using consumer grade DOCSIS cable modems in
the remote locations.It is quite possible to make a nice Asterisk install if
you are conservative in your approach, and upgrading to CVS-HEAD (or
whatever it's called these days) every night is *not* the way to do it (I'm
sure you aren't doing that, but my point is that with bleeding edge, you,
well, bleed.)

It spooks me to think of how insanely great Asterisk will be in 2 years once
the dev team is through this 1.2 - 1.3 - 1.4 - 1.X period and we can get our
hands on a stable, bugfixed, and mature 2.0 release. Until then, 1.0.9 is
for me. 

go devs go!
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Louis-David Mitterrand]

On Thu, May 04, 2006 at 10:31:17PM +0500, Vahan Yerkanian wrote:
> Andrew Kohlsmith wrote:
> >On Thursday 04 May 2006 11:31, Louis-David Mitterrand wrote:
> >>I've got this low-ping 100%-up dsl connection between two asterisk
> >>1.2.7.1 servers. And oftentimes one of them would declare its opposite
> >>UNREACHABLE.
> 
> Same, here, two asterisk 1.2.7.1 boxes connected to the same switch... 
> Over a week I see at least one case of one of the boxes becoming 
> unavailable for the other... simple iax2 reload fixes the problem.

Is SIP between two asterisk boxes more reliable? Has someone tried it?
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Tom Engleward]

--- Andrew Kohlsmith <[EMAIL PROTECTED]>
wrote:
> No, you are supposed to realize that a) this
> software cost you nothing.  Not 
> one penny.  b) this software is user-supported. 
> This means that in order to 
> make it better you need to help.  and c) we don't
> owe you anything.  Not a 
> thing.
I clearly acknowledged this in my original message.

> In either case.  you may NOT bitch and whine
I did label my message as a rant. I didn't claim that
it wasn't.

> about
> how unacceptable it is.
I didn't say "unacceptable". I said "ridiculous".
There's a difference. I would only say "unacceptable"
if I were paying for it.
I have a grievance against the software, but I don't
have any grievance against the programmers. I would
only have a grievance against the programmers if I
were paying for the software (the grievance would be
that they weren't doing what I paid them to do).

You said I should help rather than whine. Am I not
helping by announcing that I, too, am experiencing a
problem which somebody else has mentioned, thus
providing verification that the problem isn't just an
isolated quirk caused by somebody's particular
incorrect configuration?

Besides all this, I still think my suggestion is valid
that consideration be given to adding a cron job to
asterisk to periodically place an incoming call and
issue a "reload" if the call fails. It's standard in
the software industry to have such watchdog timers on
mission-critical software.
If you're inclined to respond "ok, then just add this
feature yourself," please say so only if you happen to
agree that such a feature would actually be a good
idea. In other words, I'm including a question in this
message: "would it be a good idea to add a watchdog to
reload iax2 when it fails?"


__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] asterisk <-> SIP provider, two way connection

[hechang]

Please give me some heads up.
 
I'm having troube setting up my asterisk connecting to my SIP provider (SP). Here's the setup.
 
in sip.conf
I register asterisk to SP using
 
register => 15551234567:[EMAIL PROTECTED]
 
everything was ok for incoming call until I want to dial out using the same line. since in order to dial out, i added in sip.conf
 
[sip_out_to_SP]
type=peerusername=15551234567fromuser=15551234567secret=pwd123host=sip.abc.com
port=5160   
usereqphone=yes call-limit=5 
 
Then I can make phone call out when routed to this channel SIP/sip_out_to_SP. But I no longer able to receive calls. The sniffer told me when my SP send asterisk an invite, asterisk returns a 407 to authenticate SP, which kills the connection.

 
I tried to change "type =friend" but doesn't work. I have to either remove "secret= " or "host=" to make my incoming call back.
 
So currently, I can only either dial out or receive calls on this sip line. not both. (I'm not talking about at the same time). Can anyone give me a hint. REALLY APPRECIATE IT.
 
Michael
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Voipjet Problem?

[Ira]

At 12:24 PM 5/4/2006, you wrote:

Customer
service may be stellar but when clients are actually trying to save money,
that's a damned hard sell.


My first thought too, but somewhere he said, call them for wholesale 
rates which are reasonable.


Ira 


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: Unable to get TDM400p working

[Ben Gore]

Hello again..

Disregard my last message. I recompiled Asterisk and all is well.

I guess all the messing around with trying to get Zaptel to compile 
correctly screwed something up with the Asterisk configuration.


Thanks.

-Ben

Original message:


This has got to be a stupid error I'm making...

I have been experimenting with different hardware and software
configurations before I decide what to use as a production platform. Up
until just recently things were going well.

But now it appears I'm unable to get access to my TDM400p from Asterisk. I
know the TDM card works fine, used it in another machine where it
performed flawlessly. I have been using the same set of conf files, just
copying them over from machine to machine. The hardware is Pentium 4
all-Intel chipset mainboard.

The one difference here is I'm trying CentOS. I've been through the
problems getting Zaptel to compile with the error in spinlock.h. I got
to...

Zaptel Configuration
==


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels configured.

and thought I was home free. Wrong!

zttool recognizes the card properly and reports status "OK"

Asterisk runs and I can make calls on SIP phones with no problems. However
I get no dial tone on the analog phone and outgoing calls through the TDM
(from the SIP phones of course!) produce this on the console:

NOTICE[5934]: app_dial.c:1010 dial_exec_full: Unable to create channel of
type 'Zap' (cause 66 - Channel not implemented)
 == Everyone is busy/congested at this time (1:0/0/1)
   -- Executing Congestion("SIP/100-24d1", "") in new stack
 == Spawn extension (internal, 91234567, 102) exited non-zero on
'SIP/100-24d1'

The conf files are the same as they were on another working machine, I
just copied them over. I'll be going over them /again/ next.

What am I missing?




___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Eric \"ManxPower\" Wieling]

Are you specifying the remote Asterisk box by IP or by hostname.  If by 
hostname, then specify it by IP.  Asterisk's DNS lookup support has issues.


2) What is your qualify= set to.  Set it to "yes" (2000), or don't set 
it at all.  Also look at the qualify smoothing options in iax.conf.sample.


Tom Engleward wrote:



From this thread today I've learned that the problems

I've been having the entire time I've been using
asterisk (about two weeks) stem not from NAT, as I
originally thought, but from asterisk itself, so that
if I were to move my asterisk box to a public IP
address, my iax2 connection to my PSTN originator
(which also runs asterisk) would _still_ be
unreliable.
This makes iax2 on asterisk useless for receiving
calls. No matter how many spiffy features asterisk
has, there is one simple nonnegotiable requirement: it
must always answer incoming calls. If it can't do
that, then it can't be relied on. And over iax2, it
can't do that.
Isn't asterisk supposed to by default reregister iax2
connections every minute or something like that? Why
then do I get reliable incoming connections for
several hours, and then it dies, and I have to do a
"reload"?
Am I supposed to make a cron job to automatically tell
asterisk to reload every so often, since iax2 likes to
periodically die? Or maybe am I supposed to make a
cron job to place a phone call every so often from an
external phone into my asterisk system and verify that
asterisk actually answers, and immediately issue
asterisk a reload if it fails?
This is utterly ridiculous. Yes, I know, it's free
software and all, and "you get what you pay for."
Just in a bad mood today because I've literally lost
thousands of dollars due to asterisk's failure to
reliably answer incoming calls, and I only discover
these failed incoming call attempts later when I check
my PSTN originator's logs. I then go "oh crap!" and do
a test call into my asterisk system, and get Ma Bell's
"the number you are calling has been disconnected or
is no longer in use", and I issue a "reload" to
asterisk and try again, and this time my call
succeeds. At this rate soon it will be more profitable
for me to just invest in a traditional reliable PBX
hooked to Ma Bell and be done with these problems.
I'm not a Digium customer, so they have no reason to
listen to me, but surely there are Digium customers
who are also getting bitten by this iax2 bug.
Is anybody on this list actually using iax2 for
anything mission-critical?









--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Meetme from MySQL

[Richard OSS]

try http://sourceforge.net/projects/web-meetmeChris Blunt <[EMAIL PROTECTED]> wrote:Hi List,      Is it possible to store meetme config in a MySQL table?     If so, any pointers would be appreciated.     Thanks     Chris        --     Chris Blunt  Entropy IT Ltd   ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Voicemail records funny - Asterisk 1.2.7.1

[McQuiggan, Mark xt46480]

I have asterisk 
1.2.7.1 running on Fedora core 5.  Everything looked like it compiled 
OK.
 
When a call is 
bumped to voicemail, the message prompts sound fine to the user.  However, 
when the voice message is retrieved, it sounds "compressed" or speeded 
up.  
 
I have checked this 
against the voicemail of my production version of asterisk (1.2.5, 
running on FC4), and found that the wav file recorded on the new version is 
about 1/5 the size of the wav from the old.
 
Help?
 
Thanks,
 
Mark 
McQuiggan
This message and any attachments are intended only for the use of the addressee and
may contain information that is privileged and confidential. If the reader of the 
message is not the intended recipient or an authorized representative of the
intended recipient, you are hereby notified that any dissemination of this
communication is strictly prohibited. If you have received this communication in
error, please notify us immediately by e-mail and delete the message and any
attachments from your system.

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Unable to get TDM400p working

[Sean Cook]

couple of things... was asterisk compiled after zaptel?  from the cli 
try "load chan_zap.so" and see what you get


Ben Gore wrote:

This has got to be a stupid error I'm making...

I have been experimenting with different hardware and software
configurations before I decide what to use as a production platform. Up
until just recently things were going well.

But now it appears I'm unable to get access to my TDM400p from Asterisk. I
know the TDM card works fine, used it in another machine where it
performed flawlessly. I have been using the same set of conf files, just
copying them over from machine to machine. The hardware is Pentium 4
all-Intel chipset mainboard.

The one difference here is I'm trying CentOS. I've been through the
problems getting Zaptel to compile with the error in spinlock.h. I got
to...

Zaptel Configuration
==


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels configured.

and thought I was home free. Wrong!

zttool recognizes the card properly and reports status "OK"

Asterisk runs and I can make calls on SIP phones with no problems. However
I get no dial tone on the analog phone and outgoing calls through the TDM
(from the SIP phones of course!) produce this on the console:

NOTICE[5934]: app_dial.c:1010 dial_exec_full: Unable to create channel of
type 'Zap' (cause 66 - Channel not implemented)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing Congestion("SIP/100-24d1", "") in new stack
  == Spawn extension (internal, 91234567, 102) exited non-zero on
'SIP/100-24d1'

The conf files are the same as they were on another working machine, I
just copied them over. I'll be going over them /again/ next.

What am I missing?

Thanks.

-Ben





___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
  


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Andrew Kohlsmith]

On Thursday 04 May 2006 15:51, Tom Engleward wrote:
> Am I supposed to make a cron job to automatically tell
> asterisk to reload every so often, since iax2 likes to
> periodically die? Or maybe am I supposed to make a
> cron job to place a phone call every so often from an
> external phone into my asterisk system and verify that
> asterisk actually answers, and immediately issue
> asterisk a reload if it fails?

No, you are supposed to realize that a) this software cost you nothing.  Not 
one penny.  b) this software is user-supported.  This means that in order to 
make it better you need to help.  and c) we don't owe you anything.  Not a 
thing.

If you're not a programmer, you can help with bug reports, packet traces, 
helping us run test cases of fixes, etc.  If you are a programmer, you can 
try to help us figure out what's causing it directly and creating scenarios 
in which it happens.

In either case.  you may NOT bitch and whine about how unacceptable it is.  If 
you want to pay someone to listen to you complain, buy ABE, or go buy your 
father's PBX.

> This is utterly ridiculous. Yes, I know, it's free
> software and all, and "you get what you pay for."
> Just in a bad mood today because I've literally lost
> thousands of dollars due to asterisk's failure to
> reliably answer incoming calls, and I only discover
> these failed incoming call attempts later when I check
> my PSTN originator's logs. I then go "oh crap!" and do
> a test call into my asterisk system, and get Ma Bell's
> "the number you are calling has been disconnected or
> is no longer in use", and I issue a "reload" to
> asterisk and try again, and this time my call
> succeeds. At this rate soon it will be more profitable
> for me to just invest in a traditional reliable PBX
> hooked to Ma Bell and be done with these problems.
> I'm not a Digium customer, so they have no reason to
> listen to me, but surely there are Digium customers
> who are also getting bitten by this iax2 bug.
> Is anybody on this list actually using iax2 for
> anything mission-critical?

I use Asterisk for my company's phone system.  EVERY call, and I mean every 
one (faxes too) passes through two Asterisk boxes.  One connected to our PRI 
downtown, and one connected to our Norstar here at the office.  Since January 
I've passed over 37000 calls through these boxes.  Yes, we've had bad days.  
We've had days where it's crashed and we've dropped every call in progress.  
We also run the svn trunk (i.e. bleeding edge) code, which is both a blessing 
and a curse.  :-)  Overall though, it has worked VERY well for us, and we're 
just starting to scratch the surface of the capabilities I have sold this 
solution on.  My customers, however, know and understand that this is new and 
will have some hiccups.  I try to minimize it, of course, but it's an 
inevitability.

If it's mission critical, you should also have the facilities to handle 
failover and clustering.  If it's mission critical, where was your Nagios or 
other network monitor watching, placing test calls and paging you whenever 
this happened?  If this is mission critical, where is your contingency plan?  
Losing "thousands of dollars" is never fun, but if you're not willing to help 
with the bugfixes and throw some wrench time at the problem then you should 
not be trying to sell a solution using Asterisk.  It really is that simple.  
That'd be equivalent to someone not knowing a socket wrench from a wing-nut 
and opening up his own full-service auto garage.

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Tool for Polycom configurations

[Chad Osmond]

You can use my script, based on Chris Mason's script, to do most of what 
you want, you can feed it your MAC's and Extensions and it will create the 
phones. 
 
Be warned, it's not pretty, my perl book was in storage so I did a lot of 
kludging. Feel fee to update. 
 
http://holburn.com/poly/poly-add-phone.pl
 
 
Chad



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce 
ReevesSent: May 4, 2006 2:45 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Tool for 
Polycom configurations
I am getting read to roll out close to 100 polycom phones and 
wondered if any one knows of a program to take a list of MAC addresses, 
extensions, and names and generate the configuration files?
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Volume configuration on Polycom Soundpoint 501 phone

[Jerry Jones]

Edit your config files to enable persistance

Will remain across multiple calls, but not reboots


On May 4, 2006, at 2:51 PM, Jim Freeze wrote:


We are using the polycom 501 phones, and are having some challenges
with the volume setting. When a phone call comes in, the user ups the
volume for the handset, but they have to repeat that for every call.

Currently, the volume level seems to reset itself at about 60%.
Is there a way for the user to change their default volume level?

Thanks

--
Jim Freeze
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Help with IRQ conflict between wct2xxp and eth0

[Phil Menico]

Title: Message




I have a conflict problem with the eth0 
card and wct2xxp digium board. The PRI can 
receive calls but my network connection is gone.
When I "cat /proc/interrupts" I get the 
following:
1 
..
1 ..
..
..
..
169 0 IO-APIC-level wct2xxp, 
eth0
..
etc.
even before I "modprobe 
wct2xxp"
After I "modprobe wct2xxp" and 
"modprobe wctdm" and again run "cat /proc/interrupts"
I then get:
 
..
..
..
..
..
169 118489 IO-APIC-level wct2xxp, 
eth0
201 118497 IO-APIC-level wctdm 

..
etc
 
How can I force the wct2xxp to load on 
a separate IRQ? I tried moving the eth0 to IRQ 10 but could not.
Any ideas?
 
 
Thank you.
Phil Menico 

XTEND Communications 171 Madison 
Avenue, New York, NY 10016 212-951-7632 
(Office) 212-951-7683 (Fax) www.xtend.com 

  
   
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Unable to get TDM400p working

[Ben Gore]

This has got to be a stupid error I'm making...

I have been experimenting with different hardware and software
configurations before I decide what to use as a production platform. Up
until just recently things were going well.

But now it appears I'm unable to get access to my TDM400p from Asterisk. I
know the TDM card works fine, used it in another machine where it
performed flawlessly. I have been using the same set of conf files, just
copying them over from machine to machine. The hardware is Pentium 4
all-Intel chipset mainboard.

The one difference here is I'm trying CentOS. I've been through the
problems getting Zaptel to compile with the error in spinlock.h. I got
to...

Zaptel Configuration
==


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels configured.

and thought I was home free. Wrong!

zttool recognizes the card properly and reports status "OK"

Asterisk runs and I can make calls on SIP phones with no problems. However
I get no dial tone on the analog phone and outgoing calls through the TDM
(from the SIP phones of course!) produce this on the console:

NOTICE[5934]: app_dial.c:1010 dial_exec_full: Unable to create channel of
type 'Zap' (cause 66 - Channel not implemented)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing Congestion("SIP/100-24d1", "") in new stack
  == Spawn extension (internal, 91234567, 102) exited non-zero on
'SIP/100-24d1'

The conf files are the same as they were on another working machine, I
just copied them over. I'll be going over them /again/ next.

What am I missing?

Thanks.

-Ben





___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Volume configuration on Polycom Soundpoint 501 phone

[Sean Cook]

sip.cfg

voice.volume.persist.headset="1" voice.volume.persist.handsfree="1"/>


Jim Freeze wrote:

We are using the polycom 501 phones, and are having some challenges
with the volume setting. When a phone call comes in, the user ups the
volume for the handset, but they have to repeat that for every call.

Currently, the volume level seems to reset itself at about 60%.
Is there a way for the user to change their default volume level?

Thanks

--
Jim Freeze


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
  


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] ISAC support?

[Trond G. Andersen]

That is what I thought too, but what about this:

http://lists.digium.com/pipermail/asterisk-commits/2006-February/001461.
html

???

 

>No. iSAC is a codec from GIPS. Likely the coded used by Skype.
>
>Michael
>
>On Thu, 4 May 2006 21:35:07 +1000, James Harper wrote:
>
>---
>I assume you mean this:
>http://en.wikipedia.org/wiki/ISAC
>
>but maybe you are referring to one of the controller chips on BRI 
>adapters?
>
>James
>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-

>> [EMAIL PROTECTED] On Behalf Of Trond G. Andersen
>> Sent: Thursday, 4 May 2006 20:19
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [Asterisk-Users] ISAC support?
>> 
>> Hi All.
>> 
>> Has there been done any work to support ISAC ?
>> 
>> 
>> Thanks,
>> trond
>> ___
>> --Bandwidth and Colocation provided by Easynews.com --
>> 
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>___
>--Bandwidth and Colocation provided by Easynews.com --
>
>Asterisk-Users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Tool for Polycom configurations

[Anthony Rodgers]

Hi Bruce,

We create a CSV file of our phone setup and then use shell scripts to 
parse them and generate .cfg, phone.cfg, sip.conf, 
voicemail.conf and entensions.conf entries.


Contact me off list if you would like a copy now (they're not quite 
ready for prime-time yet) - the rest of you will have to wait until 
they're finished :-) but I do intend to release a bunch of monkey-level 
helpdesk scripts that I am working on in the near future for managing 
basic MAC requests.


Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On May 4, 2006, at 11:45 AM, Bruce Reeves wrote:

 I am getting read to roll out close to 100 polycom phones and 
wondered if any one knows of a program to take a list of MAC 
addresses, extensions, and names and generate the configuration files?


--
Bruce
Nortex Networks___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: Auto Logout from queue

[Matt]

Hrmmm.
I thought there was already an option in the queue.conf or agents.conf
file (Though can't remember off hand what) that would set an agent
logged out or on 'pause' if they did not answer a call.  No?

On 5/4/06, Christopher Mayfield <[EMAIL PROTECTED]> wrote:

it is two scripts an empty_queue.sh and a fill_queue.sh and a members script
If you need intructions please tell me

1047 $  cat empty_queue.sh
#!/bin/bash

# a script to remove everyone in the members script located in the same
directory as this file
# to the Q 3901
# can be called from a script

#Local/[EMAIL PROTECTED]
#Local/[EMAIL PROTECTED]
#Local/[EMAIL PROTECTED]

for wild in `/usr/sbin/asterisk -r -x "show queue 3901"| grep -a dynamic |
awk '{ print $1 }'`;
do
/usr/sbin/asterisk -r -x "remove queue member "$wild" from 3901"|
grep -a interface;
done

1048 $  cat fill_queue.sh
#!/bin/bash

# a script to add everyone in the members script located in the same
directory as this file
# to the Q 3901
# can be called from a script

#Local/[EMAIL PROTECTED]
#Local/[EMAIL PROTECTED]
#Local/[EMAIL PROTECTED]

for wild in `cat /home/cmayfield/members `;
do
/usr/sbin/asterisk -r -x "add queue member "$wild" to 3901"| grep -a
interface;
done

1049 $  cat members
Local/[EMAIL PROTECTED]
Local/[EMAIL PROTECTED]

1050 $  crontab -l
#at 8:1 and 8:11 it will fill the queue and it is nondistructive
1,11 8   *   *   1-5   sh /home/cmayfield/fill_queue.sh
| mail -s "fill_queue" [EMAIL PROTECTED]
#at 5:31 and 5:36 it will empty the queue and it is nondistructive
31,3617   *   *   1-5   sh
/home/cmayfield/empty_queue.sh | mail -s "empty_queue" [EMAIL PROTECTED]


On 5/2/06, Tomislav Parčina <[EMAIL PROTECTED]> wrote:
> In article <
[EMAIL PROTECTED]>,
[EMAIL PROTECTED] says...
> > that is a nice function
> > I use a cronjob to logout everyone each evening if anyone wants that
script
> > I would love to provide it.
>
> Please send the script to the list.
>
>
>
> --
> Tomislav Parčina
> Lama Computers Split
> Stinice 12, 21000 Split
> Tel.: +385(21)495148
> SIP: [EMAIL PROTECTED]
> e-mail: tparcina#lama.hr
> http://www.lama.hr
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>
http://lists.digium.com/mailman/listinfo/asterisk-users
>



--
$15.95/Month DreamHost Hosting SALE
60 GB Disk Storage, 1.6 TB Transfer
Transfer Increases 16 GB Weekly
http://www.dreamhost.com/r.cgi?ap0ught/shared/
Use discount code caralena for $40 off all these low prices
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Tom Engleward]

--- Vahan Yerkanian <[EMAIL PROTECTED]> wrote:
> Andrew Kohlsmith wrote:
> > On Thursday 04 May 2006 11:31, Louis-David
> Mitterrand wrote:
> >> I've got this low-ping 100%-up dsl connection
> between two asterisk
> >> 1.2.7.1 servers. And oftentimes one of them would
> declare its opposite
> >> UNREACHABLE.
> 
> Same, here, two asterisk 1.2.7.1 boxes connected to
> the same switch... 
> Over a week I see at least one case of one of the
> boxes becoming 
> unavailable for the other... simple iax2 reload
> fixes the problem.
> 
> Been like this for ages.


>From this thread today I've learned that the problems
I've been having the entire time I've been using
asterisk (about two weeks) stem not from NAT, as I
originally thought, but from asterisk itself, so that
if I were to move my asterisk box to a public IP
address, my iax2 connection to my PSTN originator
(which also runs asterisk) would _still_ be
unreliable.
This makes iax2 on asterisk useless for receiving
calls. No matter how many spiffy features asterisk
has, there is one simple nonnegotiable requirement: it
must always answer incoming calls. If it can't do
that, then it can't be relied on. And over iax2, it
can't do that.
Isn't asterisk supposed to by default reregister iax2
connections every minute or something like that? Why
then do I get reliable incoming connections for
several hours, and then it dies, and I have to do a
"reload"?
Am I supposed to make a cron job to automatically tell
asterisk to reload every so often, since iax2 likes to
periodically die? Or maybe am I supposed to make a
cron job to place a phone call every so often from an
external phone into my asterisk system and verify that
asterisk actually answers, and immediately issue
asterisk a reload if it fails?
This is utterly ridiculous. Yes, I know, it's free
software and all, and "you get what you pay for."
Just in a bad mood today because I've literally lost
thousands of dollars due to asterisk's failure to
reliably answer incoming calls, and I only discover
these failed incoming call attempts later when I check
my PSTN originator's logs. I then go "oh crap!" and do
a test call into my asterisk system, and get Ma Bell's
"the number you are calling has been disconnected or
is no longer in use", and I issue a "reload" to
asterisk and try again, and this time my call
succeeds. At this rate soon it will be more profitable
for me to just invest in a traditional reliable PBX
hooked to Ma Bell and be done with these problems.
I'm not a Digium customer, so they have no reason to
listen to me, but surely there are Digium customers
who are also getting bitten by this iax2 bug.
Is anybody on this list actually using iax2 for
anything mission-critical?



__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Volume configuration on Polycom Soundpoint 501 phone

[Jim Freeze]

We are using the polycom 501 phones, and are having some challengeswith the volume setting. When a phone call comes in, the user ups thevolume for the handset, but they have to repeat that for every call.Currently, the volume level seems to reset itself at about 60%.
Is there a way for the user to change their default volume level?Thanks-- Jim Freeze
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Voipjet Problem?

[Matt]

Kerry,
You didn't read my entire e-mail.   How do I know that?   Because if
you re-read it you'll see that I state:
"If you are a wholesole buyer of minutes, talk to them, don't just
take their prices on the main page... those are for residential and
regular customers.  Their prices are very comparable to voipjet, and
the service is miles ahead."

If you are a regular residential customer just wanting to do talking
$7 + 3.9cents/minute may very well be cheaper then your $50.00/month
phone bill with Verizon, or BellSouth.   However, commercial
termination customers will get much better rates, and MUCH better
service then voipjet provides.


On 5/4/06, Kerry Garrison <[EMAIL PROTECTED]> wrote:

Hard to believe you arent associated with calleveryone.com as I find it hard
to believe that you would be extolling the virtues on one of, if not the
most expensive companies around. $7 a month plus 3.9 cents a minute
domestic, that's pretty much double the cost of anyone else. Customer
service may be stellar but when clients are actually trying to save money,
that's a damned hard sell.

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: Thursday, May 04, 2006 8:18 AM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Voipjet Problem?
>
> Just wanted to add my 2 cents.  We were with voipjet, and do still use
> them for occassional backup.However, their lack of personal
> service and inability to get ahold of someone drove us away.After
> several total blackouts (like what happened yesterday), and
> no responce we finally put out an SOS on the asterisk mailing
> list.  Of course there were several responces from companies
> trying to solicit us. but the one that caught our
> attention was calleveryone.com
> So far we have been rock-solid-happy with them.   We've had a few
> small bumps along the road.   For instance, once there was a router
> along our path to them that was dropping packets, but this was quickly
> resolved.   Additionally, they've worked with us on the phone to
> resolve audio problems, and diagnose carrier issues.   If I have a
> problem, I rest assured that I can call someone, or page
> someone if the situation is severe enough, and get ahold of a
> human at any hour
> of the evening.   Not so with VoipJet.   I don't want to bad mouth
> VoipJet, their service is decent... but definately not acceptable for
> a carrier grade level.   I'm not affiiliated with calleveryone in any
> way other then a very happy and satisfied customer, and would highly
> recommend them to you.   If you are a wholesole buyer of minutes, talk
> to them, don't just take their prices on the main page...
> those are for residential and regular customers.  Their
> prices are very comparable to voipjet, and the service is miles ahead.
>
> On 5/3/06, Matt <[EMAIL PROTECTED]> wrote:
> > Yup... I think they died... this is why I stopped using
> them except as
> > my backup.   It seems 64.34.45.100  is working ok as of right now.
> > It wouldn't be so bad if they had a number you could call
> for support!
> >  HERE THAT JOHN?   You need a phone number if you want to "play with
> > the big dogs".
> >
> > On 5/3/06, Mark Hulber <[EMAIL PROTECTED]> wrote:
> > > I started to have a problem today that all my calls
> through voipjet
> > > result in just timing out after my assigned timeout
> period.  I tried
> > > multiple of their servers with the same problem.  Anyone
> else having
> > > a problem?  I am running:
> > >
> > > Asterisk SVN-branch-1.2-r24381M built by root @
> asterisk.hulber.com
> > > on a
> > > i686 running Linux on 2006-05-03 14:14:07 UTC
> > >
> > > I can connect with other IAX providers.
> > >
> > > MARK.
> > > ___
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

[Scott Gifford]

Colin Anderson <[EMAIL PROTECTED]> writes:

>>Why is this hard to fake at all?  You send a different fax to your
>>system, and replace the Asterisk audio file with the one from the
>>altered fax.  Additionally, the client has no realistic way of
>>verifying the correctness of your audio-to-fax translation tool; it
>>could just as easily output a TIFF file completely different from the
>>one that was actually faxed.
>
> That's interesting, I hadn't thought of it that way. I was thinking in terms
> of subtly modifying the original audio stream not outright replacing the
> recording and faking the datestamp! Given that, essentially recording the
> audio is the *same* as retaining the TIFF in terms of integrity
> vulnerability. 
>
> How about this: (theoretical of course)
>
> 1. Fax comes in
> 2. Audio is recorded
> 3. A checksum of the audio is generated then relayed somehow to a seperate,
> secure system
> 4. In the event of a dispute, the checksum is retrieved, compared with the
> original audio file, then the original audio is "replayed" and the fax is
> regenerated.

I don't see the advantage to this; the client still has to trust that
all of this is done correctly, and if they don't trust the fax
recipient to put the correct fax in the paper file or keep the correct
TIFF, why would they trust them to do this?

Using a third party to receive and relay the fax, one which is trusted
by both the client and the fax recipient, would solve the problem; the
third party could create a document with the caller information
(ideally from ANI, which is harder to forge), the time, and the
message itself, then digitally sign it.  This might even be an
interesting business plan, for some applications where confirmed
document transmittal is important.

But it's hard for me to imagine this isn't overkill; if a client and a
service provider distrust each other so thoroughly that they have to
communicate through a third party to verify integrity, probably they
just shouldn't do business with each other.

Scott.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: Auto Logout from queue

[Christopher Mayfield]

it is two scripts an empty_queue.sh and a fill_queue.sh and a members script If you need intructions please tell me1047 $  cat empty_queue.sh#!/bin/bash# a script to remove everyone in the members script located in the same directory as this file
# to the Q 3901# can be called from a script#Local/[EMAIL PROTECTED]#Local/[EMAIL PROTECTED]#Local/[EMAIL PROTECTED]for wild in `/usr/sbin/asterisk -r -x "show queue 3901"| grep -a dynamic | awk '{ print $1 }'`;
    do    /usr/sbin/asterisk -r -x "remove queue member "$wild" from 3901"| grep -a interface;    done1048 $  cat fill_queue.sh#!/bin/bash# a script to add everyone in the members script located in the same directory as this file
# to the Q 3901# can be called from a script#Local/[EMAIL PROTECTED]#Local/[EMAIL PROTECTED]#Local/[EMAIL PROTECTED]for wild in `cat /home/cmayfield/members `;    do    /usr/sbin/asterisk -r -x "add queue member "$wild" to 3901"| grep -a interface;
    done1049 $  cat membersLocal/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]1050 $  crontab -l#at 8:1 and 8:11 it will fill the queue and it is nondistructive1,11 8   *   *   1-5   sh /home/cmayfield/fill_queue.sh | mail -s "fill_queue" 
[EMAIL PROTECTED]#at 5:31 and 5:36 it will empty the queue and it is nondistructive31,36    17   *   *   1-5   sh /home/cmayfield/empty_queue.sh | mail -s "empty_queue" 
[EMAIL PROTECTED]On 5/2/06, Tomislav Parčina
 <[EMAIL PROTECTED]> wrote:In article <
[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> that is a nice function> I use a cronjob to logout everyone each evening if anyone wants that script> I would love to provide it.Please send the script to the list.--Tomislav Parčina
Lama Computers SplitStinice 12, 21000 SplitTel.: +385(21)495148SIP: [EMAIL PROTECTED]e-mail: tparcina#lama.hrhttp://www.lama.hr
___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users-- $15.95/Month DreamHost Hosting SALE
60 GB Disk Storage, 1.6 TB TransferTransfer Increases 16 GB Weeklyhttp://www.dreamhost.com/r.cgi?ap0ught/shared/Use discount code caralena for $40 off all these low prices
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Tool for Polycom configurations

[Andrew Kohlsmith]

On Thursday 04 May 2006 15:18, Sean Cook wrote:
> http://www.freedomphones.net/polycom/files/polycom.phone1cfg.pl-script

Yep that's the one that reads sip.conf and spits out phone[exten].cfg files.  
It does not tie in mac addresses nor generate [macaddress].cfg files, though.

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

[Josh McAllister]

Sounds like a potential business opportunity. Someone could setup a fax
proxy service that provides this sort of digital signing / archiving.
The originator could simply dial a toll-free access number, receive a
2nd dialtone and then dial the destination. Meanwhile the proxy is
recording the call, then decoding and allowing the archives to be viewed
online along with all relevent call details.

Hmm... Interesting.

Josh McAllister

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: Thursday, May 04, 2006 1:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

>Why is this hard to fake at all?  You send a different fax to your 
>system, and replace the Asterisk audio file with the one from the 
>altered fax.  Additionally, the client has no realistic way of 
>verifying the correctness of your audio-to-fax translation tool; it 
>could just as easily output a TIFF file completely different from the 
>one that was actually faxed.

That's interesting, I hadn't thought of it that way. I was thinking in
terms of subtly modifying the original audio stream not outright
replacing the recording and faking the datestamp! Given that,
essentially recording the audio is the *same* as retaining the TIFF in
terms of integrity vulnerability. 

How about this: (theoretical of course)

1. Fax comes in
2. Audio is recorded
3. A checksum of the audio is generated then relayed somehow to a
seperate, secure system 4. In the event of a dispute, the checksum is
retrieved, compared with the original audio file, then the original
audio is "replayed" and the fax is regenerated.

The 3. part I leave as an exercise for the reader.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Voipjet Problem?

[Kerry Garrison]

Hard to believe you arent associated with calleveryone.com as I find it hard
to believe that you would be extolling the virtues on one of, if not the
most expensive companies around. $7 a month plus 3.9 cents a minute
domestic, that's pretty much double the cost of anyone else. Customer
service may be stellar but when clients are actually trying to save money,
that's a damned hard sell.

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: Thursday, May 04, 2006 8:18 AM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Voipjet Problem?
> 
> Just wanted to add my 2 cents.  We were with voipjet, and do still use
> them for occassional backup.However, their lack of personal
> service and inability to get ahold of someone drove us away.After
> several total blackouts (like what happened yesterday), and 
> no responce we finally put out an SOS on the asterisk mailing 
> list.  Of course there were several responces from companies 
> trying to solicit us. but the one that caught our 
> attention was calleveryone.com  
> So far we have been rock-solid-happy with them.   We've had a few
> small bumps along the road.   For instance, once there was a router
> along our path to them that was dropping packets, but this was quickly
> resolved.   Additionally, they've worked with us on the phone to
> resolve audio problems, and diagnose carrier issues.   If I have a
> problem, I rest assured that I can call someone, or page 
> someone if the situation is severe enough, and get ahold of a 
> human at any hour
> of the evening.   Not so with VoipJet.   I don't want to bad mouth
> VoipJet, their service is decent... but definately not acceptable for
> a carrier grade level.   I'm not affiiliated with calleveryone in any
> way other then a very happy and satisfied customer, and would highly
> recommend them to you.   If you are a wholesole buyer of minutes, talk
> to them, don't just take their prices on the main page... 
> those are for residential and regular customers.  Their 
> prices are very comparable to voipjet, and the service is miles ahead.
> 
> On 5/3/06, Matt <[EMAIL PROTECTED]> wrote:
> > Yup... I think they died... this is why I stopped using 
> them except as
> > my backup.   It seems 64.34.45.100  is working ok as of right now.
> > It wouldn't be so bad if they had a number you could call 
> for support!
> >  HERE THAT JOHN?   You need a phone number if you want to "play with
> > the big dogs".
> >
> > On 5/3/06, Mark Hulber <[EMAIL PROTECTED]> wrote:
> > > I started to have a problem today that all my calls 
> through voipjet 
> > > result in just timing out after my assigned timeout 
> period.  I tried 
> > > multiple of their servers with the same problem.  Anyone 
> else having 
> > > a problem?  I am running:
> > >
> > > Asterisk SVN-branch-1.2-r24381M built by root @ 
> asterisk.hulber.com 
> > > on a
> > > i686 running Linux on 2006-05-03 14:14:07 UTC
> > >
> > > I can connect with other IAX providers.
> > >
> > > MARK.
> > > ___
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Tool for Polycom configurations

[The VoIP Connection]

Hi Bruce,
 
We've written software to do this as a service for our 
customers.  I can't give you the program, but we'd be willing to program 
your phones for you.  Contact me off list.
 
Michael Crown Managing Partner 
www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] 

  
  
  From: Bruce Reeves 
  [mailto:[EMAIL PROTECTED] Sent: Thursday, May 04, 2006 
  2:45 PMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] Tool for Polycom configurations
  I am getting read to roll out close to 100 polycom phones and 
  wondered if any one knows of a program to take a list of MAC addresses, 
  extensions, and names and generate the configuration files?-- Bruce Nortex Networks 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Tool for Polycom configurations

[Mojo with Horan & Company, LLC]

Something I made might help. 
http://www.horanappraisals.com/asterisk/polycom_addphone/  -- there is a 
script, addphone, and a folder called defaults that contains the 
templates.


To use, I put the defaults folder and its contents and the addphone 
script in my ftp or tftp root.  I would make sure that phoneX.cfg 
contains the proper reg password.  and make sure 
-directory.xml contains the global dir you want all phones 
to begin with.


Then, from the (t)ftp root, run

addone macaddress extension display_name
i.e.:
addone 001122334455 110 Mojo

the results of this would be:

Creating 001122334455.cfg to point to extension 110
Creating phone110.cfg for extension 110, DisplayName Mojo, to point to 
mac address 001122334455

Creating 001122334455-directory.xml from default company directory
Done!

Any questions feel free to ask me off-list.

Moj



Bruce Reeves wrote:
I am getting read to roll out close to 100 polycom phones and wondered 
if any one knows of a program to take a list of MAC addresses, 
extensions, and names and generate the configuration files?


--
Bruce
Nortex Networks




___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Tool for Polycom configurations

[Sean Cook]

Try this one:

http://www.freedomphones.net/polycom/files/polycom.phone1cfg.pl-script

Sean

Andrew Kohlsmith wrote:

On Thursday 04 May 2006 14:45, Bruce Reeves wrote:
  

I am getting read to roll out close to 100 polycom phones and wondered if
any one knows of a program to take a list of MAC addresses, extensions, and
names and generate the configuration files?



You can do this relatively easily with Perl.  There is a script somewhere that 
will take your sip.conf and generate phone[exten].cfg files, but it knows 
nothing about MAC addresses and as such will not generate the 
[MACADDRESS].cfg files.


Again though, this isn't too tricky to do.  A few hours' worth of work.  The 
tricky part would be making sure you got the right phone to the right desk if 
the extension #s are physically important.  :-)


If you need some help with the script I am available for consulting.  Contact 
me offlist if this is something you'd like to discuss.


-A.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
  


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

[Colin Anderson]

>Why is this hard to fake at all?  You send a different fax to your
>system, and replace the Asterisk audio file with the one from the
>altered fax.  Additionally, the client has no realistic way of
>verifying the correctness of your audio-to-fax translation tool; it
>could just as easily output a TIFF file completely different from the
>one that was actually faxed.

That's interesting, I hadn't thought of it that way. I was thinking in terms
of subtly modifying the original audio stream not outright replacing the
recording and faking the datestamp! Given that, essentially recording the
audio is the *same* as retaining the TIFF in terms of integrity
vulnerability. 

How about this: (theoretical of course)

1. Fax comes in
2. Audio is recorded
3. A checksum of the audio is generated then relayed somehow to a seperate,
secure system
4. In the event of a dispute, the checksum is retrieved, compared with the
original audio file, then the original audio is "replayed" and the fax is
regenerated.

The 3. part I leave as an exercise for the reader.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Tool for Polycom configurations

[Andrew Kohlsmith]

On Thursday 04 May 2006 14:45, Bruce Reeves wrote:
> I am getting read to roll out close to 100 polycom phones and wondered if
> any one knows of a program to take a list of MAC addresses, extensions, and
> names and generate the configuration files?

You can do this relatively easily with Perl.  There is a script somewhere that 
will take your sip.conf and generate phone[exten].cfg files, but it knows 
nothing about MAC addresses and as such will not generate the 
[MACADDRESS].cfg files.

Again though, this isn't too tricky to do.  A few hours' worth of work.  The 
tricky part would be making sure you got the right phone to the right desk if 
the extension #s are physically important.  :-)

If you need some help with the script I am available for consulting.  Contact 
me offlist if this is something you'd like to discuss.

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Switchboard solutions, interactions with handset

[Arnar Birgisson]

Hi there,

I'm looking into developing an in-house switchboard application. Does
anyone here know of a way to control a hard-phone from such an
application.

For example, the attendant forwards a call with another one in queue.
Once the first call has been forwarded (by keyboard shortcuts or
dragging-n-dropping) - she presses a button (on the computer) to
answer the waiting call.

Now, if the switchboard application embeds a soft-phone, I can figure
out how to do this. But suppose the attendant is using a hard-phone
(since it's more reliable) with a headset - can she do the above
things without having to press any of the phones buttons?

Wouldn't this require the application to somehow control if the phone
is off-hook or on-hook? Is there some other way I'm not seeing and/or
has someone here implemented similar stuff?

Could I possibly keep an open channel in Asterisk to the attendants
phone, and bridge that with whatever channel requested by the
switchboard application? I have found some mention of this, bridging
channels, in the mailing list archives, but not in the AMI
documentation. Is this maybe something that's still only on the svn
trunk?

thanks,
Arnar
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Realtime rtignoreexpire bugged ??

[Matt Schulte]

All, this doesn't appear normal to me, it appears as if ast is ignoring
the itignoreexpire variable.

sip.conf snippet:
rtignoreexpire=yes


asterisk -r
 CLI>sip show settings

--snip--
  Ignore Reg. Expire: No
--snip--

Does this look like a problem? :-)

Thanks, Matt



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

[Scott Gifford]

Colin Anderson <[EMAIL PROTECTED]> writes:

>>Why not capture the faxes (in or out) in tiff format, instead of audio
>>format?  Setup your asterisk box to relay faxes!
>
> I think in this case the impact on the client would be much greater if you
> can show them a recreation of the image from the raw data; you could always
> claim that a TIFF file was altered (which it can be, trivially) but it's
> pretty much impossible to change the raw audio to your ends unless you are
> in a Tom Clancy novel. 

Why is this hard to fake at all?  You send a different fax to your
system, and replace the Asterisk audio file with the one from the
altered fax.  Additionally, the client has no realistic way of
verifying the correctness of your audio-to-fax translation tool; it
could just as easily output a TIFF file completely different from the
one that was actually faxed.

Scott.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Tool for Polycom configurations

[Bruce Reeves]

I am getting read to roll out close to 100 polycom phones and wondered if any one knows of a program to take a list of MAC addresses, extensions, and names and generate the configuration files?-- Bruce
Nortex Networks
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

[Andrew Kohlsmith]

On Thursday 04 May 2006 14:08, Steve Underwood wrote:
> I would have no problem decoding a FAX, doctoring the images, then
> creating modified audio from them. During decoding, the FAX modems
> produce a channel estimate, so reproducing the characteristics of the
> original audio path wouldn't be hard. I think it would be pretty easy to
> create fresh audio that no expert could dispute as possibly being the
> original.

Yes, but nobody disputes your godlike status with the software DSP.  :-)

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] TE410P & T400P together in a server

[rapples]

Can I mix these in a single system... having problems getting
the tor2 driver or the wct4xxp drivers to load, although they
seem fine if alone in the system.
 
span=1,0,0,esf,b8zsbchan=1-23dchan=24
span=2,0,0,esf,b8zsbchan=25-47dchan=48
span=3,0,0,esf,b8zsbchan=49-71dchan=72
span=4,0,0,esf,b8zsbchan=73-95dchan=96
span=5,0,0,esf,b8zsbchan=97-119dchan=120
span=6,0,0,esf,b8zsbchan=121-143dchan=144
span=7,0,0,esf,b8zsbchan=145-167dchan=168
span=8,0,0,esf,b8zsbchan=169-191dchan=192
loadzone=usdefaultzone=us

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

[Colin Anderson]

>I would have no problem decoding a FAX, doctoring the images, then 
>creating modified audio from them. During decoding, the FAX modems 
>produce a channel estimate, so reproducing the characteristics of the 
>original audio path wouldn't be hard. I think it would be pretty easy to 
>create fresh audio that no expert could dispute as possibly being the 
>original.

Ya, but...You Da Man. I mean doctoring fax audio so that mere mortals can
comprehend how to do it. I swear there's an i960 in your head so you can
listen to an audio stream and compose the TIFF in your mind. :-)>
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP Phones behind dynamic IPs

[Chris Bagnall]

> 'recognize'? The phone cannot know that the external IP has 
> been changed, unless it is using a STUN server and 
> periodically re-doing the STUN queries (which I doubt any phones do).

Thanks for clearing up my misunderstanding as to the point of STUN. :-) I
thought the phone would query the STUN server at regular intervals to see if
the IP had changed.

Okay, so assuming I've got to drop the re-registration to a much shorter
time than the default of every hour, what are the implications of doing so
(in terms of network traffic, load on the asterisk box, etc.)? What's the
lowest one can reasonably take it? 10 minutes? 1 minute?

Thanks again.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] QSIG support in Asterisk

[Asterisk User]

I am trying to use QSIG to interoperate with legacy PBXs. 
 
I am looking to see whether any one knows whether Call Hold, Call Transfer, MWI 
works with QSIG support in Asterisk.
 
Thanks in advance.
 
--Pillai 
On 5/4/06, Olivier Krief <[EMAIL PROTECTED]> wrote:

2006/5/3, Marco Mouta <[EMAIL PROTECTED]>:



http://www.voip-info.org/wiki-Asterisk+config+zapata.conf 
I've made some tests using this in Portugal and seems to work:--- switchtype=qsig  ; you may try this in your zapata.conf
--I suppose you are using PRI access.But i've also red that qsig is not fully compliant in Asterisk. I've found some PRI states not recognized in asterisk sending CAUSECODE=98 to my legaccy PBX when it seems to me that Asterisk should handle it... 
Any ways it seems to work in a standardt architecture:PSTN--E1---LegacyPBX---QSIG---AsteriskI hope it helps,Marco Mouta 
For curiosity, what sort of benefit were you after using QSIG ? Most vendors tout SIP as interoperability protocol.Did you use QSIG as a way, for instance, to provide Message Waiting Indicator to digital phones ?
Cheers ___--Bandwidth and Colocation provided by 
Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:  
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] OT: D-link DI-102

[Colin Anderson]

Anyone use this thing? 

http://www.dlink.com/products/?pid=426

The fab sheet is totally useless for tech info. How does it work? By ToS?
Port number? Is it programmable? Can I prioritize an arbitrary port or ToS
bit? 

tia
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Soonr

[Dean Collins]

http://www.soonr.com/web/front/features.jsp

 

Just saw this on the Always On Top 100 webcast (if you aren’t
familiar – click the url below)

http://deancollinsblog.blogspot.com/2006/05/always-on-awards-top-100-of-2006.html


 

 

Soonr looks like it rocks, haven’t tried it yet.

 

 

Cheers,

 

Dean

 






___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

[Steve Underwood]

Colin Anderson wrote:


Why not capture the faxes (in or out) in tiff format, instead of audio
format?  Setup your asterisk box to relay faxes!
   



I think in this case the impact on the client would be much greater if you
can show them a recreation of the image from the raw data; you could always
claim that a TIFF file was altered (which it can be, trivially) but it's
pretty much impossible to change the raw audio to your ends unless you are
in a Tom Clancy novel. I'm watching this thread closely because where I work
there's a lot of "he-said, she-said" over faxes too. If anyone can work out
an example with SpanDSP, please share with the class! 
 

I would have no problem decoding a FAX, doctoring the images, then 
creating modified audio from them. During decoding, the FAX modems 
produce a channel estimate, so reproducing the characteristics of the 
original audio path wouldn't be hard. I think it would be pretty easy to 
create fresh audio that no expert could dispute as possibly being the 
original.


Steve

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Vahan Yerkanian]

Andrew Kohlsmith wrote:

On Thursday 04 May 2006 11:31, Louis-David Mitterrand wrote:

I've got this low-ping 100%-up dsl connection between two asterisk
1.2.7.1 servers. And oftentimes one of them would declare its opposite
UNREACHABLE.


Same, here, two asterisk 1.2.7.1 boxes connected to the same switch... 
Over a week I see at least one case of one of the boxes becoming 
unavailable for the other... simple iax2 reload fixes the problem.


Been like this for ages.

just my 2 cents,
Vahan
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

[Colin Anderson]

>Why not capture the faxes (in or out) in tiff format, instead of audio
>format?  Setup your asterisk box to relay faxes!

I think in this case the impact on the client would be much greater if you
can show them a recreation of the image from the raw data; you could always
claim that a TIFF file was altered (which it can be, trivially) but it's
pretty much impossible to change the raw audio to your ends unless you are
in a Tom Clancy novel. I'm watching this thread closely because where I work
there's a lot of "he-said, she-said" over faxes too. If anyone can work out
an example with SpanDSP, please share with the class! 

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] ISAC support?

[Michael Graves]

No. iSAC is a codec from GIPS. Likely the coded used by Skype.

Michael

On Thu, 4 May 2006 21:35:07 +1000, James Harper wrote:

>
>I assume you mean this:
>http://en.wikipedia.org/wiki/ISAC
>
>but maybe you are referring to one of the controller chips on BRI
>adapters?
>
>James
>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Trond G. Andersen
>> Sent: Thursday, 4 May 2006 20:19
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [Asterisk-Users] ISAC support?
>> 
>> Hi All.
>> 
>> Has there been done any work to support ISAC ?
>> 
>> 
>> Thanks,
>> trond
>> ___
>> --Bandwidth and Colocation provided by Easynews.com --
>> 
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>___
>--Bandwidth and Colocation provided by Easynews.com --
>
>Asterisk-Users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245



___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] web meetme instructions

[Dan Austin]

It has been approved.  We started out trying to use 
CVS on SourceForge, but
it appears that there have been major issues with CVS, so 
we just switched to
SVN.
We need to 
checkin a baseline, and start integrating patches.
 
Dan

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Ben 
  QSent: Thursday, May 04, 2006 5:33 AMTo: Asterisk Users 
  Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] web meetme instructions
  Hi,what 
  about the web meetme sourceforge project?Has it been 
  approved?benq
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] remapping sof-keys on Polcyom 301

[Bartosz Jozwiak]

Hi,

Did anybody succeed remapping soft-keys on polycom 301 ?
I am having some problems with it.
I was trying to remap Transfer button as the first option while being in a 
call.

It works but The name of the soft key is still HOLD and while I am not
in a  call I see button "NewCall" that suddenly stopped working... nothing 
happens when you press it.


This is out of my cfg:



Bartosz 


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Andrew Kohlsmith]

On Thursday 04 May 2006 11:31, Louis-David Mitterrand wrote:
> I've got this low-ping 100%-up dsl connection between two asterisk
> 1.2.7.1 servers. And oftentimes one of them would declare its opposite
> UNREACHABLE.

I see this happen on occasion as well -- same type of setup here, static IPs, 
no DNS, route seems just fine.  I even have qualify smoothing turned on, 
because I thought that the odd UDP packet would just get lost and cause this, 
but that doesn't seem to help at all.

This occurs with every version of Asterisk I've used (svn trunk), including 
the latest checkout in late April.

-A.
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

[Technical Support]

Why not capture the faxes (in or out) in tiff format, instead of audio
format?  Setup your asterisk box to relay faxes!

MD

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Thursday, May 04, 2006 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

I have a client that 'NEEDS' (his words not mine) to make sure that all
faxes, emails, calls, and mail are archived. Phone and email are simple,
Mail depends upon the integrity of the mail room, Faxes however can be sent
from anyone. They would like this as they recently had an issue with a fax
sent. Client claims that the fax received was NOT the Fax sent. In the
client file, the correct fax is there, but client attests to a different
copy. (Client is always right)

I know that I can use HylaFax and a custom context to grab the 'dialed'
number receive the fax via HylaFax and 're-send' it out again. But I have
not had the success rate I would like with HylaFax/IAXmodem. 

However, recording a file in Asterisk is a simple and manageable concept. I
do not expect problems with the recording load as there will be less than 10
recordings at once. I will NOT be mixing them and will be off-loading the
tasks off to another machine after hours for processing, realtime archiving
is not needed.

They are also 'emotionally attached' to the rather large fax machines
(Mopier) that they have. (4)

They are a law firm that deals with rather large purchases, Planes, Boats(
SHIPS!!), small island countries, etc. etc,

They only have 3 attorneys and 3 para-legals, so they are difficult to
change in their ways, I find larger orgs are less resilient to change.

That's why..



> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users- 
> [EMAIL PROTECTED] On Behalf Of Dovid Bender
> Sent: Thursday, May 04, 2006 9:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Can I recreate a Fax from a recorded
file?
> 
> 
> > This is a very KGB / NSA / InterPOL / CIA type question, but if I 
> > have a recorded file (G.711, no compression) can I feed it into 
> > standard in of an application and have it recreate the fax that was 
> > send?
> 
> What is the specific reason as to why you want to record it to a file 
> and send it out. When you need to send it to two people why not just 
> send two faxes ?
> What am I missing ?
> Dovid
> 
> __
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around 
> http://mail.yahoo.com ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Auto Logout from queue

[Kevin Savoy]

I have tried using the autologoff in the agents.conf and it sort of works. I
set it to 5 seconds to test it and it has taken anywhere from 35 to 60
seconds to actually do something at which point it does indeed log out the
agent.

I don't want to be pestering agents with test calls to see if they are
indeed there so the below scripting isn't really practical in our
environment. 

Can anyone tell me why the agents.conf file setting doesn't work as
described? If it is set to 5 it should log them off after 5 seconds or so
not 30 - 60 seconds. I don't really want the call sitting at a logged out
agents phone for anymore then 5 seconds when there are other agents out
there waiting to take that call. Any ideas? 

Thanks

_
 
Kevin Savoy
Business Unit Telecom Analyst
2218 4th Ave W
Williston, ND 58801
Ph: 701-774-4023
Fax: 701-774-2901
http://www.novo1.com
Novo 1 is a service mark of Novo 1, Inc

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alberto
Sagredo
Sent: Tuesday, April 25, 2006 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Auto Logout from queue

Via dialplan maybe?

exten => xxx,1,Dial(SIP/101_Queue,20,tr)
exten =>xxx,2,RemoveQueueMember(Comercial_Queue,SIP/101_Queue,1)



Kerry Garrison escribió:
> Yes, that is the functionality I am looking for, just not sure how exactly
> to pull that off.
>
>
>   _  
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Alexander
> Lopez
> Sent: Tuesday, April 25, 2006 12:08 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Auto Logout from queue
>
>
> Use the local channel to call the agent first, and if there is no answer,
> log them out.
>  
>  
>
>   _  
>
> From: [EMAIL PROTECTED] on behalf of Kerry Garrison
> Sent: Tue 4/25/2006 2:38 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Auto Logout from queue
>
>
> i have a client that wants a function that will automatically logout an
> agent from a queue if they do not answer a call. This would prevent future
> calls from being sent to that phone if the agent forgot to logout. Any
> ideas?
>  
> Kerry Garrison
> Director of Technical Services
> Tech Data Pros - Orange County's Mobile IT Service Provider
> (949) 502-7819 x200 -  
> [EMAIL PROTECTED]
>   http://www.techdatapros.com 
>  
>
>   
> 
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

[Louis-David Mitterrand]

I've got this low-ping 100%-up dsl connection between two asterisk 
1.2.7.1 servers. And oftentimes one of them would declare its opposite 
UNREACHABLE.

Why can this happen? The host stanzas in iax.conf have raw IP's, so no 
DNS monkey business here.. An inquiring mind wants to know.


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Voipjet Problem?

[Matt]

Just wanted to add my 2 cents.  We were with voipjet, and do still use
them for occassional backup.However, their lack of personal
service and inability to get ahold of someone drove us away.After
several total blackouts (like what happened yesterday), and no
responce we finally put out an SOS on the asterisk mailing list.  Of
course there were several responces from companies trying to solicit
us. but the one that caught our attention was calleveryone.com  
So far we have been rock-solid-happy with them.   We've had a few

small bumps along the road.   For instance, once there was a router
along our path to them that was dropping packets, but this was quickly
resolved.   Additionally, they've worked with us on the phone to
resolve audio problems, and diagnose carrier issues.   If I have a
problem, I rest assured that I can call someone, or page someone if
the situation is severe enough, and get ahold of a human at any hour
of the evening.   Not so with VoipJet.   I don't want to bad mouth
VoipJet, their service is decent... but definately not acceptable for
a carrier grade level.   I'm not affiiliated with calleveryone in any
way other then a very happy and satisfied customer, and would highly
recommend them to you.   If you are a wholesole buyer of minutes, talk
to them, don't just take their prices on the main page... those are
for residential and regular customers.  Their prices are very
comparable to voipjet, and the service is miles ahead.

On 5/3/06, Matt <[EMAIL PROTECTED]> wrote:

Yup... I think they died... this is why I stopped using them except as
my backup.   It seems 64.34.45.100  is working ok as of right now.
It wouldn't be so bad if they had a number you could call for support!
 HERE THAT JOHN?   You need a phone number if you want to "play with
the big dogs".

On 5/3/06, Mark Hulber <[EMAIL PROTECTED]> wrote:
> I started to have a problem today that all my calls through voipjet
> result in just timing out after my assigned timeout period.  I tried
> multiple of their servers with the same problem.  Anyone else having a
> problem?  I am running:
>
> Asterisk SVN-branch-1.2-r24381M built by root @ asterisk.hulber.com on a
> i686 running Linux on 2006-05-03 14:14:07 UTC
>
> I can connect with other IAX providers.
>
> MARK.
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] number that starts with star on PAP2

[Philippe Lindheimer]

> In the PAP2's setup there are all of these "Vertical Service Activation > Codes" that start with star and "Outbound Call Codec Selection Codes", > also the setup menu is accessed by pressing star four times, could they > be intefering with dialing numbers that start with a star? And is there > any way to get *8 and *XXX to dial?  Yes - that's your problem. You need to porgram the dialpan in the PAP2 appropriately, disable functions you don't want, etc.     p
		New Yahoo! Messenger with Voice. Call regular phones from your PC and save big.___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

AW: [Asterisk-Users] DTMF detection when outgoing call to mobilephones

[Marc Scheuffler]

Yapp, timeout is set to 1500ms.

What kind of dtmf mode? As far as i know there are just 2.
Relaxdtmf yes or no
Or am I wrong?
 

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Mark Ackroyd
Gesendet: Donnerstag, 4. Mai 2006 16:52
An: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Betreff: RE: [Asterisk-Users] DTMF detection when outgoing call to mobilephones

>From my recent problem on this sort of thing, I'd suggest you set the
timeout to around 1500ms in the feature.conf file. This is of course if your 
using the DTMF digit's to activate any of the features.

also make the devices both sides of the call are using the same DTMF mode.

Mark


> 
> Hi all,
> 
> I am trying to detect DTMF keys from a mobile when asterisk make an 
> outgoing call to the mobile.
> 
> The DTMF detection on incoming calls (also FROM mobiles) is working 
> very well.
> The only problem is if asterisk called the phone... Nothing is detected.
> 
> I am using a digium te205p with PMX/PSTN connection.
> 
> Everything that I can find in forums are problems with dtmf detection 
> on SIP.
> 

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] disa and caller id

[Lacy Moore - Aspendora]

Before I go nuts trying to figure this out, is anyone using DISA in this manner?
 
exten => s,1,DISA(X|context|callerid)
 
Everything works except the caller ID part.  What I had wanted to do is to setup up a file of authorization codes where each code was associated with a context and caller id.
 
The format for the file, I thought would be:
 
|context|callerid
 
and I had it set up in extensions.conf for:
 
exten => s,1,DISA(/etc/asterisk/disa.conf)
 
However, neither method seems to pass the caller ID.  And, yes, I am able to set my own caller ID.  I can set it before the dial command and it works.  Otherwise, no caller ID is sent.
 
I just want to confirm that someone is using this and it is working for them.  If that's the case, then I know it is somewhere in my setup.  If no one else has been able to get it to work, then it may not work correctly to begin with.  I'm using the latest stable version 
1.2.7.1.-- Lacy MooreAspendora, Inc. 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

AW: [Asterisk-Users] DTMF detection when outgoing call to mobilephones

[Marc Scheuffler]

Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different 
mobile network providers. Nothing.

I played with the rx/txgain values from hearing nothing to too loud... 
I have no more ideas.

Marc 

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Steve Underwood
Gesendet: Donnerstag, 4. Mai 2006 17:00
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [Asterisk-Users] DTMF detection when outgoing call to mobilephones

Marc Scheuffler wrote:

>Hi all,
>
>I am trying to detect DTMF keys from a mobile when asterisk make an 
>outgoing call to the mobile.
>
>The DTMF detection on incoming calls (also FROM mobiles) is working 
>very well.
>The only problem is if asterisk called the phone... Nothing is detected.
>
>I am using a digium te205p with PMX/PSTN connection.
>
>Everything that I can find in forums are problems with dtmf detection 
>on SIP.
>
>Any suggestions?
>  
>
Can you hear DTMF tones from the cellphone when you call it? It is possible 
they are not produced. I'm not sure if that is a handset or network issue, but 
I had this happen a few years ago when implementing an IVR that called 
subscribers for notification, and expected input from them. Most of the time it 
worked, but some phones in some locations sent nothing. If the phone made a 
call from the same location there was no problem.

Steve

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] DTMF detection when outgoing call to mobile phones

[Steve Underwood]

Marc Scheuffler wrote:


Hi all,

I am trying to detect DTMF keys from a mobile when asterisk make an
outgoing call to the mobile.

The DTMF detection on incoming calls (also FROM mobiles) is working very
well.
The only problem is if asterisk called the phone... Nothing is detected.

I am using a digium te205p with PMX/PSTN connection.

Everything that I can find in forums are problems with dtmf detection on
SIP.

Any suggestions?
 

Can you hear DTMF tones from the cellphone when you call it? It is 
possible they are not produced. I'm not sure if that is a handset or 
network issue, but I had this happen a few years ago when implementing 
an IVR that called subscribers for notification, and expected input from 
them. Most of the time it worked, but some phones in some locations sent 
nothing. If the phone made a call from the same location there was no 
problem.


Steve

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] DTMF detection when outgoing call to mobile phones

[Mark Ackroyd]

>From my recent problem on this sort of thing, I'd suggest you set the
timeout to around 1500ms in the feature.conf file. This is of course if your
using the DTMF digit's to activate any of the features.

also make the devices both sides of the call are using the same DTMF mode.

Mark


> 
> Hi all,
> 
> I am trying to detect DTMF keys from a mobile when asterisk make an
> outgoing call to the mobile.
> 
> The DTMF detection on incoming calls (also FROM mobiles) is working very
> well.
> The only problem is if asterisk called the phone... Nothing is detected.
> 
> I am using a digium te205p with PMX/PSTN connection.
> 
> Everything that I can find in forums are problems with dtmf detection on
> SIP.
> 

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] DTMF detection when outgoing call to mobile phones

[Marc Scheuffler]

Hi all,

I am trying to detect DTMF keys from a mobile when asterisk make an
outgoing call to the mobile.

The DTMF detection on incoming calls (also FROM mobiles) is working very
well.
The only problem is if asterisk called the phone... Nothing is detected.

I am using a digium te205p with PMX/PSTN connection.

Everything that I can find in forums are problems with dtmf detection on
SIP.

Any suggestions?

Cheers 
Marc
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

[Alexander Lopez]

I have a client that 'NEEDS' (his words not mine) to make sure that all
faxes, emails, calls, and mail are archived. Phone and email are simple,
Mail depends upon the integrity of the mail room, Faxes however can be
sent from anyone. They would like this as they recently had an issue
with a fax sent. Client claims that the fax received was NOT the Fax
sent. In the client file, the correct fax is there, but client attests
to a different copy. (Client is always right)

I know that I can use HylaFax and a custom context to grab the 'dialed'
number receive the fax via HylaFax and 're-send' it out again. But I
have not had the success rate I would like with HylaFax/IAXmodem. 

However, recording a file in Asterisk is a simple and manageable
concept. I do not expect problems with the recording load as there will
be less than 10 recordings at once. I will NOT be mixing them and will
be off-loading the tasks off to another machine after hours for
processing, realtime archiving is not needed.

They are also 'emotionally attached' to the rather large fax machines
(Mopier) that they have. (4)

They are a law firm that deals with rather large purchases, Planes,
Boats( SHIPS!!), small island countries, etc. etc,

They only have 3 attorneys and 3 para-legals, so they are difficult to
change in their ways, I find larger orgs are less resilient to change.

That's why..



> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Dovid Bender
> Sent: Thursday, May 04, 2006 9:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Can I recreate a Fax from a recorded
file?
> 
> 
> > This is a very KGB / NSA / InterPOL / CIA type
> > question, but if I have a
> > recorded file (G.711, no compression) can I feed it
> > into standard in of
> > an application and have it recreate the fax that was
> > send?
> 
> What is the specific reason as to why you want to
> record it to a file and send it out. When you need to
> send it to two people why not just send two faxes ?
> What am I missing ?
> Dovid
> 
> __
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] number that starts with star on PAP2

[Time Bandit]

In the PAP2's setup there are all of these "Vertical Service Activation
Codes" that start with star and "Outbound Call Codec Selection Codes",
also the setup menu is accessed by pressing star four times, could they
be intefering with dialing numbers that start with a star?  And is there
any way to get *8 and *XXX to dial?


Why I did to mine is modify all the internal "Vertical Service
Activation Codes" to be "**x" instead of "*x". There is probably a
better way, but this worked for me.

hth
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

SV: [Asterisk-Users] Polycom 501 - Disable DND feature?

[jan.sarin]

Well, yes and no. I tested that before and it causes a silent ring 
instead of a call rejection. I actually want to disable the entire feature. So 
the phone always rings unless you're actually on the phone.
 
Thanks for the reply though!
 
Regards,Jan


Från: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] För Jerry 
JonesSkickat: den 4 maj 2006 15:00Till: Asterisk Users 
Mailing List - Non-Commercial DiscussionÄmne: Re: [Asterisk-Users] 
Polycom 501 - Disable DND feature?

Attribute Values Default 
Interpretation 
call.rejectBusyOnDnd 0, 1 1 If 
set to 1, reject all incoming calls with 
the reason “busy” if 
do-not-disturb is 
enabled. 
Have not used, but looks like it 
may ignore the key if this is 0

Let us know...



On May 4, 2006, at 2:22 AM, <[EMAIL PROTECTED]> <[EMAIL PROTECTED]> wrote:

  Hi,
  
  Is it possible to disable the DND feature on a Polycom 501?
  
  Regards,
  Jan
  ___
  --Bandwidth and Colocation provided by Easynews.com --
  
  Asterisk-Users mailing list
  To UNSUBSCRIBE or update options visit:
     http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] QSIG support in Asterisk

[Marco Mouta]

QSIG was just the protocol communication between Legaccy PBX and Asterisk.My users connect to Asterisk through SIPOn 5/4/06, Olivier Krief <
[EMAIL PROTECTED]> wrote:
2006/5/3, Marco Mouta <[EMAIL PROTECTED]>:

http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
I've made some tests using this in Portugal and seems to work:---
switchtype=qsig  ; you may try this in your zapata.conf--I suppose you are using PRI access.But i've also red that qsig is not fully compliant in Asterisk. I've found some PRI states not recognized in asterisk sending CAUSECODE=98 to my legaccy PBX when it seems to me that Asterisk should handle it...
Any ways it seems to work in a standardt architecture:PSTN--E1---LegacyPBX---QSIG---AsteriskI hope it helps,Marco Mouta
For curiosity, what sort of benefit were you after using QSIG ?
Most vendors tout SIP as interoperability protocol.Did you use QSIG as a way, for instance, to provide Message Waiting Indicator to digital phones ?Cheers

___--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] ISAC support?

[Trond G. Andersen]

 
Yes, sorry I was wondering if anyone is working on ISAC voice codec


I have seen a patch 
http://lists.digium.com/pipermail/asterisk-commits/2006-February/001461.
html
But not seen anything anywhere else...


trond




I assume you mean this:
http://en.wikipedia.org/wiki/ISAC

but maybe you are referring to one of the controller chips on BRI
adapters?

James

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users- 
> [EMAIL PROTECTED] On Behalf Of Trond G. Andersen
> Sent: Thursday, 4 May 2006 20:19
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] ISAC support?
> 
> Hi All.
> 
> Has there been done any work to support ISAC ?
> 
> 
> Thanks,
> trond
> ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

[Dovid Bender]

> This is a very KGB / NSA / InterPOL / CIA type
> question, but if I have a
> recorded file (G.711, no compression) can I feed it
> into standard in of
> an application and have it recreate the fax that was
> send?

What is the specific reason as to why you want to
record it to a file and send it out. When you need to
send it to two people why not just send two faxes ?
What am I missing ?
Dovid

__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Polycom 501 - Disable DND feature?

[Jerry Jones]

Attribute Values Default Interpretation call.rejectBusyOnDnd 0, 1 1 If set to 1, reject all incoming calls with the reason “busy” if do-not-disturb is enabled. Have not used, but looks like it may ignore the key if this is 0Let us know...On May 4, 2006, at 2:22 AM, <[EMAIL PROTECTED]> <[EMAIL PROTECTED]> wrote:Hi,Is it possible to disable the DND feature on a Polycom 501?Regards,Jan___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

[Olivier Krief]

2006/5/4, Craig Guy <[EMAIL PROTECTED]>:
If you have both sides of the call it is possible.  It may not be practicalthough.  If one side was using spandsp then it is both possible andpractical.CraigCould you elaborate ?
And if a fax is recorded with Asterisk voicemail application (in case an error in fax detection occurred), would it still be possible and pratical ?Regards
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SpeedDial on GXP-2000

[Waldo Rubinstein]

How can you store "pauses" in speed dials for the GXP-2000? I used  
something like 8005551212,,,1,7890 to dial the toll free number, wait  
6 seconds (I'm used to the commas being a 2 second delay), pressing  
1, waiting 2 more seconds and then entering 7890. However, when I  
press the speeddial button, the phone freezes.


Thanks,
Waldo

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] QSIG support in Asterisk

[Olivier Krief]

2006/5/3, Marco Mouta <[EMAIL PROTECTED]>:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
I've made some tests using this in Portugal and seems to work:---
switchtype=qsig  ; you may try this in your zapata.conf--I suppose you are using PRI access.But i've also red that qsig is not fully compliant in Asterisk. I've found some PRI states not recognized in asterisk sending CAUSECODE=98 to my legaccy PBX when it seems to me that Asterisk should handle it...
Any ways it seems to work in a standardt architecture:PSTN--E1---LegacyPBX---QSIG---AsteriskI hope it helps,Marco MoutaFor curiosity, what sort of benefit were you after using QSIG ?
Most vendors tout SIP as interoperability protocol.Did you use QSIG as a way, for instance, to provide Message Waiting Indicator to digital phones ?Cheers
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] QSIG support in Asterisk

[Olivier Krief]

2006/5/3, Asterisk User <[EMAIL PROTECTED]>:
 Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking?
Do you mean something like ECMA 336 ?http://www.ecma-international.org/publications/files/ECMA-ST/Ecma-336.pdfRegards

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users