date:20050817

Thank you W, asking differently:

(Suppose) I have a very reliable hardware,motherboard,power
supply,bios,kernel,configuration, I have reliable and fast everything.

Now what is the maximum number of FXO/FXS modules? What does it depend in
asterisk/tdm cards now?

BDM.

[EMAIL PROTECTED] wrote:
>Just Google the archive on 'IRQ issues'.  
>You can pretty much bet that 6 TDM cards on 6 PCI slots would suck hugely.
>Unless echo is your goal, you are not going to be pleased.
>
>If you have to use 24 existing POTS lines, look into a channel bank and
>interface it to a T1 card.
>If you are planning new, just get a PRI T1 and be done with it.

>Cheers,
>W
<< Is it possible to use 24 FXS/FXO lines(on 6 PCI slots) at the same time on

RE: [Asterisk-Users] Any one using the new Digium echocancellationcards

2005-08-17 Thread Shane Burrell

I have experienced similar problems for the past few weeks with no resolve.
Last message I received I was the only one with the problems but now I see
it starting to be more prevalent.  In my case the card is unusable and has
left a bad taste in everyone's mouth around here.

SB

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard A.
Smith
Sent: Wednesday, August 17, 2005 8:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Any one using the new Digium
echocancellationcards

David,

Yes we got them and they caused huge problems.  The echo training would
cause the line to mute and you would hear something like a dtmf tone briefly
and then you would be connected and talking again.  This might happen once
or 50 times during a call.  I spoke to Digium and they say there may be a
firmware upgrade coming down the line.  You are right it is a daughter board
placed in the space provided on the existing cards.  We disabled the
daughter board and are basically back to where we started from.

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David
Zanetti
Sent: Wednesday, August 17, 2005 4:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Any one using the new Digium
echocancellation cards

On Wed, 2005-08-17 at 08:56 -0500, Alan Bunch wrote:
> THe wiki doesn't seem to have any user reports.
>
> If your using them, how are the working, better, worse about the same.
>
> Also what hardware seems to be stable with them installed.

I'd also be interested if the module is available as an upgrade to
existing quad boards. It looks rather like the echo canceller is a
daughter board, connected roughly where there's a connector on existing
quad boards...

Have a rather nastry PRI echo problem which so far no fiddling with
settings (tx, rx, taps) has helped with.

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

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Re: [Asterisk-Users] Voicemail file permissions

2005-08-17 Thread hugolivude

Sorry, not following you at all.  Perhaps it's a lack of Linux knowledge.  

I understood that Asterisk puts voicemails in
var/spool/asterisk/voicemail/mailbox/###/INBOX.  Are you suggesting
that I could configure that to be a network drive on each user's
computer?  If so wouldn't that introduce all kinds of complications
like what to do when their computer is off?

Thanks for taking the time to respond though, most appreciated.

Howard

On 8/17/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Wed, Aug 17, 2005 at 07:48:29AM -0400, hugolivude wrote:
> > > > Is there a way around this w/o giving everyone root privileges!
> > >
> > > Do you want to allow every user to delete another user's voicemail?
> > >
> > > If not, how do you sync voicemail users and samba users?
> >
> > I want each user to see, read and write (delete) their own voicemail
> > ONLY (i.e. a user shouldn't be able to listen to someone elses
> > voicemails).  I gave each user an account on the Asterisk box and
> > limited their access to their mailbox folder only.
> 
> So don't waste your time on saving the voicemail on Asterisk. Save it on
> a specific folder in an imap server on the user's home directory.
> 
> If you use a decent mail client, getting notifications for new mails in
> that folders, deleting them, playing them, and whatever should be easy.
> 
> On the Asterisk side you only need to keep voicemail config in sync.
> Maybe it would be easier to just forward every mailbox nnn to
> [EMAIL PROTECTED] and use an aliases file to do the real forwarding. That
> way you keep the emails away from Asterisk's config.
> 
> The downside: no message-waiting indicator.
> 
> --
> Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
> http://tzafrir.org.il |   | a Mutt's
> [EMAIL PROTECTED] |   |  best
> ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] Voicemail file permissions

2005-08-17 Thread hugolivude

I am running Asterisk as its own user/group; at least I followed the
instructions for doing so from
http://www.voip-info.org/wiki-Asterisk+non-root.  Even so new
voicemails get created with owner/group as root.  Maybe I missed
something...

Hugh

On 8/17/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> Tzafrir Cohen wrote:
> > On Wed, Aug 17, 2005 at 07:48:29AM -0400, hugolivude wrote:
> >
> Is there a way around this w/o giving everyone root privileges!
> 
> Run asterisk as its own user/group. We do.
> 
> -Matthew
> 
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RE: [Asterisk-Users] 1-800 number

2005-08-17 Thread Don Fanning

How about a sex line? :)  They never pick up on those. Like
1-800-554-0069

800 numbers still charge the customer but in this case the customer is
the one terminating the 800 service.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christoph
Eicke
Sent: Wednesday, August 17, 2005 1:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] 1-800 number

Hi!

I'm searching for a 1-800 number that simply plays music for a long time
(>3mins) and no one picks up. I've bothered the AT&T lines so far when
trying out my SIP->PSTN connection but then always someone answered :-)
Anyone have a number?

Christoph
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RE: [Asterisk-Users] Any one using the new Digium echocancellation cards

2005-08-17 Thread Richard A. Smith

David,

Yes we got them and they caused huge problems.  The echo training would
cause the line to mute and you would hear something like a dtmf tone briefly
and then you would be connected and talking again.  This might happen once
or 50 times during a call.  I spoke to Digium and they say there may be a
firmware upgrade coming down the line.  You are right it is a daughter board
placed in the space provided on the existing cards.  We disabled the
daughter board and are basically back to where we started from.

Thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David
Zanetti
Sent: Wednesday, August 17, 2005 4:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Any one using the new Digium
echocancellation cards

On Wed, 2005-08-17 at 08:56 -0500, Alan Bunch wrote:
> THe wiki doesn't seem to have any user reports.
>
> If your using them, how are the working, better, worse about the same.
>
> Also what hardware seems to be stable with them installed.

I'd also be interested if the module is available as an upgrade to
existing quad boards. It looks rather like the echo canceller is a
daughter board, connected roughly where there's a connector on existing
quad boards...

Have a rather nastry PRI echo problem which so far no fiddling with
settings (tx, rx, taps) has helped with.

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

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[Asterisk-Users] Choppy Ringing

2005-08-17 Thread Todd Weiser









Hello All,

 

We recently changed our asterisk system to begin using
G.729a as the primary codec.  We have a Cisco 1700-series router which
connects to the PSTN via FXO ports, along with Cisco 7940 SIP phones. 
Everything is working great, except…  When an inbound caller calls
into our system, they hear an IVR.  When the caller dials an ext (SIP
phone), the ringing progress tone is choppy/distorted…  However, the
voice call itself sounds fine.  Asterisk, the Cisco phone, and the call
gateway are all configured to use rfc2833.  From my research, asterisk
generates progress tones out-of-band (I think) unless turned on.  We
don’t have any problems with the progress tones when G.711u is
used.  Any help/ideas would be greatly appreciated.

 

Todd

 






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Re: [Asterisk-Users] Any one using the new Digium echo cancellation cards

2005-08-17 Thread David Zanetti

On Wed, 2005-08-17 at 08:56 -0500, Alan Bunch wrote:
> THe wiki doesn't seem to have any user reports. 
> 
> If your using them, how are the working, better, worse about the same. 
> 
> Also what hardware seems to be stable with them installed.

I'd also be interested if the module is available as an upgrade to
existing quad boards. It looks rather like the echo canceller is a
daughter board, connected roughly where there's a connector on existing
quad boards...

Have a rather nastry PRI echo problem which so far no fiddling with
settings (tx, rx, taps) has helped with.

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

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[Asterisk-Users] How many TDM22P Card can be used on the same PC ?

  
 Is it possible to use 24 FXS/FXO lines(on 6 PCI slots) at the same time on the
same PC? 

I wonder for sound quality and power issues. Can anyone convince me that I
can(not) use 6 TDM22P cards?

Thanks in advance.
BDM.



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RE: [Asterisk-Users] How many TDM22P Card can be used on the same PC ?

2005-08-17 Thread Wiley Siler

Just Google the archive on 'IRQ issues'.  
You can pretty much bet that 6 TDM cards on 6 PCI slots would suck
hugely.
Unless echo is your goal, you are not going to be pleased.

If you have to use 24 existing POTS lines, look into a channel bank and
interface it to a T1 card.
If you are planning new, just get a PRI T1 and be done with it.

Cheers,
W


 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, August 17, 2005 4:38 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How many TDM22P Card can be used on the same
PC ?


 Is it possible to use 24 FXS/FXO lines(on 6 PCI slots) at the same time
on the same PC? 

I wonder for sound quality and power issues. Can anyone convince me that
I
can(not) use 6 TDM22P cards?

Thanks in advance.
BDM.



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[Asterisk-Users] How many TDM22P Card can be used on the same PC ?


 Is it possible to use 24 FXS/FXO lines(on 6 PCI slots) at the same time on the
same PC? 

I wonder for sound quality and power issues. Can anyone convince me that I
can(not) use 6 TDM22P cards?

Thanks in advance.
BDM.



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Re: [Asterisk-Users] Uniden UIP200 Opinions

2005-08-17 Thread David Zanetti

On Fri, 2005-08-05 at 13:30 -0400, Jim Feniello wrote:

> I've read through the archives, and wanted to get an updated opinion
> on the Uniden UIP200 phone.  Seems like there were a lot of opinions
> that it was a good phone, but there were a few items that people were
> waiting for firmware updates for, but that was in 2004.

We've deployed about 50 here. They work, mostly.

Hold works (* does MOH when on hold), transfers kinda work (using the
XFER button, the phone does seem to occasionally get confused afterwards
tho, but * does MOH), DND and Forwarding both work.

But, I would fall short of recommending them. Would really like to see
the transfer problems resolved. That and the documentation is sub-par.

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

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Re: [Asterisk-Users] IP Cop as a firewall and QOS

2005-08-17 Thread Casey Boone

i have never used ipcop in conjunction with asterisk (just starting 
asterisk) but i have used ipcop quite a bit.  ipcop is a GREAT 
alternative to appliance firewalls and does almost anything i  have ever 
needed to do.  it is easy to install and easy to maintain and just 
works.  the hardest thing is the way they name their interfaces and it 
isnt all that hard.


red is the internet port
green is the local lan port
orange (if you have one) is the DMZ
blue (if you have one) is for wireless

typically you just need to worry about red and green

i set up a series of vpn links based on  some cheap celeron 1.7s with 
256megs of ram and i could saturate a 100MBit connection without the ip 
cop machine breaking a sweat (this was during testing,  would have went 
with much lower powered machines except for the great deal the company i 
worked for had for those boxes.)


i would try it and see personally, find a box you are willing to use for 
an ipcop test and go to it.




Casey Boone

Mojo Jojo wrote:
We are looking for a good firewall replacement which will basically do 
pot blocking and QOS.


Our current solution just plain stinks..

We basically need to handle the traffic of a few web servers, mail 
server and asterisk box. The most traffic this device will need to 
handle is what can be shoved through a T1.


I don't mind buying an appliance to get something solid but IP Cop just 
looks better than he appliances I see out there.


I am only concerned if it is stable for a production environment. It 
says it's designed for a SOHO environment, we are doing a bit more than 
that.


Will this thing hold up? Can it be trusted?

Anyone using this for QOS and Asterisk in a production setup.

Any thoughts or suggestions or warnings would be appreciated!

Thanks!

--
Start Your Own Internet Service!
http://www.YourOwnISP.com

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RE: [Asterisk-Users] TDM04B, trunk group

Quoting Sascha Ferley <[EMAIL PROTECTED]>:

> Thanks for the reply
> 
> The phone lines here are already setup in a hunt group from the telco
> provider. The reason I asked was that I saw somewhere, which I can't seem to
> find right now, that one could configure all 4 lines into 1 group, like the
> channels in a PRI. 
> 
> This was basically what I was trying to do.
> 
>  
> 
> S.
> 
>   _  
> 
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of OMS
> Sent: August 17, 2005 3:33 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] TDM04B, trunk group
> 
>  
> 
> I think it should be simple if I understand your question correctly. 
> 
>  
> 
> Make 4 trunks ZAP/g0, ZAP/g1, ZAP/g2,ZAP/g3 in AMP.
> 
>  
> 
> In outbound routing make outbound routing with almost all dial patterns
> 
> Select the trunk sequence starting from the last one in the hunt group.
> 
> If your main number is connected to ZAP/g0, it should be last in the
> sequence.
> 
>  
> 
> In your scenario if will be helpful to get a SIP outbound route for long
> distance. This will reduce your cost and load on ZAP trunks.
> 
>  
> 
> Obaid.
> 
>  
> 
> - Original Message - 
> 
> From: Sascha   Ferley 
> 
> To: 'Asterisk Users Mailing List - 
> Non-Commercial Discussion' 
> 
> Sent: Wednesday, August 17, 2005 4:37 PM
> 
> Subject: [Asterisk-Users] TDM04B, trunk group
> 
>  
> 
> Hi, 
> 
> I am just trying to figure out how to setup a TDM04B card for
> incoming/outgoing calls. I have 4 lines, which are provided as a rotary
> trunk group, currently hooked into a Nortel system, which asterisk will
> replace.  I have setup a Dell 1800 (Tower) system with the TDM04B card,
> which seems to work. 
> 
>  
> 
> The question is how do I set it up that all 4 lines are part of a trunk
> group, such that all 4 lines can be used for incoming aswell as outgoing
> calls?
> 
>  
> 
> I am using [EMAIL PROTECTED] 1.4. 
> 
>  
> 
> Please let me know
> 
> Thanks
> 
> S.
> 
> 
>   _  
> 
> 
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> 



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RE: [Asterisk-Users] TDM04B, trunk group

2005-08-17 Thread Sascha Ferley









Thanks for the reply

The phone lines here are already setup in
a hunt group from the telco provider. The reason I asked was that I saw
somewhere, which I can’t seem to find right now, that one could configure
all 4 lines into 1 group, like the channels in a PRI. 

This was basically what I was trying to
do.

 

S.









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of OMS
Sent: August 17, 2005 3:33 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
TDM04B, trunk group



 



I think it should be simple if I understand your question
correctly. 





 





Make 4 trunks ZAP/g0, ZAP/g1, ZAP/g2,ZAP/g3 in
AMP.





 





In outbound routing make outbound routing with
almost all dial patterns





Select the trunk sequence starting from the last one in the
hunt group.





If your main number is connected to ZAP/g0, it should be
last in the sequence.





 





In your scenario if will be helpful to get
a SIP outbound route for long distance. This will reduce your cost and
load on ZAP trunks.





 





Obaid.





 







- Original Message - 





From: Sascha
Ferley 





To: 'Asterisk Users Mailing List -
Non-Commercial Discussion' 





Sent: Wednesday, August
17, 2005 4:37 PM





Subject: [Asterisk-Users]
TDM04B, trunk group





 



Hi, 

I am just trying to figure out how to setup a TDM04B card
for incoming/outgoing calls. I have 4 lines, which are provided as a rotary
trunk group, currently hooked into a Nortel system, which asterisk will
replace.  I have setup a Dell 1800 (Tower) system with the TDM04B card,
which seems to work. 

 

The question is how do I set it up that all 4 lines are part
of a trunk group, such that all 4 lines can be used for incoming aswell as
outgoing calls?

 

I am using [EMAIL PROTECTED] 1.4. 

 

Please let me know

Thanks

S.







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RE: [Asterisk-Users] IP Cop as a firewall and QOS

2005-08-17 Thread Wiley Siler

There are a dozen Linux based methods ranging from.  Personally I like
the Mandrake offering called Multi-Network Firewall.  It is pretty
turnkey and they have it available for download.  It also supports
bonding which allows you to use multiple nics bonded together and views
as one connection.
http://www.mandriva.com/business/mnf2

Other than that, like I said, there are dozens...

W


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo Jojo
Sent: Wednesday, August 17, 2005 3:27 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] IP Cop as a firewall and QOS

We are looking for a good firewall replacement which will basically do
pot blocking and QOS.

Our current solution just plain stinks..

We basically need to handle the traffic of a few web servers, mail
server and asterisk box. The most traffic this device will need to
handle is what can be shoved through a T1.

I don't mind buying an appliance to get something solid but IP Cop just
looks better than he appliances I see out there.

I am only concerned if it is stable for a production environment. It
says it's designed for a SOHO environment, we are doing a bit more than
that.

Will this thing hold up? Can it be trusted?

Anyone using this for QOS and Asterisk in a production setup.

Any thoughts or suggestions or warnings would be appreciated!

Thanks!

--
Start Your Own Internet Service!
http://www.YourOwnISP.com

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Re: [Asterisk-Users] How "real time" is realtime?

2005-08-17 Thread Mattt







  You didn't even bother to read the last 24 hours of list mail before
asking this?!?

  (Hint: read the last 24 hours' worth of list mail...)


Asterisk Supporter wrote:

  How "real time" is realtime?  If the extensions.conf is stored in the
database, does * query it row by row or is it "cached"?  In other words,
given the following exerpt:

exten => 5001,1,Dial(IAX2/[EMAIL PROTECTED]/s,30,g)
exten => 5001,2,Voicemail(u5001)
exten => 5001,102,Voicemail(b5001)
exten => 5001,103,Hangup

exten => 5002,1,Dial(IAX2/[EMAIL PROTECTED]/s,30)
exten => 5002,2,Voicemail(u5002)
exten => 5002,102,Voicemail(b5002)
exten => 5002,103,Hangup


If 5001 received a call at 5001,1 and was answered can row 5001,2 be
replaced with:

exten => 5001,2,Goto(5002,1)

Overlooking syntax and typos in the dial plan, on the extension 5001
hangup ("g "attribute) will the caller be "transferred" to voicemail or
extension 5002?  Can I alter the dialplan in "realtime" as long as the row
has not been processed?




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-- 
Cheers,
 Mattt.

 VoIP made easy - http://voip.abovenetworks.net
 Convergent network specialists - http://abovenetworks.net

I have an inferiority complex, but it's not a very good one...



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Re: [Asterisk-Users] TDM04B, trunk group

2005-08-17 Thread OMS

I think it should be simple if I understand your
question correctly.
Make 4 trunks ZAP/g0, ZAP/g1,
ZAP/g2,ZAP/g3 in AMP.
In outbound routing make outbound routing with
almost all dial patterns
Select the trunk sequence starting from the last
one in the hunt group.
If your main number is connected to ZAP/g0, it
should be last in the sequence.
In your scenario if will be helpful
to get a SIP outbound route for long distance. This will reduce your cost
and load on ZAP trunks.
Obaid.

- Original Message -
From:
Sascha Ferley
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Wednesday, August 17, 2005 4:37
PM
Subject: [Asterisk-Users] TDM04B, trunk
group

Hi,
I am just trying to figure out how
to setup a TDM04B card for incoming/outgoing calls. I have 4 lines, which are
provided as a rotary trunk group, currently hooked into a Nortel system, which
asterisk will replace. I have setup a Dell 1800 (Tower) system with the
TDM04B card, which seems to work.
The question is how do I set it up
that all 4 lines are part of a trunk group, such that all 4 lines can be used
for incoming aswell as outgoing calls?
I am using [EMAIL PROTECTED] 1.4.

Please let me
know
Thanks
S.

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[Asterisk-Users] How "real time" is realtime?

2005-08-17 Thread Asterisk Supporter

How "real time" is realtime?  If the extensions.conf is stored in the
database, does * query it row by row or is it "cached"?  In other words,
given the following exerpt:

exten => 5001,1,Dial(IAX2/[EMAIL PROTECTED]/s,30,g)
exten => 5001,2,Voicemail(u5001)
exten => 5001,102,Voicemail(b5001)
exten => 5001,103,Hangup

exten => 5002,1,Dial(IAX2/[EMAIL PROTECTED]/s,30)
exten => 5002,2,Voicemail(u5002)
exten => 5002,102,Voicemail(b5002)
exten => 5002,103,Hangup


If 5001 received a call at 5001,1 and was answered can row 5001,2 be
replaced with:

exten => 5001,2,Goto(5002,1)

Overlooking syntax and typos in the dial plan, on the extension 5001
hangup ("g "attribute) will the caller be "transferred" to voicemail or
extension 5002?  Can I alter the dialplan in "realtime" as long as the row
has not been processed?




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[Asterisk-Users] IP Cop as a firewall and QOS

2005-08-17 Thread Mojo Jojo

We are looking for a good firewall replacement which will basically do pot 
blocking and QOS.


Our current solution just plain stinks..

We basically need to handle the traffic of a few web servers, mail server 
and asterisk box. The most traffic this device will need to handle is what 
can be shoved through a T1.


I don't mind buying an appliance to get something solid but IP Cop just 
looks better than he appliances I see out there.


I am only concerned if it is stable for a production environment. It says 
it's designed for a SOHO environment, we are doing a bit more than that.


Will this thing hold up? Can it be trusted?

Anyone using this for QOS and Asterisk in a production setup.

Any thoughts or suggestions or warnings would be appreciated!

Thanks!

--
Start Your Own Internet Service!
http://www.YourOwnISP.com

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[Asterisk-Users] AstriCon Update: Early Bird Ends Soon - Free Asterisk Book

2005-08-17 Thread Steven Sokol

[Early Bird Ends Soon]
==

AstriCon early-bird registration ends next Thursday, August 25th. 
Early-bird registration saves 20% ($110.00 USD) off the standard price
of an AstriCon "All Access" Pass.  This pass includes the
pre-conference events (either the Meet Asterisk! seminar or the
Asterisk Developers meeting), the tutorial day, both conference days,
the exhibit hall, and more.

[Free Book - "Asterisk: The Future of Telephony"]
=

O'Reilly (the largest and most respected publisher of computer books)
has agreed to give a copy of "Asterisk: The Future of Telephony" to
the first 500 people to attend the AstriCon tutorials.  This new book
by Jim Van Meggelen, Leif Madsen, Jared Smith (of the Asterisk
Documentation Project: http://www.asteriskdocs.org) is a world-class
guide to installing and running Asterisk.

To receive a copy of the book you need to attend the AstriCon Tutorial
sessions on Wednesday, October 12.  Attendees with AstriCon "All
Access" Passes, Tutorial + Conference Passes, or Tutorials Only passes
qualify.
[Very Few IAXys Remain]
===

We are down to the last handful of free IAXys.  Digium is giving away
50 free IAXys to the first 50 to register for the full conference
pass.  The last of these IAX2-based ATA devices (valued at $100) will
go to the next few people to register, so sign up now and with any
luck you will leave AstriCon with an IAXy.

[Register Now]
==

For more information on AstriCon, see our web site at
http://www.astricon.net/2005. To register for AstriCon 2005, use the
link below:

  https://www.astricon.net/2005/register/index.php

If you have already registered and want to make hotel reservations at
the discount rate, ($114/night) please use the following link:

  https://www.astricon.net/2005/hotel.shtml

(Click on the "Special Rate" link on the Hotel & Travel page to
reserve your room.)
-- 
Steven Sokol
CEO/Manager
Sokol & Associates, LLC

Ask Me About AstriCon 2005!
http://www.astricon.net/
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Re: [Asterisk-Users] Patchy audio to and from VOIPBUSTER

2005-08-17 Thread Sean Rima

Sean Rima wrote:
> Sean Rima wrote:
>> I was playing with using VOIPBUSTER and was testing their client, which
>> I think is SIP. So I added the only setup I could find for asterisk
>> which is iax2 but I found that the speech quality is poor compared to
>> the client and there is a delay of almost 1 second whereas their client
>> there is not a real noticeable delay. Should I try with SIP. I am using
>> ISDN 64K dialup if that makes any difference
>>
>>
>> Just tried SIP and the same problem exists. Currently the Asterisk box
>> is behind a Firewall on a ISDN dialup connection which sadly the IP
>> changes. Is there anything that I can try
>>
> To my sip.conf I added the outside_addr=tcob1.no-ip.com, which I use for
> other things, but it is still very patchy, and the delay is now about 2
> to 3 seconds
> 

Done a bit of research and discovered that there is nothign I can do
until I get my new ISDN card for the Asterisk PC, cannot use the one on
the Windows PC as it is USB and soft at that :(

Sean

-- 
ICQ: 679813FidoNet: 2:263/950
Jabber: [EMAIL PROTECTED] AOL: tcobone
Vodafone Messenger: thecivvie
FreeWorldDial 689482


smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [Asterisk-Users] Patchy audio to and from VOIPBUSTER

2005-08-17 Thread Sean Rima

Sean Rima wrote:
> I was playing with using VOIPBUSTER and was testing their client, which
> I think is SIP. So I added the only setup I could find for asterisk
> which is iax2 but I found that the speech quality is poor compared to
> the client and there is a delay of almost 1 second whereas their client
> there is not a real noticeable delay. Should I try with SIP. I am using
> ISDN 64K dialup if that makes any difference
> 
> 
> Just tried SIP and the same problem exists. Currently the Asterisk box
> is behind a Firewall on a ISDN dialup connection which sadly the IP
> changes. Is there anything that I can try
> 
To my sip.conf I added the outside_addr=tcob1.no-ip.com, which I use for
other things, but it is still very patchy, and the delay is now about 2
to 3 seconds

Sean
-- 
ICQ: 679813FidoNet: 2:263/950
Jabber: [EMAIL PROTECTED] AOL: tcobone
Vodafone Messenger: thecivvie
FreeWorldDial 689482


smime.p7s
Description: S/MIME Cryptographic Signature
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[Asterisk-Users] TDM04B, trunk group

2005-08-17 Thread Sascha Ferley









Hi, 

I am just trying to figure out how to setup a TDM04B card
for incoming/outgoing calls. I have 4 lines, which are provided as a rotary
trunk group, currently hooked into a Nortel system, which asterisk will
replace.  I have setup a Dell 1800 (Tower) system with the TDM04B card, which
seems to work. 

 

The question is how do I set it up that all 4 lines are part
of a trunk group, such that all 4 lines can be used for incoming aswell as
outgoing calls?

 

I am using [EMAIL PROTECTED] 1.4. 

 

Please let me know

Thanks

S.






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[Asterisk-Users] Patchy audio to and from VOIPBUSTER

2005-08-17 Thread Sean Rima

I was playing with using VOIPBUSTER and was testing their client, which
I think is SIP. So I added the only setup I could find for asterisk
which is iax2 but I found that the speech quality is poor compared to
the client and there is a delay of almost 1 second whereas their client
there is not a real noticeable delay. Should I try with SIP. I am using
ISDN 64K dialup if that makes any difference


Just tried SIP and the same problem exists. Currently the Asterisk box
is behind a Firewall on a ISDN dialup connection which sadly the IP
changes. Is there anything that I can try

Sean
-- 
ICQ: 679813FidoNet: 2:263/950
Jabber: [EMAIL PROTECTED] AOL: tcobone
Vodafone Messenger: thecivvie
FreeWorldDial 689482


smime.p7s
Description: S/MIME Cryptographic Signature
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[Asterisk-Users] CAPI problem - need help

2005-08-17 Thread Arik Funke

I have installed a Fritz card which I use with chan_capi. If the card is 
CALLED, everything works perfectly well.


BUT: If the card is CALLING, it only sends audio but does not receive 
it. I have already changed the card, the remote devices etc. I am 
running out of ideas.


Does anybody know this phenomena? I would really appreciate any ideas I 
could try...


Cheers,
Arik
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Re: [Asterisk-Users] DECT gateways

On 17:07, Wed 17 Aug 05, Sean Milheim (iDREUS Corporation) wrote:
> Check out
> 
> H.323 / SIP DECT VoIP Router:
> http://www.idreus.com/?page=product&action=view&id=243
> 
> Working out a few bugs with asterisk integration though.  Having issues with
> receiving incoming calls.  However if you would like to be informed when
> this is fully working with * please email me off list.
> 

Nice looking device.
Does it support DECT repeaters?
I cannot rely on 1 basestation for my handset when I walk
around in the location. The Siemens stuff has 10
antennas/repeaters/extenders in the building. It's a bit
overkill, but the Siemens guys tend to love doing it BIG.
I think 3 to 6 repeaters will be way enough for the cases we
have open now.

I have to know this before considering this device as an
option, sorry. Not for me (sending mail offlist cause I want
1 at home ;) ) but for the customers.
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [Asterisk-Users] DECT gateways

2005-08-17 Thread Andreas Reich

Michiel van Baak wrote:
> They now have a Siemens or a Samsung PBX. Those PBX-es come
> with a DECT basestation and optionally repeaters etc.
> All those basestations speak some own protocol to the PBX,
> so we cannot use them with asterisk.

Why don't you use the existing PBX and connect it to Asterisk via S2M?

I'm running a DECT PBX myself and it has a "sub-PBX" feature. In that 
mode it will act as a gateway between ISDN (Asterisk) and the DECT phones.
So if you dial 1234 on a DECT phone, it will first check if there is a 
DECT extension with number 1234 assigned to it. If there isn't, it calls 
1234 on the ISDN line.

For me, that's exactly what I need to interconnect DECT phones and Asterisk.

Andreas

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[Asterisk-Users] [Fwd: [SA16438] Grandstream BudgeTone Denial of Service Vulnerability]

2005-08-17 Thread Ing CIP Alejandro Celi Mariátegui




-- 
Ing CIP Alejandro Celi Mariátegui 
<[EMAIL PROTECTED]>


-Mensaje reenviado-
From: Secunia Security Advisories <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: [SA16438] Grandstream BudgeTone Denial of Service Vulnerability
Date: Mon, 15 Aug 2005 12:49:44 +0200


--

Bist Du interessiert an einem neuen Job in IT-Sicherheit?


Secunia hat zwei freie Stellen als Junior und Senior Spezialist in IT-
Sicherheit:
http://secunia.com/secunia_vacancies/

--

TITLE:
Grandstream BudgeTone Denial of Service Vulnerability

SECUNIA ADVISORY ID:
SA16438

VERIFY ADVISORY:
http://secunia.com/advisories/16438/

CRITICAL:
Less critical

IMPACT:
DoS

WHERE:
>From local network

OPERATING SYSTEM:
Grandstream BudgeTone 100 Series SIP Phones
http://secunia.com/product/5537/

DESCRIPTION:
Pierre Kroma has reported a vulnerability in Grandstream BudgeTone
100 Series SIP Phones, which can be exploited by malicious people to
cause a DoS (Denial of Service).

The vulnerability is caused due to an error when processing large UDP
datagrams and can be exploited by sending a large UDP datagram (more
than 65534 bytes) to port 5060/udp.

Successful exploitation causes the phone to stop working by aborting
active calls, blank the display, and make the integrated HTTP server
become inaccessible.

The vulnerability has been reported in firmware release 1.0.6.7.
Other versions may also be affected.

SOLUTION:
Use the phones on trusted networks only.

PROVIDED AND/OR DISCOVERED BY:
Pierre Kroma, SySS.

--

About:
This Advisory was delivered by Secunia as a free service to help
everybody keeping their systems up to date against the latest
vulnerabilities.

Subscribe:
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Definitions: (Criticality, Where etc.)
http://secunia.com/about_secunia_advisories/


Please Note:
Secunia recommends that you verify all advisories you receive by
clicking the link.
Secunia NEVER sends attached files with advisories.
Secunia does not advise people to install third party patches, only
use those supplied by the vendor.

--

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RE: [Asterisk-Users] DECT gateways

2005-08-17 Thread Sean Milheim \(iDREUS Corporation\)

Check out

H.323 / SIP DECT VoIP Router:
http://www.idreus.com/?page=product&action=view&id=243

Working out a few bugs with asterisk integration though.  Having issues with
receiving incoming calls.  However if you would like to be informed when
this is fully working with * please email me off list.


Regards, 
 
Sean Milheim
iDREUS Corporation


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: Wednesday, August 17, 2005 4:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] DECT gateways


On 08:06, Thu 18 Aug 05, Richard Malcolm-Smith wrote:
> Michiel van Baak wrote:
> >Is there any other solution like this out there that works with 
> >asterisk ?
> 
> If you find something, I would be interested in the outcome.
> 
> I want something for the house here, at the moment I just have 2 
> analog
> dect bases plugged into the same line, but you cant roam between them so
if 
> I want to walk from one end of the section to the other I have to park the

> call, change base station and pick it up again, and then also multi
handset 
> doesnt work between bases etc etc.
> 
> I would ideally like a dect network that plugs into the lan and then 
> the
> handsets each register as an individual sip extension regardless of what 
> base they are on at the time. I could really use a couple more bases
around 
> the place here to get good coverage.

That's exactly what I want too.
All I found was the Kirk, wich let every handset register as SCCP device. I
don't care wether it's SIP or SCCP, but before I buy it and sell it to the
customer I want to know if it is going to work with asterisk or only with
CCM.

Remco Barendse, aren't you using the Kirk system (found some hit on google)
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"



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RE: [Asterisk-Users] DECT gateways

2005-08-17 Thread Dean Collins

Yep wifi doesn't do base to base roaming, stick with dect for the next
12-18 months until wifi gets it's act together.

You might also like to check out Nira (or ascom nira) as they have a
standalone dect solution if you are looking to rollout a truly large
system for a manufacturing/warehouse (hell I've seen an ericsson md110
being used as a standalone for 1 installation (a little over 40 million
square feet of coverage with 300 handsets) but I somehow don't think
this is the league you are talking about).

Cheers,
Dean
p.s. yes I know that there are 2 wifi base to base roaming solutions
available but they are dammed expensive for what you are getting and
dect is a perfectly good solution for today.



> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michiel van Baak
> Sent: Wednesday, 17 August 2005 2:51 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] DECT gateways
> 
> On 20:42, Wed 17 Aug 05, Yoann Le Bihan wrote:
> > 2005/8/17, Michiel van Baak <[EMAIL PROTECTED]>:
> > > Is there any other solution like this out there that works
> > > with asterisk ?
> >
> > Why don't you use WiFi VoIP phones like ZyXEL 2000W (which is not
such
> > expensive compared with Cisco ones...) ?
> >
> > Best regards,
> >
> 
> There are no cisco phones involved. The kirk handsets will
> be recognized by CCM as cisco phones, but they are not.
> That's what their website said. I have chan-sccp-cm running
> with some 7905's and it does the job real fine. So if those
> handsets work ok, it will do the trick for our customers.
> 
> I've been thinking about WIFI phones, but some important,
> possible issues come to mind.
> 
> People want wireless phones so they can walk around while
> calling. So the phones should do accesspoint roaming while
> in a call and not getting disconnected. Is that possible ?
> Besides that, DECT has encryption, and everywhere I read
> about WIFI phones, I hear they don't provide normal
> callquality when WEP is enabled.
> 
> So that's why I was looking for a dect solution.
> --
> Michiel van Baak
> http://michiel.vanbaak.info
> [EMAIL PROTECTED]
> GnuPG key:
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
> 
> "Why is it drug addicts and computer afficionados are both called
users?"
> 
> ___
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RE: [Asterisk-Users] Comfort Noise incomplete - No translator pathexists for channel type MGCP (native 4) to 256

2005-08-17 Thread Joshua Colp









Hello,

 

It’s trying to do a conversion
between ULAW and G729, and isn’t able to do it (probably because you don’t
have the G729 codec with licenses). 

 

Actual Error:

Aug 17 09:35:28 WARNING[26024]: channel.c:2175 ast_request: No
translator path exists for channel type MGCP (native 4) to 256

Translated Error:

I can’t convert from ULAW to G729, fool!

 

Joshua Colp.

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kurt turner
Sent: Wednesday, August 17, 2005
5:46 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Comfort
Noise incomplete - No translator pathexists for channel type MGCP (native 4) to
256



 



I had
MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable -
wanted to try red hat and got the below message - then I re-installed debian
and am still getting the same message below - any comments are greatly
appreciated - I did play with the config files with no prevail - the Adit seems
to be doing its job per tech support at CAC.  I listed my conigs below





 





 





I go off hook and get this - 





 





*CLI> 
    -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED])
created in state: Down





*CLI> 





 





I dial 123 (an extension) and get this -






*CLI> Aug 17 09:35:25 NOTICE[26024]: rtp.c:284 process_rfc3389: Comfort
noise support incomplete in Asterisk (RFC 3389).  Please turn off on
client if possible. Client IP: 192.168.0.241
    -- Executing Dial("MGCP/aaln/[EMAIL PROTECTED]",
"MGCP/aaln/[EMAIL PROTECTED]")
in new stack
Aug 17 09:35:28 WARNING[26024]: channel.c:2175 ast_request: No translator path
exists for channel type MGCP (native 4) to 256
Aug 17 09:35:28 NOTICE[26024]: app_dial.c:1091 dial_exec_full: Unable to create
channel of type 'MGCP' (cause 0 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Congestion("MGCP/aaln/[EMAIL PROTECTED]",
"") in new stack





*CLI> 





 





I hang up and get this -





 





*CLI> 
  == Spawn extension (outbound-default, 96017209841, 2) exited non-zero on
'MGCP/aaln/[EMAIL PROTECTED]'





 





here are the configs





 





:/etc/asterisk#
more extensions.conf 

[general]

static=yes

writeprotect=no

autofallthrough=yes

[globals]

 

 

 

[extensions]


exten
=> 123,1,Dial,MGCP/aaln/[EMAIL PROTECTED] 

exten
=> 124,1,Dial,MGCP/aaln/[EMAIL PROTECTED]

[directdial]

ignorepat
=> 9

exten
=> 9,1,MGCP/aaln/[EMAIL PROTECTED]

exten
=> 9,2,Congestion

 

[international]

ignorepat
=> 9

exten
=> _9011.,1,Dial(Zap/g2/${EXTEN:1})

exten
=> _9011.,2,Congestion

include
=> longdistance

[longdistance]

ignorepat
=> 9

exten
=> _91NXXNXX,1,Dial(Zap/g2/${EXTEN:1})

exten
=> _91NXXNXX,2,Congestion

include
=> local

[local]

ignorepat
=> 9

exten
=> _9NXXNXX,1,Dial,MGCP/aaln/[EMAIL PROTECTED]

exten
=> _9NXXNXX,2,Congestion

 

[outbound-default]

include =
extensions 

include =
directdial 

include =
longdistance

include =
local

IPD:/etc/asterisk#


 

 

 

***

 

:/etc/asterisk#
more mgcp.conf

; MGCP
Configuration for Asterisk

[general]

port=2727

bindaddr=0.0.0.0

allow=ulaw

allow=g729

allow=g726

tos=0x85

srvlookup=yes

wcardep=aaln/*

 

; Bob's
CMG #1

[192.168.0.241]


context=outbound-default


host=192.168.0.241

wcardep=*

line
=> *

;

; Line 1

;

callerid
= "John" <123> 

callgroup=0


pickupgroup=0

nat=no 

threewaycalling=yes


transfer=yes
; transfer requires threewaycalling=yes. Use FLASH to transfer

callwaiting=yes
; this might be a cause of trouble for ip10s

cancallforward=yes


line
=> aaln/1

;

; Line 2

; 

callerid
= "Jane" <124> 

callgroup=0


pickupgroup=0

nat=no 

threewaycalling=yes


transfer=yes
; transfer requires threewaycalling=yes. Use FLASH to transfer

callwaiting=yes
; this might be a cause of trouble for ip10s

cancallforward=yes

line
=> aaln/2 

;outbound
t1 port 5-6 cross connected to cmg 6:1:1:3-4

 

line
=> aaln/3

line
=> aaln/4

 









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[Asterisk-Users] Comfort Noise incomplete - No translator path exists for channel type MGCP (native 4) to 256

2005-08-17 Thread kurt turner


I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC.  I listed my conigs below
 
 
I go off hook and get this - 
 
*CLI>     -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
*CLI> 
 
I dial 123 (an extension) and get this -
*CLI> Aug 17 09:35:25 NOTICE[26024]: rtp.c:284 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389).  Please turn off on client if possible. Client IP: 192.168.0.241    -- Executing Dial("MGCP/aaln/[EMAIL PROTECTED]", "MGCP/aaln/[EMAIL PROTECTED]") in new stackAug 17 09:35:28 WARNING[26024]: channel.c:2175 ast_request: No translator path exists for channel type MGCP (native 4) to 256Aug 17 09:35:28 NOTICE[26024]: app_dial.c:1091 dial_exec_full: Unable to create channel of type 'MGCP' (cause 0 - Unknown)  == Everyone is busy/congested at this time (1:0/0/1)    -- Executing Congestion("MGCP/aaln/[EMAIL PROTECTED]", "") in new stack
*CLI> 
 
I hang up and get this -
 
*CLI>   == Spawn extension (outbound-default, 96017209841, 2) exited non-zero on 'MGCP/aaln/[EMAIL PROTECTED]'
 
here are the configs
 

:/etc/asterisk# more extensions.conf 
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
 
 
 
[extensions] 
exten => 123,1,Dial,MGCP/aaln/[EMAIL PROTECTED] 
exten => 124,1,Dial,MGCP/aaln/[EMAIL PROTECTED]
[directdial]
ignorepat => 9
exten => 9,1,MGCP/aaln/[EMAIL PROTECTED]
exten => 9,2,Congestion
 
[international]
ignorepat => 9
exten => _9011.,1,Dial(Zap/g2/${EXTEN:1})
exten => _9011.,2,Congestion
include => longdistance
[longdistance]
ignorepat => 9
exten => _91NXXNXX,1,Dial(Zap/g2/${EXTEN:1})
exten => _91NXXNXX,2,Congestion
include => local
[local]
ignorepat => 9
exten => _9NXXNXX,1,Dial,MGCP/aaln/[EMAIL PROTECTED]
exten => _9NXXNXX,2,Congestion
 
[outbound-default]
include = extensions 
include = directdial 
include = longdistance
include = local
IPD:/etc/asterisk# 
 
 
 
***
 
:/etc/asterisk# more mgcp.conf
; MGCP Configuration for Asterisk
[general]
port=2727
bindaddr=0.0.0.0
allow=ulaw
allow=g729
allow=g726
tos=0x85
srvlookup=yes
wcardep=aaln/*
 
; Bob's CMG #1
[192.168.0.241] 
context=outbound-default 
host=192.168.0.241
wcardep=*
line => *
;
; Line 1
;
callerid = "John" <123> 
callgroup=0 
pickupgroup=0
nat=no 
threewaycalling=yes 
transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer
callwaiting=yes ; this might be a cause of trouble for ip10s
cancallforward=yes 
line => aaln/1
;
; Line 2
; 
callerid = "Jane" <124> 
callgroup=0 
pickupgroup=0
nat=no 
threewaycalling=yes 
transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer
callwaiting=yes ; this might be a cause of trouble for ip10s
cancallforward=yes
line => aaln/2 
;outbound t1 port 5-6 cross connected to cmg 6:1:1:3-4
 
line => aaln/3
line => aaln/4
 
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Re: [Asterisk-Users] DECT gateways

On 08:06, Thu 18 Aug 05, Richard Malcolm-Smith wrote:
> Michiel van Baak wrote:
> >Is there any other solution like this out there that works
> >with asterisk ?
> 
> If you find something, I would be interested in the outcome.
> 
> I want something for the house here, at the moment I just have 2 analog 
> dect bases plugged into the same line, but you cant roam between them so if 
> I want to walk from one end of the section to the other I have to park the 
> call, change base station and pick it up again, and then also multi handset 
> doesnt work between bases etc etc.
> 
> I would ideally like a dect network that plugs into the lan and then the 
> handsets each register as an individual sip extension regardless of what 
> base they are on at the time. I could really use a couple more bases around 
> the place here to get good coverage.

That's exactly what I want too.
All I found was the Kirk, wich let every handset register as
SCCP device. I don't care wether it's SIP or SCCP, but
before I buy it and sell it to the customer I want to know
if it is going to work with asterisk or only with CCM.

Remco Barendse, aren't you using the Kirk system (found some
hit on google)
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"



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Re: [Asterisk-Users] calling number type

2005-08-17 Thread Ray Van Dolson

On Tue, Aug 16, 2005 at 10:37:01AM -0600, Damon Estep wrote:
> 
>Is there a method in SIP to set the CALLING number type to national and the
>calling  number plan to isdn? I am dealing with an issue where a media
>gateway is not sending the correct values and would like to know if SIP has
>an equivalent parameter that can be set and mapped in the media gateway
>sip-isdn translations.

Just out of curiosity, who is your provider?  We recently ran into the same
thing.  Setting the call type to "Unknown" is our provider's way of
identifying 911 calls.

Since there was no way to send this via SIP, they will most likely have to
give us some custom dial string target to pass to them during which their ISDN
interface will set the call type properly for us.

Would be interested if you find a way to do this with SIP only however.

Ray
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[Asterisk-Users] Voicemail/Directory, one person, one box, two last names

2005-08-17 Thread Ryan Stark

So one of my employees just got married and is taking her husband's
last name.  As soon as her business cards run out she will be using
the new last name for business as well.  So for example is there a way
to make it so that if someone goes to the directory to find her
extension and they dial DOE it matches her or if they type SMI it
matches her.  Alternatively, can I make a new voicemail entry for her
new last name and make it put her messages in her other box, sort of
like an alias?

Thanks,
-Ryan
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Re: [Asterisk-Users] DECT gateways

2005-08-17 Thread Richard Malcolm-Smith


Michiel van Baak wrote:

Is there any other solution like this out there that works
with asterisk ?


If you find something, I would be interested in the outcome.

I want something for the house here, at the moment I just have 2 analog dect 
bases plugged into the same line, but you cant roam between them so if I want to 
walk from one end of the section to the other I have to park the call, change 
base station and pick it up again, and then also multi handset doesnt work 
between bases etc etc.


I would ideally like a dect network that plugs into the lan and then the 
handsets each register as an individual sip extension regardless of what base 
they are on at the time. I could really use a couple more bases around the place 
here to get good coverage.


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[Asterisk-Users] chan_sip2.c compiling

2005-08-17 Thread Tomáš Komárek

Hello, I've tried to compile the new sip channel, sip_chan2.c but I am 
not succesfull. When I make * I get error messages, some of them also 
considering syntax error in the code.


Does anyone use this channel? Wuld you please give me some advices how 
to compile it?


Or do you have the source code that works???

Thanks for answers.

Tomas
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Re: [Asterisk-Users] iaxcomm huge latency

2005-08-17 Thread Michael Van Donselaar

On Wed, 17 Aug 2005 14:34:26 +0200, Juraj Bednar <[EMAIL PROTECTED]> wrote:

>Hello,
>
>   I use iaxcomm-latest from the iaxclient.sf.net page (binary
>release) on linux, also tried Mac OS X version with the same result
>and Asterisk 1.0.9 from Debian. Iaxcomm has a huge latency -- tens of
>seconds, constantly changing over time. It was run on two different
>machines, always to a SIP phone (which otherwise works correctly even
>with VoipBuster, which also uses IAX with no latency and other SIP
>phones). Is it a known bug?

What results do you get when the iaxcomms call each other (both via the asterisk
server and peer to peer)?

There have been some changes to the jitterbuffer code in the iaxclient library
since those iaxcomm binaries were posted.  You might also want to compile the
CVS code.

>
>
>  Juraj.
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Re: [Asterisk-Users] DUNDi Install

On Wed, Aug 17, 2005 at 02:38:04PM -0500, Justin Selleck wrote:
> Does any one know how to install DUNDi on the v1-0 CVS export?
> 

get HEAD, basically. AFAIK there isn't an easier way, as DUNDi is not
availble for 1.0.

-- 
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http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
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Re: [Asterisk-Users] AGI SCRIPTS USING PERL NEED SOME KIND OF COMPILATION TO WORK WITH *

On Wed, Aug 17, 2005 at 04:15:52PM -0300, j_amorim wrote:
Content-Description: Mail message body
> Hi all, 
> 
> Help needed: 
> 
> Does AGI  SCRIPTS USING PERL NEED's SOME KIND OF COMPILATION TO WORK WITH 
> * 

No.

And BTW, when you ask questions about them you don't need to SHOUT.

> 
> This simple script is not working. 

How did you set it up? Any error messages?

-- 
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RE: [Asterisk-Users] X100P dial out problem

2005-08-17 Thread Jason Walker

Shot in the dark

Do you have to dial '9' on your outside line?

Perhaps if you changed your Dial command to this:

[outgoing]
exten => _9X.,1,NoOp("Call for "${EXTEN})
exten => _9X.,2,Dial(Zap/1/${EXTEN:1})

The :1 will drop the leading '9' when it hits the outside. If this is a
regular line, there should be no need for the '9'.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Piero Baudino
Sent: Wednesday, August 17, 2005 12:13 PM
To: Tzafrir Cohen
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] X100P dial out problem

Hi Tzafrir,

thanks for your reply...
Here is what happens when I make the call:

pbx*CLI> zap show channels
   Chan Extension  Context Language   MusicOnHold
 pseudoincomingit
  1incomingit
-- Executing NoOp("SIP/6601-5d39", ""Call for "91234567") in new
stack
-- Executing Dial("SIP/6601-5d39", "Zap/1/91234567") in new stack
-- Called 1/91234567
-- Zap/1-1 answered SIP/6601-5d39
-- Hungup 'Zap/1-1'
  == Spawn extension (x-lite, 912334567, 2) exited non-zero on
'SIP/6601-5d39'
-- Unregistered SIP '6601'
pbx*CLI> exit

The Hangup happens when I hangup from XLITE.

Here is my conf:
/etc/asterisk/zapata.conf

[channels]
language=it
signalling=fxs_ks
context=incoming
channel=>1

/etc/asterisk/extensions.conf
[incoming]
exten => s,1,Dial(SIP/6601&SIP/6602&SIP/6603,20,tr)  ; corresponding
clients must be configured in sip.conf
exten => s,2,Playback(vm-goodbye)
exten => s,3,Hangup

[outgoing]
exten => _9X.,1,NoOp("Call for "${EXTEN})
exten => _9X.,2,Dial(Zap/1/${EXTEN})

[x-lite]  ; Note: SIP extensions are defined here as "66" followed by any
two digits
exten => _66XX,1,NoOp("Call for "${EXTEN})
exten => _66XX,2,Dial(SIP/${EXTEN})
exten => _66XX,3,Congestion
include => outgoing

/etc/asterisk/sip.conf
port=5060
context=default
srvlookup=yes
dtmfmode=inband
allow=aLaw
allow=uLaw
allow=gsm

[6601]
type=friend
secret=password
host=dynamic
;dtmfmode=rfc2833
context=x-lite
callerid="Piero" <6601>
allow=aLaw
allow=uLaw
allow=gsm

Thanks.
PieroB

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[Asterisk-Users] DUNDi Install

2005-08-17 Thread Justin Selleck









Does any one know how to install DUNDi on the v1-0 CVS
export?

 

I’m not sure what to download, what packages I need,
etc.

 

Thanks!

 

-Justin






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RE: [Asterisk-Users] Voicemail crashes asterisk

2005-08-17 Thread Hall, Eric M.

Thanks I will update via CVS tonight!

Thanks again! 




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Wednesday, August 17, 2005 11:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Voicemail crashes asterisk

It was fixed a while ago, download new code. There is a bug in the
tracker on it.

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users- 
> [EMAIL PROTECTED] On Behalf Of Hall, Eric M.
> Sent: Wednesday, August 17, 2005 9:23 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Voicemail crashes asterisk
> 
> When a user dial voicemail and just hangs up or enters the wrong 
> password 3 times asterisk will crash.
> 
> We are using Cisco 7960G with SIP
> My asterisk is CVS-HEAD built on 2005-08-02 23:47:59 UTC
> 
> Any help would be great!!!
> 
> 
> Thanks
> 
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[Asterisk-Users] AGI SCRIPTS USING PERL NEED SOME KIND OF COMPILATION TO WORK WITH *

2005-08-17 Thread j_amorim

Hi all, 

Help needed: 

Does AGI  SCRIPTS USING PERL NEED's SOME KIND OF COMPILATION TO WORK WITH 
* 

This simple script is not working. 

What can I do to make this interact with *? 

#!/usr/bin/perl 
# 
# 

use Asterisk::AGI; 

$AGI = new Asterisk::AGI; 

my %input = $AGI->ReadParse(); 
my $tests = 0; 
my $pass = 0; 
my $fail = 0; 

#setup callback 
$AGI->setcallback(\&mycallback); 


print STDERR "AGI Environment Dump:\n"; 
foreach $i (sort keys %input) { 
print STDERR " -- $i = $input{$i}\n"; 
} 




Thank in advance. 

Jônatas Amorim 
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Re: [Asterisk-Users] X100P dial out problem

2005-08-17 Thread Piero Baudino

Hi Tzafrir,

thanks for your reply...
Here is what happens when I make the call:

pbx*CLI> zap show channels
   Chan Extension  Context Language   MusicOnHold
 pseudoincomingit
  1incomingit
-- Executing NoOp("SIP/6601-5d39", ""Call for "91234567") in new
stack
-- Executing Dial("SIP/6601-5d39", "Zap/1/91234567") in new stack
-- Called 1/91234567
-- Zap/1-1 answered SIP/6601-5d39
-- Hungup 'Zap/1-1'
  == Spawn extension (x-lite, 912334567, 2) exited non-zero on
'SIP/6601-5d39'
-- Unregistered SIP '6601'
pbx*CLI> exit

The Hangup happens when I hangup from XLITE.

Here is my conf:
/etc/asterisk/zapata.conf

[channels]
language=it
signalling=fxs_ks
context=incoming
channel=>1

/etc/asterisk/extensions.conf
[incoming]
exten => s,1,Dial(SIP/6601&SIP/6602&SIP/6603,20,tr)  ; corresponding
clients must be configured in sip.conf
exten => s,2,Playback(vm-goodbye)
exten => s,3,Hangup

[outgoing]
exten => _9X.,1,NoOp("Call for "${EXTEN})
exten => _9X.,2,Dial(Zap/1/${EXTEN})

[x-lite]  ; Note: SIP extensions are defined here as "66" followed by any
two digits
exten => _66XX,1,NoOp("Call for "${EXTEN})
exten => _66XX,2,Dial(SIP/${EXTEN})
exten => _66XX,3,Congestion
include => outgoing

/etc/asterisk/sip.conf
port=5060
context=default
srvlookup=yes
dtmfmode=inband
allow=aLaw
allow=uLaw
allow=gsm

[6601]
type=friend
secret=password
host=dynamic
;dtmfmode=rfc2833
context=x-lite
callerid="Piero" <6601>
allow=aLaw
allow=uLaw
allow=gsm

Thanks.
PieroB

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RE: [Asterisk-Users] DECT gateways

2005-08-17 Thread Sean Milheim \(iDREUS Corporation\)

I may be wrong but most WIFI phones have very short battery life.  I've seen
some around ~3 hours talk and ~30 hours standby.  However the DECT phones
I've seen and we sell have ~100 hour standby and ~8 hour talk time. 


Regards, 
 
Sean Milheim
iDREUS Corporation
(941) 739-0051 x1005


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yoann Le Bihan
Sent: Wednesday, August 17, 2005 2:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DECT gateways


2005/8/17, Michiel van Baak <[EMAIL PROTECTED]>:
> Is there any other solution like this out there that works with 
> asterisk ?

Why don't you use WiFi VoIP phones like ZyXEL 2000W (which is not such
expensive compared with Cisco ones...) ?

Best regards,

YLB.
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RE: [Asterisk-Users] gnugk and asterisk

2005-08-17 Thread Vedran Dakic

Well, that's what I've been trying to do whole day long. I can't seem to do
that. It would be only logical that both gnugk and asterisk have "user" with
"password" to "understand each other". But I haven't been able to put
everything in the working state. Any help is welcome.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ADEGOKE ARUNA
Sent: Wednesday, August 17, 2005 8:29 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] gnugk and asterisk

Have u got ur asterisk registered as gateway in gnugk?

goksie

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[Asterisk-Users] TDM400 on BT network in UK

2005-08-17 Thread Andi Strain









I’m considering purchasing a Digium TDM400 card
with FXO module to connect my asterisk box up to BT’s PSTN.  I’ve
searched Google and have found that in the past there have been problems with
this card detecting the calling party hangup.  I am interested to know if
this issue has now been resolved and if UK caller ID is now fully supported. 
Also if anybody using the card in the UK is aware of any other issues
that I should be aware of then I would be grateful if you could let me know.

 

Kind regards

 

Andi






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Re: [Asterisk-Users] DECT gateways

On 20:42, Wed 17 Aug 05, Yoann Le Bihan wrote:
> 2005/8/17, Michiel van Baak <[EMAIL PROTECTED]>:
> > Is there any other solution like this out there that works
> > with asterisk ?
> 
> Why don't you use WiFi VoIP phones like ZyXEL 2000W (which is not such
> expensive compared with Cisco ones...) ?
> 
> Best regards,
> 

There are no cisco phones involved. The kirk handsets will
be recognized by CCM as cisco phones, but they are not.
That's what their website said. I have chan-sccp-cm running
with some 7905's and it does the job real fine. So if those
handsets work ok, it will do the trick for our customers.

I've been thinking about WIFI phones, but some important,
possible issues come to mind.

People want wireless phones so they can walk around while
calling. So the phones should do accesspoint roaming while
in a call and not getting disconnected. Is that possible ?
Besides that, DECT has encryption, and everywhere I read
about WIFI phones, I hear they don't provide normal
callquality when WEP is enabled.

So that's why I was looking for a dect solution.
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [Asterisk-Users] DECT gateways

2005-08-17 Thread Yoann Le Bihan

2005/8/17, Michiel van Baak <[EMAIL PROTECTED]>:
> Is there any other solution like this out there that works
> with asterisk ?

Why don't you use WiFi VoIP phones like ZyXEL 2000W (which is not such
expensive compared with Cisco ones...) ?

Best regards,

YLB.
[EMAIL PROTECTED]
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[Asterisk-Users] DECT gateways

Heya list,

I need some advice/experience.

Some of our customers are asking us about DECT solutions for
their asterisk install. Some others will not go to asterisk
if there won't be a DECT solution.
They now have a Siemens or a Samsung PBX. Those PBX-es come
with a DECT basestation and optionally repeaters etc.
All those basestations speak some own protocol to the PBX,
so we cannot use them with asterisk.

I been looking around on the internet and found the Kirk
gear. Anyone has any experience with them ? The website
states they are recognized as Cisco 7970 in CCM. Does
chan-sccp handel those Kirk emulated devices ?

Is there any other solution like this out there that works
with asterisk ?

Thanks for your input,

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [Asterisk-Users] Voicemail file permissions

2005-08-17 Thread Matthew Boehm


Tzafrir Cohen wrote:

On Wed, Aug 17, 2005 at 07:48:29AM -0400, hugolivude wrote:


Is there a way around this w/o giving everyone root privileges!


Run asterisk as its own user/group. We do.

-Matthew

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RE: [Asterisk-Users] gnugk and asterisk

2005-08-17 Thread ADEGOKE ARUNA

Have u got ur asterisk registered as gateway in gnugk?

goksie

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vedran Dakic
Sent: Wednesday, August 17, 2005 3:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] gnugk and asterisk

Man, I would really be grateful if you could put me out of my misery and
send me something, I don't know where's anything anymore in the config files
or anything. Too much editing those in the past 16 hours, I guess..

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Penton
Sent: Wednesday, August 17, 2005 4:19 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] gnugk and asterisk

Hey Vedran 

I did this a while ago but to put you on the write track you have to
register your gatekeeper (gnugk) with Asterisk as a gateway specifying a
prefix, let's say for arguments sake '0'. Then any numbers dialled on your
GK-managed H.323 network, that start with a zero, are routed to the gateway
(in this case asterisk)
If you still have problems I may be able to dig up some configs for you??

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RE: [Asterisk-Users] problems with eyebeam - video phone

2005-08-17 Thread Carlos Alperin

No, it tells you what codecs are enabled, and the traffic that you 're
delivering.

However you cannot complete a sip call, due the phone is not registered in
Asterisk

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jimmy Smith
Sent: Wednesday, August 17, 2005 11:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] problems with eyebeam - video phone

quickly this looks like a incompatible codec.. or unrecognized..

show codecs on CLI>

show show 
 262144 (1 << 18)  (0x4)  videoh261   (H.261 Video)
 524288 (1 << 19)  (0x8)  videoh263   (H.263 Video)
1048576 (1 << 20) (0x10)  video   h263p   (H.263+ Video)
does it ?

On 8/17/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Thank you for your answer.
> I didn't register on the domain of the Eyebeam software, actually I don't
> understand how to do that!
> I bouught 5 eyebeam activation keys and I am trying with the first 2 of
> them
> 
> On the Eyebeam side (both eyebeam), I only enabled the "Basic H.263"
codec,
> no other.
> 
> If, on the asterisk side in sip.conf, I put the gsm codec BEFORE h263, the
> two video phone speak without any problem (but without any video)
> If, on the asterisk side in sip.conf, I put the gsm codec AFTER h263, the
> first video phone call the second, the second answer and immediately
> the call ends.
> 
> If Ilook at /var/log/asterisk/full, I see:
> 
> Aug 17 08:37:06 VERBOSE[14731]: -- AGI Script dialparties.agi
> completed, returning 0
> Aug 17 08:37:06 VERBOSE[14731]: -- Executing Dial("SIP/551-eac0",
> "SIP/552|25|tr") in new stack
> Aug 17 08:37:06 DEBUG[14731]: SIMPLE DIAL (NO URL)
> Aug 17 08:37:06 DEBUG[14731]: Setting NAT on RTP to 0
> Aug 17 08:37:06 DEBUG[14731]: Setting NAT on VRTP to 0
> Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x8 formats
> Aug 17 08:37:06 DEBUG[14731]: Outgoing Call for 552
> Aug 17 08:37:06 DEBUG[14731]: Call from user '552' is 1 out of 0
> Aug 17 08:37:06 VERBOSE[14731]: -- Called 552
> Aug 17 08:37:06 DEBUG[13529]: (Provisional) Stopping retransmission (but
> retaining packet) on '[EMAIL PROTECTED]'
> Request 102: Found
> Aug 17 08:37:06 VERBOSE[14731]: -- SIP/552-ff46 is ringing
> Aug 17 08:37:10 DEBUG[13529]: Acked pending invite 102
> Aug 17 08:37:10 DEBUG[13529]: Stopping retransmission on
> '[EMAIL PROTECTED]' of Request 102: Found
> Aug 17 08:37:10 DEBUG[13529]: build_route: Contact hop:
> 
> Aug 17 08:37:10 VERBOSE[14731]: -- SIP/552-ff46 answered SIP/551-eac0
> Aug 17 08:37:10 WARNING[14731]: No path to translate from SIP/551-eac0(2)
> to SIP/552-ff46(524288)
> Aug 17 08:37:10 WARNING[14731]: Had to drop call because I couldn't make
> SIP/551-eac0 compatible with SIP/552-ff46
> Aug 17 08:37:10 DEBUG[14731]: update_user_counter(552) - decrement outUse
> counter
> 
> 
> It seems the problem documented in bug
> http://bugs.digium.com/bug_view_page.php?bug_id=0003709
> but actually it is not exactly the same.
> 
> moreover: is there any way to put the patch described in
> http://bugs.digium.com/bug_view_page.php?bug_id=0003709 (enable H263p in
*)
> in asterisk 1.0.9 and not asterisk CVS HEAD ?
> 
> Any help will be greatly appreciated.
> 
> Andrea
> 
> 
> 
> 
> "Carlos Alperin"
> <[EMAIL PROTECTED]
> om.net>To
> Sent by:  "'Asterisk Users Mailing List -
> asterisk-users-bo Non-Commercial Discussion'"
> [EMAIL PROTECTED] 
> m.com  cc
> 
>   Subject
> 16/08/2005 20.48  RE: [Asterisk-Users] problems with
>   eyebeam - video phone
> 
> Please respond to
>  Asterisk Users
>  Mailing List -
>  Non-Commercial
>Discussion
> <[EMAIL PROTECTED]
> ists.digium.com>
> 
> 
> 
> 
> 
> 
> Hi,
> 
> I get Eyebeam working with an older version of Asterisk 1.0.2(I believe).
I
> only use H.263 and SIP. (G.729)
> 
> Now, the more important question is if you register on the domain on the
> Eyebeam software. I found that this was the full secret about this.
> 
> Let me know your configuration on the Eyebeam side.
> 
> Regards,
> 
> Carlos Alperin
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Tuesday, August 16, 2005 11:28 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] problems with eyebeam - video phone
> 
> I am trying to connect two Xten eyeBeam Video Phone
> 
> No problems in voice connecting.
> 
> I tryed to modify my sip.conf
> 
> [general]
> language=it
> videosupport=yes
> ; enable Asterisk video support
> 
> port = 5060

RE: [Asterisk-Users] problems with eyebeam - video phone

2005-08-17 Thread Carlos Alperin

As I said in my previous mail, If you don't put the registration domain (IP
address of your Asterisk Server) your phones never are going to be registers
on Asterisk.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jimmy Smith
Sent: Wednesday, August 17, 2005 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] problems with eyebeam - video phone

Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x8 formats

is from what you pasted btw..

Don't know any of 0x8 formats 

is
524288 (1 << 19)  (0x8)  videoh263   (H.263 Video)

meaning it downst understand it or find it



On 8/17/05, Jimmy Smith <[EMAIL PROTECTED]> wrote:
> quickly this looks like a incompatible codec.. or unrecognized..
> 
> show codecs on CLI>
> 
> show show
>  262144 (1 << 18)  (0x4)  videoh261   (H.261 Video)
> 524288 (1 << 19)  (0x8)  videoh263   (H.263 Video)
>1048576 (1 << 20) (0x10)  video   h263p   (H.263+ Video)
> does it ?
> 
> On 8/17/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > Thank you for your answer.
> > I didn't register on the domain of the Eyebeam software, actually I
don't
> > understand how to do that!
> > I bouught 5 eyebeam activation keys and I am trying with the first 2 of
> > them
> >
> > On the Eyebeam side (both eyebeam), I only enabled the "Basic H.263"
codec,
> > no other.
> >
> > If, on the asterisk side in sip.conf, I put the gsm codec BEFORE h263,
the
> > two video phone speak without any problem (but without any video)
> > If, on the asterisk side in sip.conf, I put the gsm codec AFTER h263,
the
> > first video phone call the second, the second answer and immediately
> > the call ends.
> >
> > If Ilook at /var/log/asterisk/full, I see:
> > 
> > Aug 17 08:37:06 VERBOSE[14731]: -- AGI Script dialparties.agi
> > completed, returning 0
> > Aug 17 08:37:06 VERBOSE[14731]: -- Executing Dial("SIP/551-eac0",
> > "SIP/552|25|tr") in new stack
> > Aug 17 08:37:06 DEBUG[14731]: SIMPLE DIAL (NO URL)
> > Aug 17 08:37:06 DEBUG[14731]: Setting NAT on RTP to 0
> > Aug 17 08:37:06 DEBUG[14731]: Setting NAT on VRTP to 0
> > Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x8 formats
> > Aug 17 08:37:06 DEBUG[14731]: Outgoing Call for 552
> > Aug 17 08:37:06 DEBUG[14731]: Call from user '552' is 1 out of 0
> > Aug 17 08:37:06 VERBOSE[14731]: -- Called 552
> > Aug 17 08:37:06 DEBUG[13529]: (Provisional) Stopping retransmission (but
> > retaining packet) on '[EMAIL PROTECTED]'
> > Request 102: Found
> > Aug 17 08:37:06 VERBOSE[14731]: -- SIP/552-ff46 is ringing
> > Aug 17 08:37:10 DEBUG[13529]: Acked pending invite 102
> > Aug 17 08:37:10 DEBUG[13529]: Stopping retransmission on
> > '[EMAIL PROTECTED]' of Request 102: Found
> > Aug 17 08:37:10 DEBUG[13529]: build_route: Contact hop:
> > 
> > Aug 17 08:37:10 VERBOSE[14731]: -- SIP/552-ff46 answered
SIP/551-eac0
> > Aug 17 08:37:10 WARNING[14731]: No path to translate from
SIP/551-eac0(2)
> > to SIP/552-ff46(524288)
> > Aug 17 08:37:10 WARNING[14731]: Had to drop call because I couldn't make
> > SIP/551-eac0 compatible with SIP/552-ff46
> > Aug 17 08:37:10 DEBUG[14731]: update_user_counter(552) - decrement
outUse
> > counter
> >
> >
> > It seems the problem documented in bug
> > http://bugs.digium.com/bug_view_page.php?bug_id=0003709
> > but actually it is not exactly the same.
> >
> > moreover: is there any way to put the patch described in
> > http://bugs.digium.com/bug_view_page.php?bug_id=0003709 (enable H263p in
*)
> > in asterisk 1.0.9 and not asterisk CVS HEAD ?
> >
> > Any help will be greatly appreciated.
> >
> > Andrea
> >
> >
> >
> >
> > "Carlos Alperin"
> > <[EMAIL PROTECTED]
> > om.net>
To
> > Sent by:  "'Asterisk Users Mailing List -
> > asterisk-users-bo Non-Commercial Discussion'"
> > [EMAIL PROTECTED] 
> > m.com
cc
> >
> >
Subject
> > 16/08/2005 20.48  RE: [Asterisk-Users] problems with
> >   eyebeam - video phone
> >
> > Please respond to
> >  Asterisk Users
> >  Mailing List -
> >  Non-Commercial
> >Discussion
> > <[EMAIL PROTECTED]
> > ists.digium.com>
> >
> >
> >
> >
> >
> >
> > Hi,
> >
> > I get Eyebeam working with an older version of Asterisk 1.0.2(I
believe). I
> > only use H.263 and SIP. (G.729)
> >
> > Now, the more important question is if you register on the domain on the
> > Eyebeam software. I found that this was the full secret about this.
> >
> > Let me know your configuration on the Eyebeam side.
> >
> > Regards,
> >
> > Carlos Alperin
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > [EMAIL PROTECTED]
> > Sent: Tuesday, August 16, 2005 11:28 AM
> > To: asteris

Re: [Asterisk-Users] X100P dial out problem

On Wed, Aug 17, 2005 at 02:20:17PM +0200, Piero Baudino wrote:
> Hi all!
> 
> I'm new to asterisk and I'm trying a simple config with:
> - Debian GNU/Linux (unstable)
> - last version of Asterisk
> - a X100P card
> 
> I have a problem with dial out from a SIP software phone (XLITE) to a
> public number (ex. my mobile phone), asterisk start the call, but nothing
> happen...

CLI trace?

> If I run "ztmonitor 1" I can see the right RX level and if I try to make a
> call with an analog standard phone connected to the second plug of the
> X100P, I can see the RX level going UP and down normally, and I can also
> hear my voice during a call.
> Otherwise, when I try to dial out from XLITE, when I start the call the
> RX level go to 0 and I can only hear the numbers of che called number but
> I can hear nothing on RX and the line is "locked" until I remove the
> wcfxo kernel module; in Italy we must wait for a tone before starting the
> call
> 
> Is there anyone here with an idea for my problem ?

asterisk -rvvv

zap show channels

and report what you see, and what happens when you try to call.

Have you configured an asterisk zap channel for the phone? Are you
calling through it? 

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] Iaxy Distinctive Ring

2005-08-17 Thread Yoann Le Bihan

2005/8/17, Andrès Tello Abrego <[EMAIL PROTECTED]>:
> A iaxy, is a CPE device that provides VOIP capabilities to normal phones,
> using the iax protocol...
> 
> So is a little hardware, for telephony usages, which doesn't have a lot of
> features, and is't so cheap...

Oooh... right... it's like a ATA but using IAX protocol ?
Thanks a lot ! :)

Best regards,

YLB.
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Automatic start with SuSe linux

On Wed, Aug 17, 2005 at 01:27:08PM +0300, [EMAIL PROTECTED] wrote:
> Hi!
> I'm trying to start asterisk at boottime. Since SuSe 

It was SuSE (the old way). Now it is SUSE. Was it ever SuSe?

> has no rc.local like in
> Redhat linux, I need asterisk starting script to /etc/init.d/rc3.d -directory
> (I assume it is like that if i want automated asterisk startup).
> Do you have any experience how this is implemented in SuSe, and if you have 
> some
> useful script for starting asterisk, I would be very, i mean VERY pleased?
> 
> Thank you all in advance!

One nice thing SuSE has and most other distros lack is service
dependencies: you can define in your init.d script which services your
script needs and let insserv sert out the load order.

For instance, asterisk needs to load after zaptel. The flash operator
panel's daemon needs to start after asterisk. 

Also, if you install from RPMs, note that the init.d dir of SuSE is
actually different than the one od RH. Or at least it was last time I
looked.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
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Re: [Asterisk-Users] XORCOM RAPID Asterisk - Suggestions?

On Wed, Aug 17, 2005 at 01:36:51PM +0100, Sharadindu Mohanty wrote:
> Hey Guys,
>   Wanted a Suggestion..Howz this Xorcom Asterisk?

Great!

> I am using it and till now its fine as currently it is in testing stage 
> with 3-4 users.

I've been using it for quite a while and am very happy with it. Even
better than Debian! 

The fact the I've posted notices of its releases to this have nothing to
do with the tone of this message. I'm totally objective.

>  
> Any Ideas???

In a more serious tone, feedback would be appreciated: here, in the
Xorcom Rapid users list, or to me privately.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] XML Revisited

2005-08-17 Thread Anton Krall

Hello Guys.

I recently contacted polycoms tech support asking if their phones supported
XML pushed information to which they replied that only model 600 had a
microbrwoser capable of reading dhtml files and such.

My question to the community is: is somebody doing any XML info push to any
brand of phones except Cisco? How are you doing it?

One of the wonders of VoIP should be the means to send information back to
the phone which ould be displayed on those wonderful screens that they have
:) besides showing callerid and time which normal phones do.. 

Any ideas/comments?

Anton Krall

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[Asterisk-Users] FXS on TDM12B suddenly stopped working Properly

2005-08-17 Thread chawki hammoud

Hi:

I was running TDM12B. Both FXS and FXO were working
fine. Then all of the sudden FXS had problems. When I
pick-up the phone and dial any number, FXS doesn't
respond. I just keep hearing the normal signaling line
tone comming from the FXS. I changed the FXS module
and it had the same problem. I changed the the TDM
card and installed different FXS and nothing changed.

I appreciate any suggestions.









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Re: [Asterisk-Users] Does intel 865 board works fine with Asterisk

2005-08-17 Thread Angus Comber

I have one Asterisk system working with a Junghanns BRI card and another 
working with a Digium TDM card with an Intel D865 motherboard.


Angus



- Original Message - 
From: "jonny hashem" <[EMAIL PROTECTED]>

To: 
Sent: Wednesday, August 17, 2005 6:14 PM
Subject: [Asterisk-Users] Does intel 865 board works fine with Asterisk


Hi:

I would like to know what are the issues I need to
look for in a chipset board so I can make sure it
works fine with digium cards and Asterisk . Is intel
board 865 fits the description?

Regards;


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Re: [Asterisk-Users] Voicemail file permissions