date:20060504

 I noticed one thing: I got courtesytone = beep in my features.conf
 If I took it off, I got no sound.

That's one sorted out :-)

 Do you have this on it? Do you have a global DYNAMIC_FEATURES =
 monitor in extensions.conf ?

Yes. 


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Re: [Asterisk-Users] echo in Snom 360 phones

2006-05-04 Thread Steve Davies


On 5/3/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote:


One of my users reports frequently hearing echo on her Snom 360 phone,
even while talking to other Snom phones (via Asterisk) on the same LAN
(i.e., all-digital low-latency connection). I can never reproduce it
though, and swapping the phone didn't help.

Has anyone else seen mystery echo on Snom phones? Any suggestions for
debugging?

On my own Snom 360, I sometimes hear an echo for the first second or
two, and then it goes away. I guess an echo cancellation circuit kicks
in, inside the Snom.


We use snoms is a number of locations, and occasionally hear reports
of echo from users making internal SIP to SIP calls. I believe that
some of the older (much older) versions of the snom firmware were
quite poor at echo prevention, so the remote handset may be the
issue. Remember that echo is almost always caused by the far-end and
not the person hearing it.

We use the 4.5 firmware on snom320/snom360 phones, and 3.60x firmware
on snom190 phones with good results. There is also the possibility
that the microphone volume on the snom360 is set too high. 4 or 5 is
about normal.

Cheers,
Steve
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[Asterisk-Users] SetGroup and CheckGroup. Need some help on the dialplan

2006-05-04 Thread Arne Morten Johansen


From this list I found that I could use SetGroup and CheckGroup to do
what I wanted. But I'm not quite sure how I do it.

The case is that I have 3 user groups, and one main group. The main
group is for all the incoming calls from external phones. The main group
should be allowed to have 3 calls at the time. 

The 3 user groups are internal groups that I want to limit by ony having
one incoming call at the time. So if userA-Phone1 is on the phone.
UserB-phone 2 shouldn't get any calls.

Illustration: (hope it don't get messed up)
Incomming Call -- My Company -- Group1
 -- UserA-phone1
 -- UserA-phone2
  -- Group2
--- UserB-phone3
--- UserB-phone4   
 -- Group3
--- UserC-phone5
--- UserC-phone6

So what I want per group level is that only one user (phone) is active
at the time. And if all of the groups are busy, I want to send the
caller to voicemail.  

Everyone can call out at the same time, but it must update the group
count the phone belongs to. 

The main problem for me is the dialplan that decides witch Groups are
available. And also how to decide witch group to update on outgoing
calls. (all the users use the same extensions and peer for outgoing)

Hope that made sense.

Thanks

Regards,
Arne Morten


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RE: [Asterisk-Users] hyperthreading and zaptel

 Finally, I decided to turn hyperthreading back on, and everything is back
 to normal.
 
 Unless there is somewhere in CentOS 4.3 that has the processor count
 hardcoded from the install, I am baffled by this.

Was it on when * and zaptel was compiled?. Maybe the compiler produced HT
optimized binaries.  Just a thought?

Mark

 

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

2006-05-04 Thread Hadley Rich

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

-- 
You buttered your bread, now lie in it.
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[Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.

2006-05-04 Thread Asterisk


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] 
mailto:[EMAIL PROTECTED]

And tell them you have this issue..


James


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Re: [Asterisk-Users] TDM400P and monoBRI auto-dial call difference: caller phone does not ring

2006-05-04 Thread picciuX

probably it's better to auto-dial the calling phone first, and then let the established channel go out to the recipient!
So when the calling phone answers, the call will go out to the recipient.
Hope this helps...
2006/5/4, Giorgio Incantalupo [EMAIL PROTECTED]:
Hi,I'm using Asterisk 1.2.1 on a Debian Sarge with a TDM400P and a monoBRIusing chan-mISDN from beronet site.
It seems to work all right except for autodial calls, monoBRI ISDNchannel behaves differently waiting for the caller to answer and thencontinue.Asterisk console says:analog: -- Attempting call on Zap/2/3391818250 for 
[EMAIL PROTECTED]:1 (Retry 1) Channel Zap/2-1 was answered. -- Executing DeadAGI(Zap/2-1, exten2.py|ticket=19) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/exten2.py
ISDN: -- Attempting call on mISDN/1/3391818250/s for [EMAIL PROTECTED]:1 (Retry 1)*Asterisk stops here for the caller to answer then go on to show the rest:* Channel mISDN/1-u8 was answered.
 -- Executing DeadAGI(mISDN/1-u8, exten2.py|ticket=21) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/exten2.pyWhy this pause? This is a problem because with ISDN the calling party
phone does not ring.Is there some parametere to set in misdn.conf??TIAGiorgio Incantalupo___--Bandwidth and Colocation provided by 
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AW: [Asterisk-Users] SIP Phones behind dynamic IPs

2006-05-04 Thread Vinzens, Joeran

I have thew same problem.

Ui tried with dyn dns in the externip field in sip.conf but I think the 
Asterisk does not allow this. Unfortunally I have every day a new ip. Maybe I 
can write a script witch takes my actual ip from externat and put it into the 
externip field. Maybe this solves the problem.

 

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Gareth Blades
Gesendet: Donnerstag, 4. Mai 2006 09:59
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [Asterisk-Users] SIP Phones behind dynamic IPs

I would also recomend that you upgrade to the latest firmware 1.0.2.13
(contact grandstream) as it does fix some registeration issues and have
extra NAT/STUN features.

On Wed, 2006-05-03 at 17:15, Chris Bagnall wrote:
 Greetings list,
 
 I'm coming across an issue with some of the GXP-2000 phones we have out in
 the wild at clients' employees' homes. In most cases they're behind consumer
 ADSL NAT routers on a dynamic IP from their ISP.
 
 In a nutshell, the phone is unable to be called unless it's restarted first,
 after which it's fine for a good few hours, then it stops working until
 restarted again.
 
 The problem doesn't seem to be anywhere near as regular with users that are
 on cable connections (these tend to have much more sticky IP addresses -
 they change only every few months rather than every time the ADSL router
 connects), and non-existent on ADSL connections with static IPs.
 
 I've tried various permutations - with STUN, without STUN, NAT keep-alives
 down as low as 10 seconds, nat=yes in sip.conf, ports forwarded to the
 phone, ports *not* forwarded to the phone, etc.
 
 I think what's happening is that the ADSL router is reconnecting after a
 break in the connection (as it should), getting a different IP, but the
 phones don't seem to be recognising they've got a different IP and updating
 the asterisk server with the good news.
 
 Has anyone else encountered similar issues? Anything else I can try (bearing
 in mind I have no control over the ADSL connections the users are subscribed
 to)?
 
 Thanks in advance.
 
 Regards,
 
 Chris

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Re: [Asterisk-Users] TDM400P and monoBRI auto-dial call difference: caller phone does not ring

2006-05-04 Thread Giorgio Incantalupo


Hi picciux,
maybe it could work, even if I don't know how to call a phone a channel 
and create a bridge between them.
I'd prefer to use Asterisk inner features like the auto-dial out call 
moving a .call file to /var/spool/asterisk/outgoing: this works for 
analog line but not for ISDN. But only in this case.when normally 
calling from a phone using an ISDN line, everything works fine. It is 
the .call mechanism which does not workand I want to understand why, 
but it seems nobody had this problem before. It is also true not many 
people use BRI ISDN.


Btw, thanx again.

Giorgio Incantalupo


picciuX wrote:
probably it's better to auto-dial the calling phone first, and then 
let the established channel go out to the recipient!
So when the calling phone answers, the call will go out to the 
recipient.

Hope this helps...

 
2006/5/4, Giorgio Incantalupo [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]:


Hi,
I'm using Asterisk 1.2.1 on a Debian Sarge with a TDM400P and a
monoBRI
using chan-mISDN from beronet site.
It seems to work all right except for autodial calls, monoBRI ISDN
channel behaves differently waiting for the caller to answer and then
continue.
Asterisk console says:

analog:

   -- Attempting call on Zap/2/3391818250 for
[EMAIL PROTECTED]:1 (Retry 1)
   Channel Zap/2-1 was answered.
   -- Executing DeadAGI(Zap/2-1, exten2.py|ticket=19) in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/exten2.py

ISDN:

   -- Attempting call on mISDN/1/3391818250/s for
[EMAIL PROTECTED]:1 (Retry 1)

*Asterisk stops here for the caller to answer then go on to show
the rest:*

   Channel mISDN/1-u8 was answered.
   -- Executing DeadAGI(mISDN/1-u8, exten2.py|ticket=21) in
new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/exten2.py

Why this pause? This is a problem because with ISDN the calling party
phone does not ring.
Is there some parametere to set in misdn.conf??

TIA

Giorgio Incantalupo
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Re: [Asterisk-Users] TDM400P and monoBRI auto-dial call difference: caller phone does not ring

2006-05-04 Thread picciuX

hi giorgio...

when i said ring the calling phone first I mean using a .call file!
I think now you are doing, in your .call files,something like this:

Channel: Zap/2/3391818250 or mISDN/1/3391818250/s
.
.
.
and the rest to send this channel to the calling phone.
This way, you have to wait the dial-out channel to be answered before connect it to the calling phone. The fact that it works with TDM400 is only due to the fact that analog lines don't support call progress, so the call appears answered as soon as the fxo channel starts dialing out. With digital lines, instead, the channel is answered only when the remote party pickups the handset, that is correct. You will find same beaviour if you dial avoip line.

But, if you make your .callto connect to the calling phone first, then the call will go out ONLY when the calling party pickups! Which is correct!
By the way, doing like you do, you could have a situation where the remote party (33918) pickups an incoming call and hears a ringing tone waiting the calling phone to answer.

Don't know if I make it clear... anyway... you're italian i think... if you want we can talk more about that privately, and in italian!

Bye

picciuX
2006/5/4, Giorgio Incantalupo [EMAIL PROTECTED]:
Hi picciux,maybe it could work, even if I don't know how to call a phone a channeland create a bridge between them.
I'd prefer to use Asterisk inner features like the auto-dial out callmoving a .call file to /var/spool/asterisk/outgoing: this works foranalog line but not for ISDN. But only in this case.when normally
calling from a phone using an ISDN line, everything works fine. It isthe .call mechanism which does not workand I want to understand why,but it seems nobody had this problem before. It is also true not many
people use BRI ISDN.Btw, thanx again.Giorgio IncantalupopicciuX wrote: probably it's better to auto-dial the calling phone first, and then let the established channel go out to the recipient!
So when the calling phone answers, the call will go out to the recipient. Hope this helps... 2006/5/4, Giorgio Incantalupo
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi, I'm using Asterisk 1.2.1 on a Debian Sarge with a TDM400P and a
monoBRI using chan-mISDN from beronet site. It seems to work all right except for autodial calls, monoBRI ISDN channel behaves differently waiting for the caller to answer and then
continue. Asterisk console says: analog:-- Attempting call on Zap/2/3391818250 for [EMAIL PROTECTED]:1 (Retry 1) Channel Zap/2-1 was answered.
-- Executing DeadAGI(Zap/2-1, exten2.py|ticket=19) in new stack-- Launched AGI Script /var/lib/asterisk/agi-bin/exten2.py ISDN:-- Attempting call on mISDN/1/3391818250/s for
[EMAIL PROTECTED]:1 (Retry 1) *Asterisk stops here for the caller to answer then go on to show the rest:* Channel mISDN/1-u8 was answered.
-- Executing DeadAGI(mISDN/1-u8, exten2.py|ticket=21) in new stack-- Launched AGI Script /var/lib/asterisk/agi-bin/exten2.py Why this pause? This is a problem because with ISDN the calling party
phone does not ring. Is there some parametere to set in misdn.conf?? TIA Giorgio Incantalupo ___
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RE: [Asterisk-Users] Dial Option wW picking up the *1 is a bit flaky

All sorted now. The features timeout needs to be quite high on mobiles.
After a few tests, it works perfectly.

thanks for your help :-)
Mark

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[Asterisk-Users] ISAC support?

2006-05-04 Thread Trond G. Andersen

Hi All.

Has there been done any work to support ISAC ?


Thanks,
trond
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[Asterisk-Users] Pattern matching DISA

2006-05-04 Thread Nicu

I would like to create an variable length extension that when used with 
DISA ends when i dial the pound sign (#) but i cant figure out how to do it

something like
exten = _*21*.#  ; but this doesn't work, after i dial # it still waits 
a few seconds


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[Asterisk-Users] Pattern matching DISA

2006-05-04 Thread Nicu

I would like to create an variable length extension that when used with 
DISA ends when i dial the pound sign (#) but i cant figure out how to do it

something like
exten = _*21*.#  ; but this doesn't work, after i dial # it still waits 
a few seconds

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

2006-05-04 Thread Warren Burstein

We have some extensions in our dialplan that start with a star.  We can 
dial them from Zap phones and SIP phones, but not from phones connected 
to a PAP2.  After the user presses star follwed by two digits (our 
extensions are dialed with star followed by three digits) he hears a 
fast-busy that comes from the PAP2, not from Asterisk.  This also 
happens with the builtin *8 (call pickup).


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?


thanks
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[Asterisk-Users] Internet exposed asterisk server.

2006-05-04 Thread Jan du Toit


Hi.

I have a soft phone (X-Lite) which registers with a asterisk server that 
can only be accessible once we have some virtual private network 
software up and running.

With the above scenario everything works fine.

In the mean time the asterisk server was exposed to the internet, thus 
the virtual private network software is no longer needed. But when I try 
and register it gives me the following:


   Registration from 'User sip:[EMAIL PROTECTED]' failed for 
'yyy.yyy.yyy.yyy' - Wrong password


Were xxx.xxx.xxx.xxx is the internal ip of the asterisk server, not the 
ip the external ip (the ip on the internet) and were yyy.yyy.yyy.yyy is 
my external ip address as seen on the internet.


I trippled check all the authentication details.
Why is it not working on the exposed server?

Did I do something wrong? Is there special confugurations when exposing 
an asterisk server?


Thanks in advance.




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[Asterisk-Users] Meetme from MySQL

2006-05-04 Thread Chris Blunt









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








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

2006-05-04 Thread Tony Mountifield

In article [EMAIL PROTECTED],
Chris Stenton [EMAIL PROTECTED] wrote:
 This is a known problem and it does not matter what zaptel timer you use. A 
 solution is available in  'svn head' by using
 
 asterisk.conf
 internal_timing = yes
 
 OR
 Enable internal timing support (-I)
 
 on the command line. I don't know if this has been backported to the stable 
 branch.

It hasn't, specifically, but the required changes are not large, and it is
easy to apply the changes by hand; I do.

Go to http://bugs.digium.com/view.php?id=5374 and download the last
asynchronous patch, 2005-10-04-3-asynchronous.patch

Then apply it by hand to channel.c, and also to app_milliwatt.c and app_sms.c
if you happen to be using those applications. I don't think the app_chanspy.c
patch is required any more.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] RE: meetme conference latency degrades...

2006-05-04 Thread Hagen Rode

I think you need to upgrade to the latest Asterisk. Your version is pretty
ancient. 

 We are using v1.0.8

- Original Message - 
From: Michael George [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, May 04, 2006 2:48 AM
Subject: [Asterisk-Users] meetme conference latency degrades...

 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

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RE: [Asterisk-Users] ISAC support?

2006-05-04 Thread James Harper

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
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RE: [Asterisk-Users] Unwanted conference with snom320 and asterisk 1.07bristuffed

2006-05-04 Thread Alexander Lopez

Title: Messaggio








Under Advanced make sure this is set:



Call join on Xfer (2 calls): to off 



















From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tommaso Calosi
Sent: Thursday, May 04, 2006 4:02
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Unwanted
conference with snom320 and asterisk 1.07bristuffed







Ihave 13 Snom
320 with asterisk 1.07 bristuffed. The problem is that sometimes on random
basis, when one customer is placed on hold and another call arrives, the
customers are put in conference with each other. This look very strange to me,
but I've disabled the confernce button on the snom phones to prevent the human
errors, but it still occurs. 

Investigating I've discovered that a similar problem was fixed with the Snom320
Release 5.2 (http://www.snom.com/snom320_release_notes.html
) 

It says: 
fixed unwanted conference bug in offhook/enter during ringback with an incoming
call 

BUT my phones are already running 5.2 firmware. 

Any idea? 

Am I the only one with this problem? 
Do you think is the usual buggy-snom firmware problem? Or it might be an
Asterisk problem? 





























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

2006-05-04 Thread Craig Guy

If you have both sides of the call it is possible.  It may not be practical 
though.  If one side was using spandsp then it is both possible and 
practical.

Craig

- Original Message - 
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Wednesday, May 03, 2006 11:02 PM
Subject: RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

Maybe if you had the un-muxed sending side but I really have no idea. 
Interesting question though.

-Original Message- 
From: Alexander Lopez [mailto:[EMAIL PROTECTED]

Sent: Wed 5/3/2006 9:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: [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?

I don’t know enough about the Fax handshaking to understand this.

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[Asterisk-Users] PCI voltage

2006-05-04 Thread Giordano Grandis









Hi all,

I have to bought a PCI with 4 PRI but on digium site
I saw that there a re two different kind (3,3V and 5v). Whats the
difference?

Which one I have to buy for do not have any problem
with this motherboard? (Gygabyte GA-8S661FXM-775). I checked on Gigabyte
website but I dont find any kind of this value K



Thanks all



Giordano








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[Asterisk-Users] Re: hyperthreading and zaptel

2006-05-04 Thread Steven

cat /proc/interrupts
   CPU0   CPU1   CPU2   CPU3
  0:422  0  0   46196367IO-APIC-edge  timer
  8:  0  0  0155IO-APIC-edge  rtc
  9:  0  0  0  0   IO-APIC-level  acpi
 14: 414685  0  0241IO-APIC-edge  ide0
169:  0  0  0  0   IO-APIC-level  uhci_hcd
177:  0  0  0  0   IO-APIC-level  uhci_hcd
185:  0  0  0  0   IO-APIC-level  uhci_hcd
193:  0  0  0   2105   IO-APIC-level  ehci_hcd
201:  61965  0  0   4612   IO-APIC-level  megaraid
209:  0  0  0   46015177   IO-APIC-level  wct4xxp
217: 399933  0  0333   IO-APIC-level  eth0
NMI:   46196423   46196379   46196377   46196376
LOC:   46196579   46196300   46196579   46196294
ERR:  0
MIS:  0

The Interrupt addresses were the same with and without hyperthreading, just the 
number of CPUs was two.

Mark suggested that the binaries might have been HT optimized.
I did a quick search of the code and didn't find anything, but I am not exactly 
sure what the keyword might be for that.
I did recompile and install zaptel with hyperhtreading off (with no success) 
and that is the build I am using with it back on now.





-- 
-- 
Steven

http://www.glimasoutheast.org



James Harper [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Turning on hyperthreading may have changed the way interrupts are
routed. Were you using the same kernel (eg SMP kernel even with
hyperthreading disabled)? The BIOS may have configured things
differently too if you disabled it there.

I'm not sure, but you may be able to keep hyperthreading on in the BIOS
and boot into a UP kernel and have the same net effect as having ht
disabled.

You mention you have looked at /proc/interrupts, are there any
differences between the interrupt numbers assigned in the ht enabled and
ht disabled cases?

When the kernel boots, it dumps some info about IRQ routing, compare
those.

Maybe post /proc/interrupts and the relevant bits of the kernel boot
logs here if you aren't sure, someone might be able to spot something
out of the ordinary.

James

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Steven
 Sent: Thursday, 4 May 2006 09:18
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] hyperthreading and zaptel

 My Dell 2800 Dual 64bit  Proc. machine came in with hyperthreading
 enabled. (they call it virtual processor??)

 I have been intending for a month to disable it.

 Tonight, I rebooted, turned it off, and let the system come up.

 zaptel loaded and asterisk loaded, but both of my t1s were red. (it is
a
 TE411P)
 /proc/interrupts looked OK, zttest gave OK numbers.
 I doublechecked all of the files in case I changed something else
 accidentally.
 I tried various combinations of unloading, loading the modules and
ztcfg,
 etc.

 Finally, I decided to turn hyperthreading back on, and everything is
back
 to normal.

 Unless there is somewhere in CentOS 4.3 that has the processor count
 hardcoded from the install, I am baffled by this.






 --
 --
 Steven

 http://www.glimasoutheast.org





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Re: [Asterisk-Users] PCI voltage

2006-05-04 Thread Rob Lith

5 volt will be for desktop class motherboards and 3.3v for server class.See http://www.digium.com/en/docs/misc/pci_slot.phpRob
On 04/05/06, Giordano Grandis [EMAIL PROTECTED] wrote:













Hi all,

I have to bought a PCI with 4 PRI but on digium site
I saw that there a re two different kind (3,3V and 5v). What's the
difference?

Which one I have to buy for do not have any problem
with this motherboard? (Gygabyte GA-8S661FXM-775). I checked on Gigabyte
website but I don't find any kind of this value K




Thanks all



Giordano









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Re: [Asterisk-Users] web meetme instructions

2006-05-04 Thread Ben Q

Hi,what about the web meetme sourceforge project?Has it been approved?benq
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[Asterisk-Users] Developping SoftPhone

2006-05-04 Thread Olivier Saulnier


Hello,

I would like to use an ocx for integrated a softphone in an existant 
program developped in Windev (from PC Soft).
I try IaxClientOCx, but nothing happen at initialising. Then, I try some 
softphone make with it, it doesn't function either...


Do you know any other OCX for try?

Best regards,
Olivier S.
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Re: [Asterisk-Users] QSIG support in Asterisk

2006-05-04 Thread 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

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Re: [Asterisk-Users] QSIG support in Asterisk

2006-05-04 Thread 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
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[Asterisk-Users] SpeedDial on GXP-2000

2006-05-04 Thread 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

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

2006-05-04 Thread 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
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Re: [Asterisk-Users] Polycom 501 - Disable DND feature?

2006-05-04 Thread 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 ___
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Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

2006-05-04 Thread Dovid Bender

snip
 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?
/snip
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

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RE: [Asterisk-Users] ISAC support?

2006-05-04 Thread 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
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Re: [Asterisk-Users] QSIG support in Asterisk

2006-05-04 Thread 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

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

2006-05-04 Thread 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
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Re: [Asterisk-Users] number that starts with star on PAP2

2006-05-04 Thread 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
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RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

2006-05-04 Thread 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?
 
 snip
  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?
 /snip
 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
 
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[Asterisk-Users] DTMF detection when outgoing call to mobile phones

2006-05-04 Thread 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
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RE: [Asterisk-Users] DTMF detection when outgoing call to mobile phones

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.
 

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

2006-05-04 Thread 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

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

2006-05-04 Thread 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

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[Asterisk-Users] disa and caller id

2006-05-04 Thread 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. 
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AW: [Asterisk-Users] DTMF detection when outgoing call to mobilephones

2006-05-04 Thread 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.
 

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

2006-05-04 Thread 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
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Re: [Asterisk-Users] Voipjet Problem?

2006-05-04 Thread 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.
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[Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

2006-05-04 Thread 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.


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RE: [Asterisk-Users] Auto Logout from queue

2006-05-04 Thread 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 -  mailto:[EMAIL PROTECTED]
 [EMAIL PROTECTED]
  http://www.techdatapros.com/ http://www.techdatapros.com 
  

   
 

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

2006-05-04 Thread 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?
 
 snip
  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?
 /snip
 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
 
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Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

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

2006-05-04 Thread 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:

keys key.scrolling.timeout=1 key.IP_300.28.function.prim=Transfer/

Bartosz 


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RE: [Asterisk-Users] web meetme instructions

2006-05-04 Thread 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
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RE: [Asterisk-Users] ISAC support?

2006-05-04 Thread 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
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--
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



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

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! 

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

2006-05-04 Thread 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
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Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

2006-05-04 Thread 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

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[Asterisk-Users] Soonr

2006-05-04 Thread Dean Collins









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



Just saw this on the Always On Top 100 webcast (if you arent
familiar  click the url below)

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






Soonr looks like it rocks, havent tried it yet.





Cheers,



Dean








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

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
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Re: [Asterisk-Users] QSIG support in Asterisk

2006-05-04 Thread 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 ?
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RE: [Asterisk-Users] SIP Phones behind dynamic IPs

2006-05-04 Thread 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


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

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. :-)
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[Asterisk-Users] TE410P T400P together in a server

2006-05-04 Thread 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

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

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.
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[Asterisk-Users] Tool for Polycom configurations

2006-05-04 Thread 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
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Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

2006-05-04 Thread 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.
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[Asterisk-Users] Realtime rtignoreexpire bugged ??

2006-05-04 Thread 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
 CLIsip show settings

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

Does this look like a problem? :-)

Thanks, Matt



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[Asterisk-Users] Switchboard solutions, interactions with handset

2006-05-04 Thread 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
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Re: [Asterisk-Users] Tool for Polycom configurations

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.
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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.
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Re: [Asterisk-Users] Tool for Polycom configurations

2006-05-04 Thread 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.
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Re: [Asterisk-Users] Tool for Polycom configurations

2006-05-04 Thread 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




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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
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RE: [Asterisk-Users] Tool for Polycom configurations

2006-05-04 Thread 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 
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RE: [Asterisk-Users] Voipjet Problem?

2006-05-04 Thread 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.
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RE: [Asterisk-Users] Can I recreate a Fax from a recorded file?

2006-05-04 Thread 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.
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Re: [Asterisk-Users] Tool for Polycom configurations

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

2006-05-04 Thread 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
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Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

2006-05-04 Thread 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.
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Re: [Asterisk-Users] Voipjet Problem?

2006-05-04 Thread 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.
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[Asterisk-Users] Volume configuration on Polycom Soundpoint 501 phone

2006-05-04 Thread 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
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Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

2006-05-04 Thread 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.

rant
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?
/rant


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

2006-05-04 Thread 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
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Re: [Asterisk-Users] Tool for Polycom configurations

2006-05-04 Thread Anthony Rodgers


Hi Bruce,

We create a CSV file of our phone setup and then use shell scripts to 
parse them and generate mac-address.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
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RE: [Asterisk-Users] ISAC support?

2006-05-04 Thread 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
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Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

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



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

2006-05-04 Thread Sean Cook


sip.cfg

volume voice.volume.persist.handset=1 
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


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[Asterisk-Users] Unable to get TDM400p working

2006-05-04 Thread 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





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

2006-05-04 Thread 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 

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

2006-05-04 Thread 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
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RE: [Asterisk-Users] Tool for Polycom configurations

2006-05-04 Thread 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?
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Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?