[Asterisk-Users] Does Atcom AU-200 work with XLite?

[Melisa Teoh]

Would appreciate any learnings on the AU-200 model.

Thanks.

Melisa.
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RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

[Koopmann, Jan-Peter]

On Thursday, March 09, 2006 8:18 AM Douglas Garstang wrote: 

> By 'code for asterisk' are you referring to the Asterisk source code?
> If so, step back and think about your statement for a moment. If, for
> Asterisk to be enterprise class, it's source code needs to be
> modified from it's current content, it's hardly enterprise class, is
> it?

What is this thread all about? Is Asterisk "enterprise class"? The answer is
obvious: It depends on your definition of "enterprise class".

If you definition includes things like "RTP in/out traffic on multiple
interfaces must work" then the answer is no.

If you definition is somewhere along the line "can be used in most
enterprises without problems" then the answer is yes.


If you need a feature you at least have the possibility to code it yourself
(yes, source code). Avaya&Co give you the opportunity to hand in a feature
request but nothing more. Unless you pay for the feature they will probably
not implement it and you have no way of doing so yourself. 


Asterisk does not really meet my personal definition of "enterprise class"
but since there is no commonly accepted definition in the first place, why
trust Digiums words on the website at all? I usually do not trust any
marketing phrase like that no matter what the product is. Try Asterisk
yourself and if your decision is that you cannot use it, then don't! But
please stop getting on peoples nerves bashing on the term "enterprise
class". It will not get you or us anywhere.

If you are not satisfied with what Asterisk can achieve you have plenty of
other choices. Feel free to use them. Feel free to contribute to the
project. Constructive criticism is wanted (at least in most of the cases
this seems to be true even though there is room for improvement, agreed).
Currently you are not helping at all. Annoying is a term that comes to mind
though...


Kind regards,
  JP


smime.p7s
Description: S/MIME cryptographic signature
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Re: [Asterisk-Users] Any way to change dns timeout value? Asterisk hangs if internet unreachable

[Olle E Johansson]

The real solution is to implement asynchronus DNS. We are looking into
doing that with the C-ares library. No promises yet, it all depends on
funding for this development work.

If anyone is interested in funding it, please contact me off list.

/O




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Re: [Asterisk-Users] Real Time Asterisk

[Olle E Johansson]

8 mar 2006 kl. 17.07 skrev Fernando Lujan:


Hi guys,

I want to setup a environment where asterisk load all information  
from a Postgresql database. So here goes my questions:


1) Is real time asterisk  stable enough?
2) Where can I found documentation about using it with Postgresql?  
( including meet me conferences)




There is a realtime PostgreSQL driver for testing in the bug tracker,  
please test it!


http://bugs.digium.com/view.php?id=5637

Regards,
/Olle

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Re: [Asterisk-Users] parking slot lights - testers wanted

[Olle E Johansson]

8 mar 2006 kl. 15.19 skrev Dr. Michael J. Chudobiak:


Hi all,

The "metermaid" patch allows you to use the programmable buttons  
and LEDs on phones (like the Snom 2xx or 3xx) to view the status of  
parking slots and transfer to them. This should be really useful  
for small-office environments.


Anyway, the patch seems to work with Snom phones (and hopefully  
others) now! The curious are encouraged to download the "metermaid- 
v3.txt" patch against v1.2.4 for testing and feedback! See http:// 
bugs.digium.com/view.php?id=5779 for details.



BTW, this is included in test-this-branch and the real code you want  
to test is in

the metermaids-trunk branch.

...and it is my code ;-)

Thanks for testing!

/Olle

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RE: [Asterisk-Users] res_mysql.conf & DNS SRV lookup

[Douglas Garstang]

If you read the list, you will see that several people have noted the 
exceedingly long time for posts to appear in the list today.

-Original Message- 
From: Matt Riddell [NZ] [mailto:[EMAIL PROTECTED] 
Sent: Wed 3/8/2006 11:47 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: Re: [Asterisk-Users] res_mysql.conf & DNS SRV lookup



Douglas Garstang wrote:
> Good grief! I posted the message below at 1:17pm... and it appeared 
on the list after 8pm.
> Nice

I just posted mine and it arrived 30 seconds later...from New Zealand.
Maybe your mail servers are b0rk3n:

hehe

:D

It varies from time to time, but the mails do tend to go through!

--
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Matt Riddell
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RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

[Douglas Garstang]

By 'code for asterisk' are you referring to the Asterisk source code? If so, 
step back and think about your statement for a moment. If, for Asterisk to be 
enterprise class, it's source code needs to be modified from it's current 
content, it's hardly enterprise class, is it?
 
If 'code for asterisk' refers to extensions.conf and the like, I fail to see 
how anything within the asterisk dial plan would account for the apparent 
inability of asterisk to listen for RTP traffic on all network interfaces.
 
Doug.

-Original Message- 
From: Matt Riddell [NZ] [mailto:[EMAIL PROTECTED] 
Sent: Wed 3/8/2006 11:43 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: Re: [Asterisk-Users] Oh this is bad bindaddr and rtp 
traffic



Douglas Garstang wrote:
> Asterisk calls the Business Edition 'enterprise grade'. It's right 
there on the Digium website. It's the same dang code as the open source 
version, just older.

We are using it successfully in quite a few enterprise roll outs.  If
you are unable to, maybe you should attend one of our training sessions,
which among other things discuss how to code for Asterisk.

If however you'd rather just complain, please do so to /dev/null

--
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Matt Riddell
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Re: [Asterisk-Users] sending text to display of sip phones

[Olle E Johansson]

9 mar 2006 kl. 07.50 skrev Matt Riddell [NZ]:


Alejandro Vargas wrote:

I red that it is possible to send instant messages to the displays of
sip phones. How can I do it using Asterisk?


You can either do sendtext from an agi on that channel, or using my  
new

patch ( http://bugs.digium.com/view.php?id=6131 ), you can do it from
the manager interface just by passing the channel of a call.

...or use the sendtext() dialplan application.

The important fact about all of these is that they only work during a  
call,

not between calls to the phone.

/Olle


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Re: [Asterisk-Users] Multiple Subscriptions to SIP accounts at Same Domain

[Olle E Johansson]

THis is too hard to solve in Asterisk, even though it can be solved.  
I've answered
the question far too many times to answer again - search the mailing  
lists and the
wiki and you will find out how to work with peer matching to fix this  
issue.


In the "sipregister" development branch I am working with Luigi Rizzo  
to solve
this issue once and for all. In that code, we change the registration  
process so
you simply add "register=yes" in a peer section and we will match all  
incoming
calls to that peer. If you have multiple accounts with the same  
service provider,

simply create more peers and we will match each one properly.

This patch is part of the test-branch "test-this-branch" if you want  
to test it.

I need feedback from testers, so please do.

/Olle


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Re: [Asterisk-Users] Professional Recordings

[Matt Riddell [NZ]]

Waldo Rubinstein wrote:
> Can anyone recommend a company that does professional Asterisk
> recordings for things like IVR, greetings, MOH, announcements, etc?

http://www.digium.com/index.php?menu=product_category&category=thevoice

-- 
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Re: [Asterisk-Users] Re: PLEASE respond: how to get Asterisk to change coders on RTP handoff?? HELLO???

[Matt Riddell [NZ]]

Dan Miller wrote:
> So, when I get no comments on this at all, either here or on any of the 
> forums, does that mean nobody knows what I'm talking about??  Or does nobody 
> know the answer??  Or is it just a stupid question and nobody wants to bother 
> telling me where to look??
> 
> It *is* a question that I have to answer somehow; I've read all through TFOT 
> and see nothing relevant to this issue.  It's silly to spend $15000 on a G723 
> license just so I can play back menu messages from Asterisk (since the actual 
> call decoding is done by the external boxes, which have already paid the 
> licensing fees).

You can not really currently change codecs mid call (in most situations)
although work has been progressing in this area for some time.

Theoretically you should be able as others have based IAX devices around
this concept, but I don't think its available for sip.

Your other option would be to convert the audio files from GSM to G723.1
and that way, playing them would not require transcoding.

-- 
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Matt Riddell
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Re: [Asterisk-Users] What port mpg123 uses for MoH?

[Matt Riddell [NZ]]

Zach A wrote:
> Hi,
> 
> What port does mpg123 uses to play music on when it starts MoH after
> asterisk has put called on hold?

As far as I'm aware it writes to standard output and reads from standard
input (i.e. no ports involved)

-- 
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Re: [Asterisk-Users] parking slot lights - testers wanted

[Matt Riddell [NZ]]

Dr. Michael J. Chudobiak wrote:
> Hi all,
> 
> The "metermaid" patch allows you to use the programmable buttons and
> LEDs on phones (like the Snom 2xx or 3xx) to view the status of parking
> slots and transfer to them. This should be really useful for
> small-office environments.
> 
> Anyway, the patch seems to work with Snom phones (and hopefully others)
> now! The curious are encouraged to download the "metermaid-v3.txt" patch
> against v1.2.4 for testing and feedback! See
> http://bugs.digium.com/view.php?id=5779 for details.

Is this the same one in the test-this branch?

-- 
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Re: [Asterisk-Users] sending text to display of sip phones

[Matt Riddell [NZ]]

Alejandro Vargas wrote:
> I red that it is possible to send instant messages to the displays of
> sip phones. How can I do it using Asterisk?

You can either do sendtext from an agi on that channel, or using my new
patch ( http://bugs.digium.com/view.php?id=6131 ), you can do it from
the manager interface just by passing the channel of a call.

-- 
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Matt Riddell
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Re: [Asterisk-Users] res_mysql.conf & DNS SRV lookup

[Matt Riddell [NZ]]

Douglas Garstang wrote:
> Good grief! I posted the message below at 1:17pm... and it appeared on the 
> list after 8pm. 
> Nice

I just posted mine and it arrived 30 seconds later...from New Zealand.
Maybe your mail servers are b0rk3n:

hehe

:D

It varies from time to time, but the mails do tend to go through!

-- 
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Matt Riddell
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Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

[Matt Riddell [NZ]]

Douglas Garstang wrote:
> Asterisk calls the Business Edition 'enterprise grade'. It's right there on 
> the Digium website. It's the same dang code as the open source version, just 
> older. 

We are using it successfully in quite a few enterprise roll outs.  If
you are unable to, maybe you should attend one of our training sessions,
which among other things discuss how to code for Asterisk.

If however you'd rather just complain, please do so to /dev/null

-- 
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Matt Riddell
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Re: [Asterisk-Users] Memory Problems

[Joseph Tanner]

Something like:

up2date -i kernel-hugemem

Then make the appropriate changes in /etc/grub.conf, reboot, and see
if it works.  Of course, that's an overly simplified explanation, if
this is a production system please research this first.  If it's a
test system, well what's the worst that could happen?

Joseph Tanner

On 3/9/06, Dumpolid Exeplish <[EMAIL PROTECTED]> wrote:
> So how do I enable a High mem Kernel? Do i have to recomplile the kernel to
> use highmem ??
>
>
> On 3/9/06, Joseph Tanner < [EMAIL PROTECTED]> wrote:
> > The answer's just below the part you bolded.  "Use a HIGHMEM enabled
> kernel."
> >
> > Joseph Tanner
> >
> > On 3/8/06, Dumpolid Exeplish <[EMAIL PROTECTED]> wrote:
> > > Hello,
> > > This is not a question directly related to asterisk.
> > > I am currently rinning ansterisk on  a Debian server and i just upgraded
> my
> > > memory from 1GB to 2GB. However, my linux OS does not recognise the
> memory
> > > upgrade. The BIOS does, but the Debian Linux refuses to use the entier
> > > memory, currently, it registered only 900MB.
> > > Can anyone tell me why thi is and a solution to this??
> > >
> > > My Debian version is "Linux asterisk 2.6.12.3 #1 Mon Aug 1 19:33:51 WAT
> 2005
> > > i686 GNU/Linux"
> > >
> > > The server is currently routing calls from SIP internal users through an
> E1
> > > card (TE410)
> > >
> > > OUTPUT FROM dmesg command
> > >
> > > 009dc00 (usable)
> > >  BIOS-e820: 0009dc00 - 000a (reserved)
> > >  BIOS-e820: 000f - 0010 (reserved)
> > >  BIOS-e820: 0010 - 7fee (usable)
> > >  BIOS-e820: 7fee - 7fee3000 (ACPI NVS)
> > >  BIOS-e820: 7fee3000 - 7fef (ACPI data)
> > >  BIOS-e820: 7fef - 7ff0 (reserved)
> > >  BIOS-e820: fec0 - 0001 (reserved)
> > > Warning only 896MB will be used.
> > > Use a HIGHMEM enabled kernel.
> > > 896MB LOWMEM available.
> > > found SMP MP-table at 000f5a20
> > > On node 0 totalpages: 229376
> > >   DMA zone: 4096 pages, LIFO batch:1
> > >   Normal zone: 225280 pages, LIFO batch:31
> > >   HighMem zone: 0 pages, LIFO batch:1
> > >
> > >
> -END
> > >
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Re: [Asterisk-Users] [Slightly OT] Does TE110P (a 32-bit PCI) fit into PCIe x8 slot?

[Josip Gracin]

Josip Gracin wrote:

Does TE110P (a 32-bit PCI) fit into PCI Express x8 slot?


It turned out that it doesn't.  Which leaves me with the question: does 
Digium produce PCI-express cards?


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RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe

[David Phelan]

Best of luck :-D
I would be interested in your progress on this.

I am having very little problem in convincing ppl to upgrade their multiple
BRI cricuits for a single pri.  The cost difference between a te110 (or a
Sangoma A101) MORE than covers the difference from the customer stand point,
especially once you are up to 3 ISDN-2 Interfaces.

Dave


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Harper
Sent: Thursday, 9 March 2006 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe

I have received the card.

It comes with some closed source capi drivers, which I haven't tried as I
don't believe that is in acceptable solution anyway.

I had a look at hacking qozap to make it work, but haven't gone there at the
moment. What I'm looking at now is visdn. 0.14 doesn't even want to compile
against 2.6.15, but the latest development snapshot does, and after I added
in the correct PCI ID's, it detects the card.

I have no idea if the development vISDN HFC-4S drivers are even in a
workable state, but they do detect L1 status, and asterisk is able to detect
an incoming call but won't answer it.
 
The card itself is the 'Saphir III ML PCI'. Older versions of it used
another chipset ('Infineon' I think), but this newer one definitely uses the
HFC-4S chipset, and is definitely detected as such by the vISDN driver.

The only supplier I have found in Australia for it is
http://www.voipnow.com.au/, and they are the ones who have supplied the one
I am testing. On their web site, the picture is of the old version with 4
large chips on it, but the new one is pictured at
http://hstnet.de/english/index.asp.

I'll follow up if I have any further success, or if I give up.

James



> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users- 
> [EMAIL PROTECTED] On Behalf Of David Hindmarsh
> Sent: Sunday, 5 March 2006 22:37
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one -
maybe
> 
> Hi James,
> 
> I am definitely interested in the card and also in the results of your 
> testing.
> 
> Regards,
> 
> David
> 
> 
> LEXNET PTY LTD
> [e] [EMAIL PROTECTED]
> [m] 0411 172 667
> Mail: PO Box R1180
> Royal Exchange, Sydney NSW 1225
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of James 
> > Harper
> > Sent: Saturday, 4 March 2006 12:03
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] MultiBRI in Australia - found one - maybe
> >
> > I may have found a source of an A-Ticked HFC 4BRI PCI adapter in 
> > Australia, and will be testing one next week if all goes well. I 
> > don't want to post the details of the reseller online unless invited 
> > to do so, so if nobody replies and says they are interested then I 
> > won't :)
> >
> > I'll follow up once I've tested it.
> >
> > Let me know if you want the details.
> >
> > James
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> >
> >
> > --
> > Internal Virus Database is out-of-date.
> > Checked by AVG Free Edition.
> > Version: 7.1.375 / Virus Database: 267.15.11/264 - Release
> > Date: 17/02/2006
> >
> >
> 
> --
> Internal Virus Database is out-of-date.
> Checked by AVG Free Edition.
> Version: 7.1.375 / Virus Database: 267.15.11/264 - Release Date:
> 17/02/2006
> 
> 
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Re: [Asterisk-Users] Memory Problems

[Dumpolid Exeplish]

So how do I enable a High mem Kernel? Do i have to recomplile the kernel to use highmem ??On 3/9/06, Joseph Tanner <
[EMAIL PROTECTED]> wrote:The answer's just below the part you bolded.  "Use a HIGHMEM enabled kernel."
Joseph TannerOn 3/8/06, Dumpolid Exeplish <[EMAIL PROTECTED]> wrote:> Hello,> This is not a question directly related to asterisk.> I am currently rinning ansterisk on  a Debian server and i just upgraded my
> memory from 1GB to 2GB. However, my linux OS does not recognise the memory> upgrade. The BIOS does, but the Debian Linux refuses to use the entier> memory, currently, it registered only 900MB.> Can anyone tell me why thi is and a solution to this??
>> My Debian version is "Linux asterisk 2.6.12.3 #1 Mon Aug 1 19:33:51 WAT 2005> i686 GNU/Linux">> The server is currently routing calls from SIP internal users through an E1
> card (TE410)>> OUTPUT FROM dmesg command>> 009dc00 (usable)>  BIOS-e820: 0009dc00 - 000a (reserved)>  BIOS-e820: 000f - 0010 (reserved)
>  BIOS-e820: 0010 - 7fee (usable)>  BIOS-e820: 7fee - 7fee3000 (ACPI NVS)>  BIOS-e820: 7fee3000 - 7fef (ACPI data)>  BIOS-e820: 7fef - 7ff0 (reserved)
>  BIOS-e820: fec0 - 0001 (reserved)> Warning only 896MB will be used.> Use a HIGHMEM enabled kernel.> 896MB LOWMEM available.> found SMP MP-table at 000f5a20
> On node 0 totalpages: 229376>   DMA zone: 4096 pages, LIFO batch:1>   Normal zone: 225280 pages, LIFO batch:31>   HighMem zone: 0 pages, LIFO batch:1>> -END
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[Asterisk-Users] re: Billing Package for Asterisk

[VIC IP Communications]

Hi,


Can anyone recommend a good billing package for use with Asterisk?
We would prefer something that has a Customer and Provider web
interface/access.

 

Thanks,

 

Bruce

VIC IP Communications

 






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Re: [Asterisk-Users] PAP2 won't make two g729 calls at the same time

[Tom Vile]

Actually its hardware related.

On 3/8/06, Nick Hoffman <[EMAIL PROTECTED]> wrote:
> On Thu March 9 2006 03:43, Warren Burstein <[EMAIL PROTECTED]> wrote:
> > I have a Linksys PAP2.  Identical setups for the two channels in both
> > the unit and in Asterisk.  In particular, both channels enable g729 and
> > set it as the preferred codec, and have disallow=all and allow=g729 in
> > sip.conf.
> >
> > If we make a call on one channel, it works (and uses g729), but if we
> > make a call on the other channel when the first one is still connected,
> > it fails.  We have three g729 licenses, and no others were in use at the
> > times this happened, but even if we didn't have enough, how would the
> > PAP2 know that?
>
>
> Hi Warren. On the PAP2, if you can make 2 simultaneous calls but only 1 can
> use G.729, I would hazard a guess that the PAP2 only has 1 G.729 license
> installed on it. I doubt that can be increased.
>
> Hope that helps.
> -- Nick
> e: [EMAIL PROTECTED]
> p: +61 7 5591 3588
> f: +61 7 5591 6588
>
> If you receive this email by mistake, please notify us and do not make any
> use of the email.  We do not waive any privilege, confidentiality or
> copyright associated with it.
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] PAP2 won't make two g729 calls at the same time

[Kristian Kielhofner]

Joseph Tanner wrote:

The PAP2 can only handle one g729 call at one time.  Whether that's a
hardware limitation, or licensing, or both, I don't know.

Joseph Tanner


Hardware.  The PAP2 (and SPA2000) can only do one g729 call at a time. 
Any other call will have to use g711.


--
Kristian Kielhofner
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Re: [Asterisk-Users] System Design

[Kristian Kielhofner]

Colin Anderson wrote:
This doesn't directly answer your question, because every integration 
scenario is different, but one of the nice things about Asterisk is that 
the barrier to entry to get a system working and play around with it is 
very low. What you might want to consider doing is get your Asterisk box 
working, minus the PRI card, get the Cisco phones running (you're going 
to buy them anyway) and put them in place, side by side with the current 
phones, and just have the everyone play with them. That alone will 
answer a lot of your questions about how to engineer it without 
commiting to a particular way.
 
However, in your remote office, I would ditch the crap router and at 
least use a Monowall http://m0n0.ch/wall/ because you can prioritize 
traffic with it, it's super easy to set up, and you can make it work 
with odds and ends you have laying around. If you have more than one 
static IP, you can even do a Monowall to Monowall VPN and leave your PIX 
in place, and then run the VoIP over the VPN. Monowall supports IPSec 
VPN's, so you can interface it with a lot of other firewalls out there, 
including Pix.
 
Running VoIP over a VPN is sometimes problematic (but sometimes it works 
great!) so again you can try it out without committing.
 
Sometimes it makes sense to have a remote Asterisk server at the other 
end and route calls via IAX, IAX is like a VoIP dream protocol, but on 
the other hand it adds complexity where complexity is undesirable. You 
should try it both ways: Stick in a remote Asterisk server on the other 
end, route calls via IAX, and also have some Cisco's register with your 
main Asterisk server over SIP (both with and without the VPN)
 
The Dell will probably be fine, compatibility issues with Digium TDM 
cards nonwithstanding (there are some - ask Digium when you buy) and in 
your case, overkill. I'm running a Netfinity Xeon 550 (yup, 550 
Mhz) with 2 Te110P cards right now supporting 180 users in 32 locations 
in a 50 mile radius. Looking at the console right now I have 36 of 46 
channels open to my PRI's, 50 mixed SIP and IAX calls, and top says 
about 16% with load average about .53. And I'm recording all the calls.
 
On my remote IAX sites (30), I have between 2 to 5 users that do SIP to 
a local IAX server, then IAX here to the main office and out the PRI. 
What's running on the remote servers? Frigging P-II 233's. That's all. 
The reason it works is because I am careful with codec selection so 
there's no transcoding. And the call quality is just fine, thank you.
 
Ask the boss for a couple of weeks to experiment, get the gear, and 
test. That will give you the optimum result, instead of my jackass opinion.
 
hth




	It's good to see people using "low-end" hardware with Asterisk. 
Running applications on Linux/FreeBSD/OpenBSD/whatever is always 
refreshing to me in the days of 3.2ghz desktops.  I don't know about all 
of you, but as the "computer guy" in my family, I am constantly asked by 
aunts, uncles, cousins, etc if the computer they bought will be fast 
enough for the internet, word, burning cds, etc.  When they show me the 
hardware specs I have to laugh to myself.  Why?  Lately it's been 3+ Ghz 
cpus and 1gb of RAM or more.  With 10K SATA drives.  Fast enough?  You 
have got to be kidding me!  I immediately think of what that machine 
would be capable of if it were used in a non I/O bound server 
application with Linux...


	I am still AMAZED at what a well configured Linux machine can do on low 
end hardware.  Not just with Asterisk, but with Apache, MySQL, whatever. 
 As far as Asterisk goes, I think it is a safe bet that most setups are 
overpowered to the point of ridiculous.  Want to know what you should 
buy for your office?  Find some old junker that can barely run Windows 
2000, install Linux and Asterisk and see what you can do.  Read up on 
how to optimize a few things and you should be set (reliability not 
withstanding).  If not, do the math and find out what you need to buy 
(or what else to re-use).


	From what I can remember, this is how Linux got a foothold back in the 
90's.  Daring admins would take a recyled Windows desktop and make a 
print server, file server, web server, etc.  Thanks to Asterisk, admins 
of the 21st century can make a revolutionary PBX/telephone 
appliance/phone switch/alarm clock/etc!


--
Kristian Kielhofner
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Re: [Asterisk-Users] PAP2 won't make two g729 calls at the same time

[Leo Ann Boon]

Warren Burstein wrote:

I have a Linksys PAP2.  Identical setups for the two channels in both 
the unit and in Asterisk.  In particular, both channels enable g729 
and set it as the preferred codec, and have disallow=all and 
allow=g729 in sip.conf.


If we make a call on one channel, it works (and uses g729), but if we 
make a call on the other channel when the first one is still 
connected, it fails.  We have three g729 licenses, and no others were 
in use at the times this happened, but even if we didn't have enough, 
how would the PAP2 know that?


It's a PAP2 feature. The PAP2 hardware is only capable of 1 (ONE) G.729 
call at any time. The limit also applies if you're doing conferencing on 
the PAP2.




Here's a good, and a bad INVITE message, from the log file with sip 
debug enabled.  Has anyone seen anything like this?


INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3e521aa
From: PAP 220 ;tag=6b66e68deef168b2o0
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PAP 220 
Expires: 240
User-Agent: Linksys/PAP2-3.1.3(LS)
Content-Length: 246
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 261305180 261305180 IN IP4 192.168.254.44
s=-
c=IN IP4 192.168.254.44
t=0 0
m=audio 16392 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3fda1e15
From: PAP 220 ;tag=b8b86be991749af5o0
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PAP 220 
Expires: 240
User-Agent: Linksys/PAP2-3.1.3(LS)
Content-Length: 267
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 261589835 261589835 IN IP4 192.168.254.44
s=-
c=IN IP4 192.168.254.44
t=0 0
m=audio 16400 RTP/AVP 0 8 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv




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[Asterisk-Users] Re: [asterisk-biz] Professional Recordings

[asterisk_help]

http://www.mikesullivan.com/
http://thevoice.digium.com/

On Wed, 8 Mar 2006, Waldo Rubinstein wrote:
Can anyone recommend a company that does professional Asterisk recordings for 
things like IVR, greetings, MOH, announcements, etc?

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Re: [Asterisk-Users] MWI, SER and asterisk

[Nick Hoffman]

On Thu March 9 2006 08:52, [EMAIL PROTECTED] wrote:
> Hello,
>
> You can use ser as an outbound sip proxy and asterisk
> as a register server .
>
> Your sip agents will get MWI, ...
>
> Harry


Hi guys. With that solution, remember that Asterisk can handle a fraction 
of the number of registrations that SER can handle. So if you need to be 
able to support like 200+ SIP registrations, they should really be 
registering with SER. At least, that's what I understand from what I've 
read. Please correct me if I'm wrong.
-- Nick
e: [EMAIL PROTECTED]
p: +61 7 5591 3588
f: +61 7 5591 6588

If you receive this email by mistake, please notify us and do not make any 
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RE: [Asterisk-Users] HW Echo Cancellers

[Darren Wright]

Forget the orion.lots of DTMF problemstech support is not
Terribly well spoken.

Look for ANY of the 257* series...

Just ebay for "t1 echo"

-D


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Wednesday, March 08, 2006 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] HW Echo Cancellers

> > Tellabs looks a little too up-scale for what I need :). $1k for a
> > single port orion unit might be worth considering for really
stubborn
> > installs though.
> >
>
> Why? they go for around $100.00 on eBay.

What goes for $100 on eBay?  Tellabs?  or Orion?  I can't find any
Orion equipment on eBay.  What model Tellabs am I looking for?
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RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe

[James Harper]

I have received the card.

It comes with some closed source capi drivers, which I haven't tried as
I don't believe that is in acceptable solution anyway.

I had a look at hacking qozap to make it work, but haven't gone there at
the moment. What I'm looking at now is visdn. 0.14 doesn't even want to
compile against 2.6.15, but the latest development snapshot does, and
after I added in the correct PCI ID's, it detects the card.

I have no idea if the development vISDN HFC-4S drivers are even in a
workable state, but they do detect L1 status, and asterisk is able to
detect an incoming call but won't answer it.
 
The card itself is the 'Saphir III ML PCI'. Older versions of it used
another chipset ('Infineon' I think), but this newer one definitely uses
the HFC-4S chipset, and is definitely detected as such by the vISDN
driver.

The only supplier I have found in Australia for it is
http://www.voipnow.com.au/, and they are the ones who have supplied the
one I am testing. On their web site, the picture is of the old version
with 4 large chips on it, but the new one is pictured at
http://hstnet.de/english/index.asp.

I'll follow up if I have any further success, or if I give up.

James



> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of David Hindmarsh
> Sent: Sunday, 5 March 2006 22:37
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one -
maybe
> 
> Hi James,
> 
> I am definitely interested in the card and also in the results of your
> testing.
> 
> Regards,
> 
> David
> 
> 
> LEXNET PTY LTD
> [e] [EMAIL PROTECTED]
> [m] 0411 172 667
> Mail: PO Box R1180
> Royal Exchange, Sydney NSW 1225
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > James Harper
> > Sent: Saturday, 4 March 2006 12:03
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] MultiBRI in Australia - found one - maybe
> >
> > I may have found a source of an A-Ticked HFC 4BRI PCI adapter
> > in Australia, and will be testing one next week if all goes
> > well. I don't want to post the details of the reseller online
> > unless invited to do so, so if nobody replies and says they
> > are interested then I won't :)
> >
> > I'll follow up once I've tested it.
> >
> > Let me know if you want the details.
> >
> > James
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> > Checked by AVG Free Edition.
> > Version: 7.1.375 / Virus Database: 267.15.11/264 - Release
> > Date: 17/02/2006
> >
> >
> 
> --
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> Checked by AVG Free Edition.
> Version: 7.1.375 / Virus Database: 267.15.11/264 - Release Date:
> 17/02/2006
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RE: [Asterisk-Users] Dial command

[Alexander Lopez]

Exten => 222,1,Dial(SIP/polycom601||20)
Exten => 222,2,Dail(Zap/2/ww09123456789#

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Ronald Wiplinger
> Sent: Wednesday, March 08, 2006 5:26 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Dial command
> 
> I have an ZAP extension number 222 which is connected instead 
> to a phone to a FXS/FXO converter and from there to a CDMA gateway.
> 
> To dial my mobile phone I use:
> 
> 222 (wait 2 seconds) 09123456789
> 
> I cannot figure out how to write this into the dialplan as a 
> default number!
> 
> 222 as above I will use for dialing any other number, but I 
> want to add this phone as an extension which rings if 601 is 
> not picking up within 20 seconds.
> 
> How to write this?
> 
> 
> Some parts of my existing dial plan:
> [Globals]
> PHONE_222=ZAP/2r1; transfer to mobile phone <===
> hier I want to add the mobile phone number
> 
> [incoming]
> ...
> exten =>
> s,7,Dial(${PHONE_601}&${PHONE_621}&${PHONE_603}&${PHONE_610},3
> 0,tr)  ; ring phone_601, 621 & 603 for 30 seconds exten => 
> s,8,Dial(${PHONE_222},30,tr)  ; ring phone_222
> 
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[Asterisk-Users] Openline4 and [EMAIL PROTECTED]

[Chuck Fletcher]

Hi all,

Can somebody help me get my OpenLine4 card running with [EMAIL PROTECTED]

I've got my VPB drivers configured, but can't figure out how to map 
trunks and channels the typical way in the AMP config interface for 
[EMAIL PROTECTED]


Apparently I'm supposed to use /vpb/1 type commands, but I'm not sure 
how it should work.


I'm relatively new to asterisk, I did a bunch of stuff with Bayonne a 
few years ago.


I was successful using [EMAIL PROTECTED] to setup a sip based setup thanks 
to the tutorial at nerd vittles.


Anyway, can somebody give me the high level concept and a few examples 
of how to get my openline4 card rocking with [EMAIL PROTECTED]


Thanks,

Chuck
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Re: [Asterisk-Users] Is everyone getting mails except me?

[Darrick Hartman]

Ron McCarthy wrote:
I havent got any mails since 2:42 this morning..usually i get at least 
the normal 10-15 a hour, if someone gets this can they reply?


About once a week for the past three weeks I've experienced periods of 
time where no mail is received from the Asterisk mailing list.  After 
the delay, a bunch of delayed messages are received.  I don't know what 
is wrong or why this continues to happen (and NO I do not have gmail). 
During this same period of time, I continue to receive all other mail 
without problems.  I operate my own mail server and have watched the 
logs.  Nothing wrong on this side.  There does appear to be something 
wrong that someone at Digium should look into deeper.


Darrick
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
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Re: [Asterisk-Users] Reverse group in zapata.conf

[Time Bandit]

> I have a situation where I have 8 lines from the phone company in a hunt
> group coming in to my asterisk box.  These are the same lines I'm using
> for outgoing calls ( named g0 ).



> Is this possible?  If it isn't, I plan to reverse the order in which the
> lines are connected to my * box, having the same effect ( with no
> configuration changes.  :) ).  Anybody have any advice why I shouldn't
> do this either?  Any other suggestions?

Reversing the order would do the job, but you can also use G0 instead of g0.

from http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels

Dialing a Group

*  g: select the lowest-numbered non-busy Zap channel (aka.
ascending sequential hunt group).
* G: select the highest-numbered non-busy Zap channel (aka.
descending sequential hunt group).
* r: use a round-robin search, starting at the next highest
channel than last time (aka. ascending rotary hunt group).
* R: use a round-robin search, starting at the next lowest channel
than last time (aka. descending rotary hunt group).


> Thanks
You're welcome
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Re: [Asterisk-Users] HW Echo Cancellers

[Doug Lytle]

Matt wrote:

Tellabs looks a little too up-scale for what I need :). $1k for a
single port orion unit might be worth considering for really stubborn
installs though.

  

Why? they go for around $100.00 on eBay.



What goes for $100 on eBay?  Tellabs?  or Orion?  I can't find any
Orion equipment on eBay.  What model Tellabs am I looking for?
___
  

http://cgi.ebay.com/Tellabs-2572-64ms-T1-echo-canceller_W0QQitemZ5863816619QQcategoryZ51279QQssPageNameZWDVWQQrdZ1QQcmdZViewItem

Doug

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RE: [Asterisk-Users] No DTMF

[Mark Edwards]

Try dtmfmode=info and see if that works.

 

Mark

 

-Original Message-
From: Dovid Bender
[mailto:[EMAIL PROTECTED] 
Sent: Thursday, 9 March 2006 6:08
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] No DTMF

 



Some one was on my server
making changes to my sip.conf files. I am now having trouble with DTMF. No
matter what I use (inband,auto,rfc2833) the dtmf tones seem to not come thru. I
compared it to the wiki and all the configs seem to be in order.





 





Here is my sip.conf





 





[general]
disallow=all
;allow=g729 ; requires license for g729
allow=ulaw
port = 5060
nat=yes
context=from-sip
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
maxexpirey=4800 ; Maximum expiration for registrations
defaultexpirey=1800 ; Default expiration for registrations
canreinvite=no ; Allow clients to directly connect if set to yes. Set to no if
behind NAT.
tos=reliability
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
videosupport=no ; Turn on support for SIP video
dtmfmode=rfc2833 ;rfc2833 ;inband ;rfc2833 ; DTMF inband need to be set here.
pedantic=no
externip=..XXX





;Sip Media
register => XX:[EMAIL PROTECTED]/7322761368





[sipmedia6]
type=friend
user=XX ;(Phone Number)
username=XX ;(Phone Number)
fromuser=XX ;(Phone Number)
authname=XX ;(Phone Number)
secret= ;(SIP Password)
host=sip.sipmedia.com

disallow=all
allow=ulaw
context=ServerHighway
realm=sip1.xchangetele.com
fromdomain=sip.sipmedia.com
dtmfmode=rfc2833
canreinvite=no 
insecure=very





 





Here is my
extensions.conf





[general]
static=yes
writeprotect=yes





[ServerHighway]
;Play Server Highway IVR






Exten => s,1,Background(server-highway-ivr)
Exten => s,2,Background(blank-file-10)





Exten => 1,1,Ringing()
Exten => 1,2,Wait(15)
Exten => 1,3,Macro(stdexten,9511,9511)
Exten => 2,1,Ringing()
Exten => 2,2,Wait(15)
Exten => 2,3,Macro(stdexten,9512,9512)
Exten => 3,1,Ringing()
Exten => 3,2,Wait(15)
Exten => 3,3,Macro(stdexten,9513,9513)
Exten => 4,1,Ringing()
Exten => 4,2,Wait(15)
Exten => 4,3,Macro(stdexten,9514,9514)
Exten => i,1,Background(invalid)
Exten => i,2,Goto(s,1)





Exten => t,1,Goto(s,1)





exten => 9,1,Goto(s,1)
;Extension To Record Main IVR Message
exten => 500,1,Authenticate(XXX)
exten => 500,2,Record(ServerHighwayIvr:gsm)









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Bring photos to life! New
PhotoMail makes sharing a breeze. 






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Re: [Asterisk-Users] PAP2 won't make two g729 calls at the same time

[Nick Hoffman]

On Thu March 9 2006 03:43, Warren Burstein <[EMAIL PROTECTED]> wrote:
> I have a Linksys PAP2.  Identical setups for the two channels in both
> the unit and in Asterisk.  In particular, both channels enable g729 and
> set it as the preferred codec, and have disallow=all and allow=g729 in
> sip.conf.
>
> If we make a call on one channel, it works (and uses g729), but if we
> make a call on the other channel when the first one is still connected,
> it fails.  We have three g729 licenses, and no others were in use at the
> times this happened, but even if we didn't have enough, how would the
> PAP2 know that?


Hi Warren. On the PAP2, if you can make 2 simultaneous calls but only 1 can 
use G.729, I would hazard a guess that the PAP2 only has 1 G.729 license 
installed on it. I doubt that can be increased.

Hope that helps.
-- Nick
e: [EMAIL PROTECTED]
p: +61 7 5591 3588
f: +61 7 5591 6588

If you receive this email by mistake, please notify us and do not make any 
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Re: [Asterisk-Users] Memory Problems

[Joseph Tanner]

The answer's just below the part you bolded.  "Use a HIGHMEM enabled kernel."

Joseph Tanner

On 3/8/06, Dumpolid Exeplish <[EMAIL PROTECTED]> wrote:
> Hello,
> This is not a question directly related to asterisk.
> I am currently rinning ansterisk on  a Debian server and i just upgraded my
> memory from 1GB to 2GB. However, my linux OS does not recognise the memory
> upgrade. The BIOS does, but the Debian Linux refuses to use the entier
> memory, currently, it registered only 900MB.
> Can anyone tell me why thi is and a solution to this??
>
> My Debian version is "Linux asterisk 2.6.12.3 #1 Mon Aug 1 19:33:51 WAT 2005
> i686 GNU/Linux"
>
> The server is currently routing calls from SIP internal users through an E1
> card (TE410)
>
> OUTPUT FROM dmesg command
>
> 009dc00 (usable)
>  BIOS-e820: 0009dc00 - 000a (reserved)
>  BIOS-e820: 000f - 0010 (reserved)
>  BIOS-e820: 0010 - 7fee (usable)
>  BIOS-e820: 7fee - 7fee3000 (ACPI NVS)
>  BIOS-e820: 7fee3000 - 7fef (ACPI data)
>  BIOS-e820: 7fef - 7ff0 (reserved)
>  BIOS-e820: fec0 - 0001 (reserved)
> Warning only 896MB will be used.
> Use a HIGHMEM enabled kernel.
> 896MB LOWMEM available.
> found SMP MP-table at 000f5a20
> On node 0 totalpages: 229376
>   DMA zone: 4096 pages, LIFO batch:1
>   Normal zone: 225280 pages, LIFO batch:31
>   HighMem zone: 0 pages, LIFO batch:1
>
> -END
>
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Re: [Asterisk-Users] PAP2 won't make two g729 calls at the same time

[Tom Vile]

This ATA can only do 1 g729 call at a time.  The sipura 2002 is the
same way.  It's outlined in the datasheet.

On 3/8/06, Warren Burstein <[EMAIL PROTECTED]> wrote:
> I have a Linksys PAP2.  Identical setups for the two channels in both
> the unit and in Asterisk.  In particular, both channels enable g729 and
> set it as the preferred codec, and have disallow=all and allow=g729 in
> sip.conf.
>
> If we make a call on one channel, it works (and uses g729), but if we
> make a call on the other channel when the first one is still connected,
> it fails.  We have three g729 licenses, and no others were in use at the
> times this happened, but even if we didn't have enough, how would the
> PAP2 know that?
>
> Here's a good, and a bad INVITE message, from the log file with sip
> debug enabled.  Has anyone seen anything like this?
>
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3e521aa
> From: PAP 220 ;tag=6b66e68deef168b2o0
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: PAP 220 
> Expires: 240
> User-Agent: Linksys/PAP2-3.1.3(LS)
> Content-Length: 246
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> Content-Type: application/sdp
>
> v=0
> o=- 261305180 261305180 IN IP4 192.168.254.44
> s=-
> c=IN IP4 192.168.254.44
> t=0 0
> m=audio 16392 RTP/AVP 18 100 101
> a=rtpmap:18 G729a/8000
> a=rtpmap:100 NSE/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
>
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3fda1e15
> From: PAP 220 ;tag=b8b86be991749af5o0
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: PAP 220 
> Expires: 240
> User-Agent: Linksys/PAP2-3.1.3(LS)
> Content-Length: 267
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> Content-Type: application/sdp
>
> v=0
> o=- 261589835 261589835 IN IP4 192.168.254.44
> s=-
> c=IN IP4 192.168.254.44
> t=0 0
> m=audio 16400 RTP/AVP 0 8 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:100 NSE/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
>
>
>
>
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] Upgrading Asterisk witk G729 license installed

[Mojo with Horan & Company, LLC]

I would think that it would be OK to upgrade, but to be sure, your old 
license file should exist at 
/var/lib/asterisk/licenses/G729-.lic and could be backed up from 
there.  After the install, copy this back in.  And make sure you still 
have your codec_g729.so file to put in the modules directory.


Moj

Álvaro Palma wrote:
I've an Asterisk 1.2.4 installation, where I've also installed the G729 
codec license. I'd like to upgrade that installation to 1.2.5, but I'm 
not sure if I'll lost the license in the process (and if I'll be able to 
recover it later!!!).


Is there any special consideration I've to keep in mind in this case, or 
should I just run the typical "make + make install" and it will take 
care of keeping the license information?


Thanks a lot for your attention.



--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] PAP2 won't make two g729 calls at the same time

[Joseph Tanner]

The PAP2 can only handle one g729 call at one time.  Whether that's a
hardware limitation, or licensing, or both, I don't know.

Joseph Tanner

On 3/8/06, Warren Burstein <[EMAIL PROTECTED]> wrote:
> I have a Linksys PAP2.  Identical setups for the two channels in both
> the unit and in Asterisk.  In particular, both channels enable g729 and
> set it as the preferred codec, and have disallow=all and allow=g729 in
> sip.conf.
>
> If we make a call on one channel, it works (and uses g729), but if we
> make a call on the other channel when the first one is still connected,
> it fails.  We have three g729 licenses, and no others were in use at the
> times this happened, but even if we didn't have enough, how would the
> PAP2 know that?
>
> Here's a good, and a bad INVITE message, from the log file with sip
> debug enabled.  Has anyone seen anything like this?
>
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3e521aa
> From: PAP 220 ;tag=6b66e68deef168b2o0
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: PAP 220 
> Expires: 240
> User-Agent: Linksys/PAP2-3.1.3(LS)
> Content-Length: 246
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> Content-Type: application/sdp
>
> v=0
> o=- 261305180 261305180 IN IP4 192.168.254.44
> s=-
> c=IN IP4 192.168.254.44
> t=0 0
> m=audio 16392 RTP/AVP 18 100 101
> a=rtpmap:18 G729a/8000
> a=rtpmap:100 NSE/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
>
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3fda1e15
> From: PAP 220 ;tag=b8b86be991749af5o0
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: PAP 220 
> Expires: 240
> User-Agent: Linksys/PAP2-3.1.3(LS)
> Content-Length: 267
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> Content-Type: application/sdp
>
> v=0
> o=- 261589835 261589835 IN IP4 192.168.254.44
> s=-
> c=IN IP4 192.168.254.44
> t=0 0
> m=audio 16400 RTP/AVP 0 8 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:100 NSE/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
>
>
>
>
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Re: [Asterisk-Users] Is everyone getting mails except me?

[Matt]

I have recived 7 mails since that time this morning GMT+10


Ron McCarthy wrote:
I havent got any mails since 2:42 this morning..usually i get at least 
the normal 10-15 a hour, if someone gets this can they reply?


Thanks!
Ron


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Re: [Asterisk-Users] Is everyone getting mails except me?

[Nick Hoffman]

On Thu March 9 2006 02:14, "Ron McCarthy" <[EMAIL PROTECTED]> wrote:
> I havent got any mails since 2:42 this morning..usually i get at least
> the normal 10-15 a hour, if someone gets this can they reply?
>
> Thanks!
> Ron


Hi Ron, I've received many emails from the mailing list over the past 24 
hours. I recommend not sending emails such as these to the mailing list 
though, as it's completely off-topic, and some people unfortunately get 
quite frustrated. The easiest thing to do is check the mailing list's 
archive to see if there are any [new] messages that you're missing. That 
way, you can get an answer immediately.

Cheers,
-- Nick
e: [EMAIL PROTECTED]
p: +61 7 5591 3588
f: +61 7 5591 6588

If you receive this email by mistake, please notify us and do not make any 
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RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

[Michael Collins]

For the record, Douglas is correct on this point of "enterprise-grade"
being on ABE:
http://www.digium.com/index.php?menu=product_category&category=software

Copied and pasted right from the website, it says:

Asterisk Business Edition(tm)
Digium(tm), the leader in open source telephony, offers Asterisk
Business Edition, an enterprise-grade version of its acclaimed open
source PBX for the Linux operating system. This version provides tested
reliability of critical functions and features, tailored for small- and
medium-sized business applications.

Now, as to the debate about what is and is not available in an
"enterprise-grade" product, I will have to defer to those who actually
use Asterisk in the enterprise - I only use it for tinkering and minor
voice broadcasting campaigns.

-MC

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Douglas Garstang
> Sent: Wednesday, March 08, 2006 7:19 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp
traffic
> 
> I can't be bothered looking for the link right now, but it's
definitely
> stated somewhere on Digium's website.
> 
> -Original Message-
> From: Alexander Lopez [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, March 07, 2006 3:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp
> traffic
> 
> 
> To retort, Digium has ever to my knowledge, stamped an 'Enterprise
> Grade' mark on the product.  If you are worried about a single point
of
> failure you may want to replace your toaster.
> 
> Asterisk is missing a 'few features' no doubt about it, but it is open
> source, it will be a welcome addition if you would like to add
> multi-homing support in, might as well do media multi-homing with call
> diversity. This will definably be a non-trivial re-architecture of the
> core.
> 
> The 'missing a few features' way of thinking is what has made Asterisk
> what it is today.
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
[mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Douglas Garstang
> > Sent: Tuesday, March 07, 2006 11:46 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp
> traffic
> >
> > Pardon my candour, but for a product Digium calls 'enterprise grade'
> it
> > sure seems to be missing a few features.
> >
> > -Original Message-
> > From: Alexander Lopez [mailto:[EMAIL PROTECTED]
> > Sent: Tuesday, March 07, 2006 9:39 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp
> > traffic
> >
> >
> > Asterisk does not like multiple interfaces in the way you are
> configured.
> > You can either:
> >
> > A) use the bindaddr in the sip.conf to limit where the packsge come
> and
> > go.
> >
> > B) use an outside traffic manager
> >
> > Look up the archives, kpf explained why this would not work, as
> asterisk
> > can't do load balancing at this time
> >
> >
> > -Original Message-
> > From: "Robert Webb" <[EMAIL PROTECTED]>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>  > [EMAIL PROTECTED]>
> > Sent: 3/7/06 11:27 AM
> > Subject: Re: [Asterisk-Users] Oh this is bad bindaddr and rtp
> traffic
> >
> >
> > On Tue, 7 Mar 2006 09:12:25 -0700
> >   "Douglas Garstang" <[EMAIL PROTECTED]> wrote:
> > > I have a configuration where RTP traffic is going out
> > >interface pub0, and coming back into through pub1.
> > > I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an
> > >shows:
> > >
> > > udp0788 0.0.0.0:50600.0.0.0:*
> > >
> > > which means that Asterisk is listening on all addresses
> > >(on all interfaces?).
> > >
> > > Anyway, when the RTP traffic comes back in on interface
> > >pub0, Asterisk does nothing with it. A 'rtp debug' shows
> > >it's receiving the RTP packets, it just seems it does
> > >nothing with them.
> > >
> > > Anyone seen this?
> > >
> > > Doug.
> > >
> > >
> >
> > I thought all RTP was controlled through rtp.conf and only
> > the SIP traffic was controlled through SIP.conf. I am not
> > sure what settings, beside the RTP port range, you can out
> > into the rtp.conf though.
> >
> > Robert
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Re: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3

[C F]

Douglas Garstang,
Your inability to keep your mouth shut (err hands closed when writing
emails) is sometimes astonishing.


On 3/7/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> Docs? Polycom has docs? Where would one find this fabled land of... err I 
> mean Polycom does stating what ftp servers are supported?
>
> Doug.
>
> -Original Message-
> From: Ken D'Ambrosio [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, March 07, 2006 12:12 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3
>
>
> HTTP's nice, but FTP does the job.  Check the docs for supported FTP
> servers -- many of the stock Linux FTP servers will give the exact problem
> you discussed, below.  I should know -- took me almost a week before
> trying proftpd, and WHAMMO, worked like a champ.
>
> -Ken
>
> On Tue, March 7, 2006 12:37 pm, William M Conlon wrote:
> > I spent a weekend battling similar issues with 501s, using FC4/
> > proftpd.  I finally switched from FTP to HTTP.
> >
> >
> > On Mar 7, 2006, at 9:53 AM, Kristian Kielhofner wrote:
> >
> >
> >> Hello everyone,
> >>
> >>
> >> Please forgive the exclamation points but I have been battling
> >> this one off and on for about four days now.  Sorry for the cross post.
> >>
> >> It all started with a box of IP 501s.  I contacted my reseller and
> >> obtained the latest BootRom and SIP firmware.  Unzipped, configured,
> >> copied over to my FTP server (running AstLinux, of course).  The phone
> >> booted, so far so good.  Updated bootrom, nice.  Rebooted again. Updated
> >> sip firmware.  Also nice.
> >>
> >> Upon the next reboot, the wheels started falling off.  The phones
> >> would not get changes I made to any of the .cfg files.  Several phones
> >> would take 20 minutes or more to boot, only to display a "0x4000 config
> >> file error".  What happened?
> >>
> >> I have been using various Polycom's with AstLinux (and vsftpd
> >> 2.0.3 that I include with it) for quite some time, with no problems
> >> whatsoever.  Until now.
> >>
> >> I had been running bootrom 3.0.1 and various versions of the SIP
> >> image at several other sites with no problem.  At this point I was still
> >> unable to accept the fact that I might not be able to run this latest
> >> bootrom.  After many trial and tribulations, I finally rsync'ed (with
> >> -avr) the FTP directory from the AstLinux machine to
> >> my laptop running CentOS 4.  I configured the vsftpd daemon (version
> >> 2.0.1) IDENTICALLY (with the exception of PAM and TCP
> >> wrappers) and crossed my fingers...
> >>
> >> After re-configuring the IP 501 to use my laptop, imagine my
> >> surprise when the most problematic of them booted right away without
> >> problems. Again and again, everything was fine.
> >>
> >> So now I just had to break out ethereal and see what was going on.
> >> While I have not completely finished my analysis, it appears that
> >> Polycom firmware 3.1.3 bombs out when transferring files with
> >> vsftpd 2.0.3.  The symptom appears to be repeated TCP SYNs from the
> >> Polycom to the ftp daemon on port 20.  The Polycom will keep
> >> retrying and increment its source port number by one every few minutes.
> >> Like I said, I need to dig into this more, but I figured
> >> I'd report what I know and see if anyone out there can fill in the
> >> holes.
> >>
> >> Here's what I did.  It appears that BootRom 3.1.3 works with
> >> vsftpd 2.0.1, so I placed bootrom 3.0.1 (which I know works with vsftpd
> >> 2.0.3) on my CentOS server and downgraded the phone to
> >> 3.0.1.  I then placed 3.0.1 and SIP app 1.6.5 (which I was using
> >> the whole time, btw) on my AstLinux server running vsftpd 2.0.3.
> >>
> >> All was good.  So now I am successfully running with the following:
> >>
> >>
> >> Polycom IP 501
> >> Bootrom 3.0.1
> >> SIP 1.6.5
> >> AstLinux 0.3.7
> >> vsftpd 2.0.3
> >>
> >> I will also try to fix (or workaround) this by trying the following:
> >>
> >>
> >> upgrading AstLinux to include vsftpd 2.0.4 trying an intermediate
> >> BootRom release between 3.0.1 and 3.1.3
> >> (find out exactly where/when it broke)
> >> trying an even newer Polycom BootRom when it becomes available upgrading
> >> the kernel in AstLinux (I doubt that's it) fiddling with iptables rules
> >> in AstLinux (iptables was loaded, but obviously 3.0.1 doesn't have a
> >> problem with it)
> >>
> >> This also might be related to the problems described here:
> >>
> >>
> >> http://forums.digium.com/viewtopic.php?
> >> p=14847&sid=6e70577c37bd345cfc164a01e64e113a
> >>
> >>
> >> Any thoughts?  Comments?  Suggestions?
> >>
> >>
> >> P.S. - I will be updating the Polycom config files at http://
> >> www.krisk.org/asterisk/pcom/ to reflect some new changes in this firmware
> >> release.  I just need to get my phones working first :)!
> >>
> >> --
> >> Kristian Kielhofner
> >> 

[Asterisk-Users] MINNESOTA: TwinCities Asterisk Users Group - Saturday 03/11/2006

[asterisk_help]

SPONSORED THIS MONTH BY: SOUND CHOICE COMMUNICATIONS LLC
  "Keep in touch with the World"

Hello,

The next Asterisk Users Group meeting has been scheduled for this Saturday 
March 11th at 11:30am.


Meetings are held monthly on the second Saturday of each month, excluding 
July and December.


Meetings are held at Sound Choice Communications LLC...
http://maps.google.com/maps?oi=map&q=7839%2012th%20Ave%20S%2055425

Sound Choice Communications is located in Bloomington Minnesota, just 1/2 
mile west of the Mall of America. The address is: 7839 12th Ave S, 
Bloomington Minnesota 55425.  We are just south of Hwy494 on 12th Ave. 
-12th Avenue is one exit west of Hwy 77 (Ceder Ave).



This month we'll hear from Shane Young and Dave Walters as they discuss 
integrating Asterisk with Tivo, Home security, Home Audio, and possibly 
X-10.


If you're having a problem with Asterisk, bring your questions to a 
meeting for free help. We love helping new users!


Come to a meeting to meet other asterisk users, see asterisk solutions, 
win a door prize, eat food, or for the good company, to look for work, 
if your looking for employees, to go out for a drive, to get out of your 
house, whatever, JUST COME TO THE MEETING!


Last month we gave away two licenses for the Cepstral Text to Speech 
software voices. Thank's Cepstral for your support!


In November we gave away an autographed copy of the O'Reilly book 
"Asterisk - The Future of Telephony". All three authors, plus Mark Spencer 
personally signed the book.


New visitors can help themselves to FREE FXO Interface cards (So you can 
connect your phone line, and have a timing source for meetme and IAX 
protocols). Some members have been known to swap hardware at the 
meetings. Have extra VoIP gear, looking for VoIP gear?  There's plenty of 
hardware to see. Have you been to a meeting recently?


Please come and share your own ideas and learn from others. As always, 
free food.



We are always looking for help with meeting topics. If you feel like 
taking the lead, please do and simply let me know if you need anything.


Meeting starts at 11:30am and parking is available in the rear of the 
building. Runs about 2 hours or less, and we'll order Pizza to the meeting 
for lunch.


Look forward to seeing you there.

http://www.voip-info.org/tiki-index.php?page=Asterisk%20User%20Group%20TwinCities%20Minnesota%20USA


If you have a product or service you'd like to introduce to our members, 
send a private message to ejo1(at)soundchoicecomm.com and we'll see if we 
can't get you listed as next month's sponsor.


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Re: [Asterisk-Users] 160 analogue phones..

[Conrad Wood]

On Sun, 2006-03-05 at 16:05 +0200, Tele Cost Price Reducer wrote:
> Conrad,
> i would go with following solution:
> 1. 6 sets of Audio Codes of 24 FXS ports conected by SIP accounts to
> the system. the type is MP 124. then you open the conector on the
> initial MDF and then the users have the same phone on their table
> 2. one dual Xeon system (or even stronger - 2 Dual Core system). such
> a configuration can take 60 calls at g711.
> 3. 16 IP phones for the medium up users
>  
I quite like the idea of the audio codes MP124 - it was my initial
feeling to use something like that. I'll give that a go. Thanks a lot!!
to save others some googling I found the product at:
http://www.voipsupply.com/product_info.php?cPath=3_26&products_id=207

Conrad


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RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel

[Sina Bahram]

I sent my reply to this to your off list email to me, which I greatly
appreciate.
 
We can send the results once we fix the problem, to the list?

Take care,
Sina

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Daragon
Sent: Wednesday, March 08, 2006 5:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6
kernel

Sina, hi;

Let's just do a little recap.

You've downloaded zaptel-1.2.4 and done the

make linux26
make install
make config

thing on it.  If you don't uncomment anything, the builds complete without
error and modules are installed in

/lib/modules/`uname -r`/extra.

You've performed the 2.6 kernel udev configuration :


edit /etc/udev/rules.d/50-udev.rules

and insert the lines :

KERNEL="zapctl",NAME="zap/ctl"
KERNEL="zapchannel",NAME="zap/channel"
KERNEL="zaptimer",  NAME="zap/timer"
KERNEL="zappseudo", NAME="zap/pseudo"
KERNEL="zap[0-9]*", NAME="zap/%n"


Assuming you're using a user called asterisk...

edit

/etc/udev/permissions.d/50-udev.permissions


and insert :

zap/* asterisk:asterisk:660


If running

/etc/init.d/zaptel start

still fails, then  run

/etc/init.d/zaptel stop

and then

sh -x /etc/init.d/zaptel start

You should be able to work out what's failing from the output here. If you
can't, post the output to the list or email it to me.

If, for example, modprobe is failing on ztdummy.ko, then run

strace modprobe ztdummy

and look at the output. This will identify problems like the modules being
in a directory that modprobe isn't looking at, &c &c.

Again, if the cause isn't clear either post the last (say) 20 lines of the
strace err... trace her or email them to me.

Let's put this one to bed, huh ?

jd
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RE: [Asterisk-Users] System Design

[Colin Anderson]

This 
doesn't directly answer your question, because every integration scenario is 
different, but one of the nice things about Asterisk is that the barrier to 
entry to get a system working and play around with it is very low. What you 
might want to consider doing is get your Asterisk box working, minus the PRI 
card, get the Cisco phones running (you're going to buy them anyway) and put 
them in place, side by side with the current phones, and just have the everyone 
play with them. That alone will answer a lot of your questions about how to 
engineer it without commiting to a particular way. 
 
However, in your remote office, I would ditch the crap router and at 
least use a Monowall http://m0n0.ch/wall/ because you can 
prioritize traffic with it, it's super easy to set up, and you can make it work 
with odds and ends you have laying around. If you have more than one static IP, 
you can even do a Monowall to Monowall VPN and leave your PIX in place, and then 
run the VoIP over the VPN. Monowall supports IPSec VPN's, so you can interface 
it with a lot of other firewalls out there, including Pix. 
 
Running VoIP over a VPN is sometimes problematic (but sometimes it works 
great!) so again you can try it out without committing. 
 
Sometimes it makes sense to have a remote Asterisk server at the other 
end and route calls via IAX, IAX is like a VoIP dream protocol, but on the other 
hand it adds complexity where complexity is undesirable. You should try it both 
ways: Stick in a remote Asterisk server on the other end, route calls via IAX, 
and also have some Cisco's register with your main Asterisk server over SIP 
(both with and without the VPN)
 
The 
Dell will probably be fine, compatibility issues with Digium TDM cards 
nonwithstanding (there are some - ask Digium when you buy) and in your case, 
overkill. I'm running a Netfinity Xeon 550 (yup, 550 Mhz) with 2 Te110P 
cards right now supporting 180 users in 32 locations in a 50 mile radius. 
Looking at the console right now I have 36 of 46 channels open to my PRI's, 50 
mixed SIP and IAX calls, and top says about 16% with load average about .53. And 
I'm recording all the calls. 
 
On my 
remote IAX sites (30), I have between 2 to 5 users that do SIP to a local IAX 
server, then IAX here to the main office and out the PRI. What's running on the 
remote servers? Frigging P-II 233's. That's all. The reason it works is because 
I am careful with codec selection so there's no transcoding. And the call 
quality is just fine, thank you. 
 
Ask 
the boss for a couple of weeks to experiment, get the gear, and test. That will 
give you the optimum result, instead of my jackass opinion.
 
hth

  -Original Message-From: Jason Adams 
  [mailto:[EMAIL PROTECTED]Sent: Tuesday, March 07, 2006 4:26 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: [Asterisk-Users] System 
  Design
  Hey 
  Everyone,
   
  We are in the 
  works of planning a new * installation for our company.  We have 20 users 
  in our main office and 5 users in a remote office a couple of states 
  away.  Our call volume for the main office will be anywhere from 5-10 
  concurrent calls.  The remote office will have about 3 heavy users with 
  two users making calls occasionally.
   
  Right now we have 
  an existing PBX.  We have a T-1/PRI coming into the main office and a DSL 
  connection at the remote office.  We have a Cisco 2610/PIX 501 at the 
  main office a cheesy linksys router at the remote site.
   
  We are planning on 
  purchasing new Cisco IP phones for everyone.
   
  My main question 
  is this:  What type of hardware/network design would be best for this 
  situation?  Would a full T-1 at the remote site work with a VPN between 
  the offices?  Or would a higher bandwidth DSL work with a VPN?  Or 
  should we move to a Point-to-Point connection?  What type of hardware 
  would be best for the end-to-end communication in regards to QoS?  I know 
  the PIX 501 doesn't support it.
  Would it be best 
  to have two * servers in each office or for that call volume at the remote 
  office does it make sense?  I was thinking of a Dell Power Edge server 
  with 4GB of ram and a dual processor.. is that enough?
   
  Sorry for all the 
  questions!
   
   - 
Jason
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Re: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3

[Kristian Kielhofner]

Azfhasterisk wrote:

We had the same issue but we found that it was really the MS proxy server
that the phone was going though. Set it up to use a different route out to
the server and everything worked fine.

Had to prove it to the admin at the location too, that was fun!

Rick



Rick,

	Even though I like to blame MS products wherever possible, I don't 
think this one was there fault.  Now I'm really starting to think that 
Polycom broke something in the FTP protocol on 3.1.3.


--
Kristian Kielhofner
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RE: [Asterisk-Users] Reverse group in zapata.conf

[Michael Collins]

> Hey all,
> 
> I have a situation where I have 8 lines from the phone company in a
hunt
> group coming in to my asterisk box.  These are the same lines I'm
using
> for outgoing calls ( named g0 ).
> 
> The problem arises when someone dials our number at the same time
> asterisk tries to put a call out on one of the zap channels in the g0
> group.  This has happened twice that I know of so far, once to myself.
> Asterisk opens the line before it's answered, and tries to dial.  This
> has the effect of connecting the outside caller to the dialing party,
> which is the problem.
> 
> My rather messy solution would be to have a reverse 'group' command in
> my zapata.conf file.  So if I try dialing out on g1 ( my reverse
group,
> 24-17 ), it starts at the top and works it's way down.  Meanwhile, my
> external hunt group would still ring normally ( 17-24 ), thus
minimizing
> the potential for conflict to a level that I'm comfortable with.
> 
> Is this possible?  If it isn't, I plan to reverse the order in which
the
> lines are connected to my * box, having the same effect ( with no
> configuration changes.  :) ).  Anybody have any advice why I shouldn't
> do this either?  Any other suggestions?
> 
> Thanks
> 
> Sean Kennedy

Sean, what is your dial command?  I believe that if you use capital G0
instead of lowercase g0 then the dial out will start at the bottom of
the hunt group and work it's way up, that is, it will start with line
number 8, then 7, etc. while your inbound hunt will start with line 1,
then line 2...

I think it will look like this:
exten => 123,n,Dial(ZAP/G0/2025551212) instead of "Dial(ZAP/g0/...)"

Let us know if that works.  

-MC
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Re: [Asterisk-Users] System Design

[Joseph Tanner]

Lot of questions, lots of variables, but I'll touch base on a few things.

5-10 concurrent calls is hardly anything.  A plain T1 will more than
handle that, even at ulaw or alaw (non)compression.  Throw in a decent
codec, and 10 calls won't even put a dent in your T1.  Heck, it'd
handle all 20 users in your main office, and the 5 users in your
remote office with G729, no problem.

How reliable is the remote office's DSL connection?  I'd make sure you
have a static ip for it (dynamic ips are just slightly problematic,
especially if you have slightly flaky service, coupled with a slightly
flaky modem).  If it's reliable, then just keep that.  What's the
connection speed?  Need to know the upload and download.  If it's
ADSL, then the upload will be a fraction of the download, and will be
the limiting factor.

Since I don't know your specific setup, I can't tell you specifically
what to do.  I'll make some guesses though.  Keep DSL.  No need to use
VPN just for asterisk.  Make sure each end has a static ip (dynamic ip
will work, but is harder to setup and more prone to errors).  Have
each asterisk box register to the other.  For normal incoming and
outgoing calls, just have the asterisk box at that particular location
handle it (no need for the remote office to connect to the main
office's asterisk box, then call out via iax or sip for a
long-distance phone call).  You can create "local" extensions that
when dialed, will ring a person on the other asterisk box.  I.e., a
user at the main office can dial 2001, and get a user at the remote
office.  If you deal with call queues you can group users from both
offices together, no problem.

A T1 or a point to point connection at the remote office would work,
but is probably unecessary.  If their DSL connection is flaky and
unreliable, then start looking at both options.  I'd probably go with
whichever is cheapest, be sure to factor in equipment costs (you can
generally lease equipment with a T1 line, but not with a point to
point connection).

As far as server specs, if all it's going to run is asterisk, then
that's overkill even if it was handling all the calls.  If you think
you need that much server but are on a budget, then get one setup for
dual processors but with just one installed, and less ram but that has
room to add more.  If budget's not a problem, I say go for it!  That
system should last you for quite a while.

As for QOS, sorry I can't help you there.  You could get a cheap
router that has QOS built-in, or run a separate low-end server just
for QOS.  Personally my asterisk box also serves as my nat server, so
I just run QOS directly on it.  It's probably not something you want
to do in an office environment, but it's better than no QOS at all. 
Hopefully someone else will give you some good advice on QOS
equipment.

Joseph Tanner

On 3/7/06, Jason Adams <[EMAIL PROTECTED]> wrote:
>
> Hey Everyone,
>
> We are in the works of planning a new * installation for our company.  We
> have 20 users in our main office and 5 users in a remote office a couple of
> states away.  Our call volume for the main office will be anywhere from 5-10
> concurrent calls.  The remote office will have about 3 heavy users with two
> users making calls occasionally.
>
> Right now we have an existing PBX.  We have a T-1/PRI coming into the main
> office and a DSL connection at the remote office.  We have a Cisco 2610/PIX
> 501 at the main office a cheesy linksys router at the remote site.
>
> We are planning on purchasing new Cisco IP phones for everyone.
>
> My main question is this:  What type of hardware/network design would be
> best for this situation?  Would a full T-1 at the remote site work with a
> VPN between the offices?  Or would a higher bandwidth DSL work with a VPN?
> Or should we move to a Point-to-Point connection?  What type of hardware
> would be best for the end-to-end communication in regards to QoS?  I know
> the PIX 501 doesn't support it.
> Would it be best to have two * servers in each office or for that call
> volume at the remote office does it make sense?  I was thinking of a Dell
> Power Edge server with 4GB of ram and a dual processor.. is that enough?
>
> Sorry for all the questions!
>
>
>
>  - Jason
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>
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>
>
>
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RE: [Asterisk-Users] Reverse group in zapata.conf

[Greg Scasny]

I think what your asking is pretty easy, just change the lowercase g in
your extensions.conf file to an uppercase G. If you have a TRUNK type
variable declared, this will be cake. If not you will need to change the
little g, as in Zap/g1 to Zap/G1 everywhere you have it used.

Hope that helped>> 


Gregory P. Scasny
Golden Technologies, Inc.
http://www.golden-tech.com
[EMAIL PROTECTED]
219-462-7200 - Ph.
574-233-1300 - Ph.
(866) 806-7127 - Toll Free
219-462-7257 - Fax.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
Kennedy
Sent: Tuesday, March 07, 2006 8:04 PM
To: Asterisk - Users
Subject: [Asterisk-Users] Reverse group in zapata.conf

Hey all,

I have a situation where I have 8 lines from the phone company in a hunt
group coming in to my asterisk box.  These are the same lines I'm using
for outgoing calls ( named g0 ). 

The problem arises when someone dials our number at the same time
asterisk tries to put a call out on one of the zap channels in the g0
group.  This has happened twice that I know of so far, once to myself.  
Asterisk opens the line before it's answered, and tries to dial.  This
has the effect of connecting the outside caller to the dialing party,
which is the problem.

My rather messy solution would be to have a reverse 'group' command in
my zapata.conf file.  So if I try dialing out on g1 ( my reverse group,
24-17 ), it starts at the top and works it's way down.  Meanwhile, my
external hunt group would still ring normally ( 17-24 ), thus minimizing
the potential for conflict to a level that I'm comfortable with.

Is this possible?  If it isn't, I plan to reverse the order in which the
lines are connected to my * box, having the same effect ( with no
configuration changes.  :) ).  Anybody have any advice why I shouldn't
do this either?  Any other suggestions?

Thanks

Sean Kennedy
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[Asterisk-Users] [EMAIL PROTECTED] Servers Connecting Portugal to Brazil (offices)

[Marco Mouta]

Hi all,

I'm planning to connect 2 office from one company.

I'm the developer, so i hope i can get all the features working well.

[EMAIL PROTECTED](Portugal)-IAX2/[EMAIL PROTECTED](Brazil)

1- First i'm integrating Asterisk in Portugal's company office, one
[EMAIL PROTECTED] with TE110P connecting to an old PBX. (the same is done
in Brazil, but only VoIP no TE110P)

For [EMAIL PROTECTED] PCs:
-P4 1GRam 100GHard Disk
(About 20 to 50 users initially)

2- For Portugal internal VoIP calls as well as VoIP to PSTN i think it
would be all ok.

I would like to hear from you? (using Alaw will I need QoS on our LAn?
we have Gigabit Lan)

Main doubts are the the connection between Brazil and Portugal. Will
it work only using IAX2 and Alaw?

Will I need G729 for this connections? Does DTMF works fine with G729?

(I'm planning maximum 4 simultaneous calls to Brazil)

We have broadband connection 4Mbit.


I hope this Excellent mailing list could help me on giving me some
Feedback and or advices/tips.

Best regards,
Marco Mouta
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[Asterisk-Users] overlap dialing with polycom?

[asterisk]

Where's the setting for overlap dialing with Polycom IP601?

-Dan
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[Asterisk-Users] Professional Recordings

[Waldo Rubinstein]

Can anyone recommend a company that does professional Asterisk  
recordings for things like IVR, greetings, MOH, announcements, etc?


Thanks,
Waldo
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[Asterisk-Users] Unicall, Fax and Echo cancellation

[Carlos Chavez]

    Does Unicall support disabling echo cancellation on an E1 circuit when a fax tone is detected?  I think this is the reason why I cannot send or receive faxes on my Asterisk server.





-- 
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001








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Re: [Asterisk-Users] sending text to display of sip phones

[asterisk]

On Wed, 8 Mar 2006, Alejandro Vargas wrote:

I red that it is possible to send instant messages to the displays of
sip phones. How can I do it using Asterisk?


Your phone needs to support it. Few do.

-Dan
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RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic

[Douglas Garstang]

Asterisk calls the Business Edition 'enterprise grade'. It's right there on the 
Digium website. It's the same dang code as the open source version, just older. 

-Original Message-
From: Matt Riddell [NZ] [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 07, 2006 6:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Oh this is bad bindaddr and rtp
traffic


Douglas Garstang wrote:
> Pardon my candour, but for a product Digium calls 'enterprise grade' it sure 
> seems to be missing a few features.

Um...it's Open Source.  Why don't you add the features you require
yourself or pay someone to add them for you...

This is your third similar post in as many days.

-- 
Cheers,

Matt Riddell
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[Asterisk-Users] Re: PLEASE respond: how to get Asterisk to change coders on RTP handoff?? HELLO???

[Dan Miller]

So, when I get no comments on this at all, either here or on any of the 
forums, does that mean nobody knows what I'm talking about??  Or does 
nobody know the answer??  Or is it just a stupid question and nobody wants 
to bother telling me where to look??
 
It *is* a question that I have to answer somehow; I've read all through 
TFOT and see nothing relevant to this issue.  It's silly to spend $15000 on 
a G723 license just so I can play back menu messages from Asterisk (since the 
actual call decoding is done by the external boxes, which have already paid the 
licensing fees).
 
Dan Miller

  - Original Message - 
  From: 
  Dan Miller 
  
  To: asterisk-users@lists.digium.com 
  
  Sent: Monday, March 06, 2006 10:11
  Subject: PLEASE respond: how to get 
  Asterisk to change coders on RTP handoff??
  
  I have a hardware FXO/FXS which handle my voip calls, and they 
  support G723 internally.  Asterisk hands off these calls just fine, and 
  everything works, as long as I don't want PBX menues 
  available...  The problem is, once I want it to return messages, it will 
  only return them as GSM... which is fine, since my FXO/FXS support multiple 
  coders.  However, even though Asterisk lets me specify a list of valid 
  coders, it will only use one...
   
  I want Ast to use GSM to playback messages, then when it hands off the 
  call to the endpoints, it should tell them to use G723 in the RE-INVITE 
  messages... I don't see any way to get it to do this; *is* there some 
  way??
   
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Re: [Asterisk-Users] MWI, SER and asterisk

[hgaillac-sip]

Hello,

You can use ser as an outbound sip proxy and asterisk
as a register server .

Your sip agents will get MWI, ...

Harry
--- Christian B <[EMAIL PROTECTED]> a écrit :

> Hi Sharon!
> 
> This is pretty difficult, i was not able to
> implement it so far(though
> my ser-skills are pretty basic).
> At http://www.voip-info.org/wiki-Asterisk+at+large
> you'll find some
> howto's, method 2 seems to be the most promising to
> me...
> 
> regards
> christian
> 
> On Tue, 7 Mar 2006 15:36:57 -0600
> Sharon <[EMAIL PROTECTED]> wrote:
> 
> > I have my peers registered to SER.asterisk seems
> to be sending mwi for
> > the peers seen in the sip show peers CLI command.
> i have my ser server
> > registered with asterisk as a type=friend and all
> clients register to
> > ser.how do i get mwi to work for these clients
> registered to SER.
> > 
> > Thank you,
> > -AA
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[Asterisk-Users] Re: System Design

[Paul Davidson]

Date: Tue, 7 Mar 2006 18:26:12 -0500From: "Jason Adams" <
[EMAIL PROTECTED]>Subject: [Asterisk-Users] System DesignTo: "Asterisk Users Mailing List - Non-Commercial Discussion"Message-ID:<[EMAIL PROTECTED]>Content-Type: text/plain; charset="us-ascii"
Hey Everyone,We are in the works of planning a new * installation for our company.We have 20 users in our main office and 5 users in a remote office acouple of states away.  Our call volume for the main office will be
anywhere from 5-10 concurrent calls.  The remote office will have about3 heavy users with two users making calls occasionally.Right now we have an existing PBX.  We have a T-1/PRI coming into themain office and a DSL connection at the remote office.  We have a Cisco
2610/PIX 501 at the main office a cheesy linksys router at the remotesite.We are planning on purchasing new Cisco IP phones for everyone.My main question is this:  What type of hardware/network design would be
best for this situation?  Would a full T-1 at the remote site work witha VPN between the offices?  Or would a higher bandwidth DSL work with aVPN?  Or should we move to a Point-to-Point connection?  What type of
hardware would be best for the end-to-end communication in regards toQoS?  I know the PIX 501 doesn't support it.Would it be best to have two * servers in each office or for that callvolume at the remote office does it make sense?  I was thinking of a
Dell Power Edge server with 4GB of ram and a dual processor.. is thatenough?Sorry for all the questions! - Jason-- next part --Jason-
You're right, that's a lot of questions.  Let me try to net it out a little for you.First off, it sounds as if you're using the Internet to connect the two offices.  Understand- nothing presently there will provide QOS over the Internet- from that perspective, your existing equipment is just fine.  If you're considering making changes, and budget is not an issue, a private WAN setup- Frame Relay, for instance, that provides low latency between the two points, is what you're looking for, as you can perform QOS on it.  Speedwise, however, depending on the level of compression you go with, at 5 simultaneous calls (my assumed maximum with 5 remote users- YMMV), you really dont need anything faster than a 256K connection between the two points- assuming the latency can be kept in the 60ms range or less.  DSL, specificall aDSL is notoriously awful for high-bandwidth VoIP applications, as it's asymetric (faster download than upload in general), and the speed will vary at random based on the carrier and time of day.  If you compress, you can get away with 128K for the voice portion of the link (Remember, that's 256K for the Voice side, not counting whatever other traffic is going on at the time)- and if you trunk IAX, you can potentially get even smaller.  The big question is- at 20 users and 5 users- how many calls are going on across the VoIP link?
Secondly, consider your PSTN connections.  Are you using a PRI at the main office, and some POTS lines at the remote? Do you need to use a VoIP provider for all of it? Want to get rid of those POTs lines and use the PRI for the remote office as well? All of which will change the equations as far as how much bandwidth and what kind of hardware you need in each office.
Finally, hardware.  That dual CPU machine is a cadillac for 20 users- even 25 users.  I won't go in to my opinion of Dell, that's a theological discussion- but I'd sandbox that setup on something far smaller- a 1Ghz Celeron should be more than up to the task for Asterisk, depending on what else you're doing at the time, even with transcoding going on.  I personally would recommend two low cost servers- look in to astlinux and a Soekris box for the remote office (might be pushing it, but again, it depends on your apps and transcode requirements), and a cheap commodity machine for the main branch- it should be just fine for what you're looking to do.  Trunk the two machines together, and you've got all the power in the world- Dundi or IAX switched dialplans will take care of most of the headache for you.
If you want to dig deeper into details, I'd be happy to offlist- I'm posting this here as more of a general selection/architecture guide.-Paul Davidson PlanCommunications, LLC
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Re: [Asterisk-Users] MWI, SER and asterisk

[David Thomas]

Method 3 is the one I was speeking of. As long as you plan to continue
to have SER in front of Asterisk it should be fine.

David

On 3/8/06, Christian B <[EMAIL PROTECTED]> wrote:
> Hi Sharon!
>
> This is pretty difficult, i was not able to implement it so far(though
> my ser-skills are pretty basic).
> At http://www.voip-info.org/wiki-Asterisk+at+large you'll find some
> howto's, method 2 seems to be the most promising to me...
>
> regards
> christian
>
> On Tue, 7 Mar 2006 15:36:57 -0600
> Sharon <[EMAIL PROTECTED]> wrote:
>
> > I have my peers registered to SER.asterisk seems to be sending mwi for
> > the peers seen in the sip show peers CLI command. i have my ser server
> > registered with asterisk as a type=friend and all clients register to
> > ser.how do i get mwi to work for these clients registered to SER.
> >
> > Thank you,
> > -AA
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[Asterisk-Users] Random Zap port going crazy When channel released after a flash.

[Dennis Walker]

On 1.2.x I have a random problem when a Zap/x channel flashes to transfer 
or make a three way call.

The Zap/x-2 channel is created and the transfer or three way proceeds, but 
on hangup the Zap/x-1 channel fails to destroy the old bridge and asterisk 
goes crazy logging the problem.  Here is an example debug log.

This happens only once a day or so, with 100 or so users transfering and 
three way calling all the time.

Anyone having a simular problem.


Thanks for you help

Mar  7 11:21:29 VERBOSE[8204] logger.c: -- Starting simple switch on 
'Zap/99-1'
Mar  7 11:21:31 DEBUG[8204] chan_zap.c: DTMF digit: 8 on Zap/99-1
Mar  7 11:21:32 DEBUG[8204] chan_zap.c: DTMF digit: 3 on Zap/99-1
Mar  7 11:21:32 DEBUG[8204] chan_zap.c: DTMF digit: 3 on Zap/99-1
Mar  7 11:21:33 DEBUG[8204] chan_zap.c: DTMF digit: 5 on Zap/99-1
Mar  7 11:21:33 DEBUG[8204] chan_zap.c: DTMF digit: 6 on Zap/99-1
Mar  7 11:21:33 DEBUG[8204] chan_zap.c: DTMF digit: 8 on Zap/99-1
Mar  7 11:21:33 DEBUG[8204] chan_zap.c: DTMF digit: 6 on Zap/99-1
Mar  7 11:21:34 DEBUG[8204] chan_zap.c: DTMF digit: 8 on Zap/99-1
Mar  7 11:21:34 DEBUG[8204] chan_zap.c: Enabled echo cancellation on 
channel 99
Mar  7 11:21:34 VERBOSE[8204] logger.c: -- Executing 
SetCallerID("Zap/99-1", "9377738550") in new stack
Mar  7 11:21:34 VERBOSE[8204] logger.c: -- Executing 
SetCallerPres("Zap/99-1", "allowed") in new stack
Mar  7 11:21:34 VERBOSE[8204] logger.c: -- Executing Dial("Zap/99-1", 
"Zap/G1/9373356868||Wg") in new stack
Mar  7 11:21:34 VERBOSE[8204] logger.c: -- Requested transfer 
capability: 0x00 - SPEECH
Mar  7 11:21:34 DEBUG[25354] channel.c: Avoiding initial deadlock for 
'Zap/22-1'
Mar  7 11:21:34 VERBOSE[8204] logger.c: -- Called G1/9373356868
Mar  7 11:21:34 DEBUG[25368] chan_zap.c: Queuing frame from 
PRI_EVENT_PROCEEDING on channel 0/22 span 1
Mar  7 11:21:34 VERBOSE[8204] logger.c: -- Zap/22-1 is proceeding 
passing it to Zap/99-1
Mar  7 11:21:34 DEBUG[8204] chan_zap.c: Requested indication 15 on channel 
Zap/99-1
Mar  7 11:21:34 DEBUG[8204] chan_zap.c: Received AST_CONTROL_PROCEEDING on 
Zap/99-1
Mar  7 11:21:36 DEBUG[25368] chan_zap.c: Enabled echo cancellation on 
channel 22
Mar  7 11:21:36 DEBUG[25354] channel.c: Avoiding initial deadlock for 
'Zap/22-1'
Mar  7 11:21:36 VERBOSE[8204] logger.c: -- Zap/22-1 is ringing
Mar  7 11:21:36 DEBUG[8204] chan_zap.c: Requested indication 3 on channel 
Zap/99-1


Mar  7 11:21:56 VERBOSE[8204] logger.c: -- Zap/22-1 answered Zap/99-1
Mar  7 11:21:56 DEBUG[8204] chan_zap.c: Requested indication -1 on channel 
Zap/99-1
Mar  7 11:21:56 DEBUG[8204] chan_zap.c: Took Zap/99-1 off hook
Mar  7 11:21:56 VERBOSE[8204] logger.c: -- Attempting native bridge of 
Zap/99-1 and Zap/22-1

Mar  7 11:22:03 DEBUG[8204] chan_zap.c: Exception on 145, channel 99
Mar  7 11:22:03 DEBUG[8204] chan_zap.c: Got event Wink/Flash(3) on channel 
99 (index 0)
Mar  7 11:22:03 DEBUG[8204] chan_zap.c: Winkflash, index: 0, normal: 145, 
callwait: -1, thirdcall: -1
Mar  7 11:22:03 DEBUG[8204] chan_zap.c: Already have a dsp on Zap/99-2?
Mar  7 11:22:03 DEBUG[8204] chan_zap.c: Swapping 2 and 0
Mar  7 11:22:03 DEBUG[8204] chan_zap.c: disabled echo cancellation on 
channel 99

>   Mar  7 11:22:03 VERBOSE[8229] logger.c: -- Starting simple switch 
on 'Zap/99-2'

Mar  7 11:22:03 VERBOSE[8204] logger.c: -- Started three way call on 
channel 99
Mar  7 11:22:03 VERBOSE[8204] logger.c: -- Started music on hold, class 
'default', on channel 'Zap/22-1'
Mar  7 11:22:03 DEBUG[8204] channel.c: Scheduling timer at 160 sample 
intervals
Mar  7 11:22:03 DEBUG[8204] chan_zap.c: Updated conferencing on 99, with 0 
conference users
Mar  7 11:22:03 DEBUG[8204] channel.c: Generator got voice, switching to 
phase locked mode
Mar  7 11:22:03 DEBUG[8204] channel.c: Scheduling timer at 0 sample 
intervals
Mar  7 11:22:03 DEBUG[8204] channel.c: Auto-deactivating generator
Mar  7 11:22:03 VERBOSE[8204] logger.c: -- Stopped music on hold on 
Zap/22-1
Mar  7 11:22:03 DEBUG[8204] channel.c: Scheduling timer at 0 sample 
intervals


>   Mar  7 11:22:04 DEBUG[8229] chan_zap.c: DTMF digit: 8 on Zap/99-2
Mar  7 11:22:05 DEBUG[8229] chan_zap.c: DTMF digit: 3 on Zap/99-2
Mar  7 11:22:05 DEBUG[8229] chan_zap.c: DTMF digit: 3 on Zap/99-2
Mar  7 11:22:05 DEBUG[8229] chan_zap.c: DTMF digit: 5 on Zap/99-2
Mar  7 11:22:05 DEBUG[8229] chan_zap.c: DTMF digit: 6 on Zap/99-2
Mar  7 11:22:06 DEBUG[8229] chan_zap.c: DTMF digit: 8 on Zap/99-2
Mar  7 11:22:06 DEBUG[8229] chan_zap.c: DTMF digit: 6 on Zap/99-2
Mar  7 11:22:06 DEBUG[8229] chan_zap.c: DTMF digit: 8 on Zap/99-2
Mar  7 11:22:06 DEBUG[8229] chan_zap.c: Enabled echo cancellation on 
channel 99
Mar  7 11:22:06 VERBOSE[8229] logger.c: -- Executing 
SetCallerID("Zap/99-2", "9377738550") in new stack
Mar  7 11:22:06 VERBOSE[8229] logger.c: -- Executing 
SetCallerPres("Zap/99-2", "allowed") in new

Re: [Asterisk-Users] sending text to display of sip phones

[C F]

I think its sendtext, to cinfirm do a show applications like text from the CLI

On 3/8/06, Alejandro Vargas <[EMAIL PROTECTED]> wrote:
> I red that it is possible to send instant messages to the displays of
> sip phones. How can I do it using Asterisk?
> --
> Alejandro Vargas
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Re: [Asterisk-Users] OT: Polycom Registration Weirdness

[C F]

Are the Polycoms doing this on a different network than the Polycoms
not doing this?

On 3/7/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> This is a SER/Polycom question, but I hoped we may have some SER guru's 
> here...
>
> I have a series of Polycom phones that are tying to register with OpenSER. 
> The phone sends a REGISTER message, and OpenSER replies with Unauthorised 
> (all normal). The phone re-sends the REGISTER with the credentials, and 
> OpenSER sends Ok.
>
> Here's where it goes downhill. The polycom's appearance display does not 
> change from an unregistered to a registered state, ie it does not change from 
> an empty phone to a filled in one. It doesn't think it's registered 
> eventhought it's gotten an OK. Then, a regular intervals it keeps trying to 
> register again, because it still thinks it wasn't successful.
>
> I have some other Polycom phones that are not doing this. All have the same 
> SIP software version, and all essentially have the same xml config files, 
> with minor variations. Happening with OpenSER 1.0.0 and 1.0.1
>
> I have pasted ngrep output of one of these below. Anyone got any ideas?
>
> #
> U 216.187.128.72:5060 -> 216.187.140.233:5060
> REGISTER sip:ipt.oneeighty.com SIP/2.0.
> Via: SIP/2.0/UDP 216.187.128.72;branch=z9hG4bKca28b8d3BC755D46.
> From: "Sandy Sauvageau" ;tag=2A2425B5-B64A4132.
> To: .
> CSeq: 1 REGISTER.
> Call-ID: [EMAIL PROTECTED]
> Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, 
> INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER".
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067.
> Max-Forwards: 70.
> Expires: 3600.
> Content-Length: 0.
> .
>
> #
> U 216.187.140.233:5060 -> 216.187.128.72:5060
> SIP/2.0 401 Unauthorized.
> Via: SIP/2.0/UDP 216.187.128.72;branch=z9hG4bKca28b8d3BC755D46.
> From: "Sandy Sauvageau" ;tag=2A2425B5-B64A4132.
> To: ;tag=136c3bb27674cf7e44f7b05275ffaecc.0629.
> CSeq: 1 REGISTER.
> Call-ID: [EMAIL PROTECTED]
> WWW-Authenticate: Digest realm="ipt.oneeighty.com", 
> nonce="440e4b3f113243b90ba483b6a2f243ea51377e2d".
> Server: OpenSer (1.0.0 (i386/linux)).
> Content-Length: 0.
> .
>
> #
> U 216.187.128.72:5060 -> 216.187.140.233:5060
> REGISTER sip:ipt.oneeighty.com SIP/2.0.
> Via: SIP/2.0/UDP 216.187.128.72;branch=z9hG4bKff7cb008AF06A32B.
> From: "Sandy Sauvageau" ;tag=2A2425B5-B64A4132.
> To: .
> CSeq: 2 REGISTER.
> Call-ID: [EMAIL PROTECTED]
> Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, 
> INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER".
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067.
> Authorization: Digest username="2944029", realm="ipt.oneeighty.com", 
> nonce="440e4b3f113243b90ba483b6a2f243ea51377e2d", 
> uri="sip:ipt.oneeighty.com", response="9d8b4708296f3fb88d5cfd453121860d", 
> algorithm=MD5.
> Max-Forwards: 70.
> Expires: 3600.
> Content-Length: 0.
> .
>
> #
> U 216.187.140.233:5060 -> 216.187.128.72:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 216.187.128.72;branch=z9hG4bKff7cb008AF06A32B.
> From: "Sandy Sauvageau" ;tag=2A2425B5-B64A4132.
> To: ;tag=136c3bb27674cf7e44f7b05275ffaecc.32b4.
> CSeq: 2 REGISTER.
> Call-ID: [EMAIL PROTECTED]
> Contact: ;expires=3600.
> Server: OpenSer (1.0.0 (i386/linux)).
> Content-Length: 0.
> .
>
>
>
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Re: [Asterisk-Users] Re: ON DEMAND call Recording

[Ronald Wiplinger]

Tomislav Parcina wrote:

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Martin Joseph

Sent: 7. ozujak 2006 18:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: ON DEMAND call Recording


On Mar 7, 2006, at 2:38 AM, Giridhar Bandi wrote:



ya i found it it *1 to start recording from the caller end

  

Also pushing *1 again stops recording.



Do you know how to send that recording to e-mail address that is specified in 
voicemail.conf? That will be a real cool option.

  

I would find two possibilities:
1. on demand. Dial another extension number after the call, what 
executes a system command

2. automatically. Add in the dialplan the system command after hanging up.

(just to start somewhere)


bye

Ronald Wiplinger
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[Asterisk-Users] Dial command

[Ronald Wiplinger]

I have an ZAP extension number 222 which is connected instead to a phone
to a FXS/FXO converter and from there to a CDMA gateway.

To dial my mobile phone I use:

222 (wait 2 seconds) 09123456789

I cannot figure out how to write this into the dialplan as a default number!

222 as above I will use for dialing any other number, but I want to add
this phone as an extension which rings if 601 is not picking up within
20 seconds.

How to write this?


Some parts of my existing dial plan:
[Globals]
PHONE_222=ZAP/2r1; transfer to mobile phone <===
hier I want to add the mobile phone number

[incoming]
...
exten =>
s,7,Dial(${PHONE_601}&${PHONE_621}&${PHONE_603}&${PHONE_610},30,tr)  ;
ring phone_601, 621 & 603 for 30 seconds
exten => s,8,Dial(${PHONE_222},30,tr)  ; ring phone_222

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Re: [Asterisk-Users] Asterisk Prepaid Card

[Dovid Bender]

why not use astcc ? it comes with asterisk and does
all that you have requested. we have scripts running.
one that works via CID and one the user enters the
number.

--- leonimar cape <[EMAIL PROTECTED]> wrote:

> Hi group,
> 
> I am currently looking for a prepaid application
> that
> can do the following:
>> Use the Caller ID/Card Number for
> authentication
>> Can map a rate plan on a specific Caller
> ID/Card
> Number
>> Supports prepaid functionality in terms of
> trunk
> connection.
> 
> These functionalities seems feasible in A2billing
> but
> the problem is I cannot find a proper documentation
> of
> setting it up. Can anyone show point to the right
> direction? Does any one has a better suggestion? 
> 
> Thank you very much in advance!
> 
> Leonimar Cape
> 
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Re: [Asterisk-Users] Asterisk download file locations

[Dovid Bender]

we mirror all the files our selves so our scripts work
flawlessly. 

--- Alistair Cunningham <[EMAIL PROTECTED]>
wrote:

> This is a request to the website manager for
> asterisk.org.
> 
> The build scripts for our ITSP product include the
> URLs to download the 
> Asterisk files, such as:
> 
> wget
>
"http://ftp.digium.com/pub/asterisk/asterisk-1.2.5.tar.gz";
> 
> However, if a new version is released,
> asterisk-1.2.5.tar.gz is moved to 
> the "old" directory. This breaks our scripts until
> we can update them 
> and send them to our resellers.
> 
> Would it be possible to have a fixed address for a
> particular asterisk 
> release that will never (or at least not for a long
> time) change? 
> Perhaps put all (except very old) versions in the
> same directory, with a 
>   'latest' link to the latest one?
> 
> -- 
> Alistair Cunningham,
> Integrics Ltd,
> +44 20 799 39 799
> sip:[EMAIL PROTECTED]
> http://integrics.com/
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RE: [Asterisk-Users] res_mysql.conf & DNS SRV lookup

[Bob McDowell]

Good to know I'm not the only one...

I thought perhaps I had been expelled from the list...


Bob McDowell

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, March 07, 2006 10:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] res_mysql.conf & DNS SRV lookup

Good grief! I posted the message below at 1:17pm... and it appeared on
the list after 8pm.
Nice

-Original Message-
From: Douglas Garstang
Sent: Tue 3/7/2006 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: RE: [Asterisk-Users] res_mysql.conf & DNS SRV lookup



Yay!

-Original Message-
From: Olle E Johansson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 07, 2006 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] res_mysql.conf & DNS SRV lookup



7 mar 2006 kl. 19.03 skrev Douglas Garstang:

> My bad. SRV lookups work, but Asterisk only uses the first
entry
> right? This means there's no redundancy.

That is correct. That is what we try to fix.

/O
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[Asterisk-Users] Problem ChanSpy

[David Guarnido]

Sorry, This is a mistake, sip.conf:

 

[302]canreinvite=no [301]canreinvite=no  Any idea? Thanks     

 

 

 






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[Asterisk-Users] Any way to change dns timeout value? Asterisk hangs if internet unreachable

[Joseph Tanner]

I don't have the most reliable internet connection in the world. 
Whenever it goes out, I can't receive any incoming calls at all, not
even from pstn.  When it first goes out I can still make outgoing
calls through pstn, but eventually that fails too (as does voicemail,
everything's out).  Yes, asterisk and the local phones are all on the
same network and can communicate fine.

Ok, that's the symptom, and I believe I know what's causing it. 
Asterisk seems to be hanging on dns lookups.  After a while, it gets
so bad that it won't process anything at all.  The reason incoming
calls via pstn won't work is because I have a calleridname.agi script
that runs as soon as a call comes in.  Instead of trying for say, 5
seconds and then giving up, asterisk just sits there forever waiting
for it to resolve.  Once asterisk gives up, the caller has hung up
ages ago.  Obviously, I don't want pstn calls to be dependent on my
internet connection, kinda defeats having a pstn line at all.

Now, as soon as the internet connection craps out, I can still make
outgoing calls via pstn, access voicemail, etc.  If it's a long outage
(like this morning, some fiber cut and the whole county is without
internet, redundancy anyone?), eventually everything stops.  I think
it's because asterisk is re-trying to register with a host, before the
dns timed out, and the built-up dns queries just bring the whole thing
to a halt eventually.  This morning after I noticed the internet
connection was down, I tried to call the phone company (through the
pstn line) and could not.  When I watched the CLI, I noticed it try to
call a minute or two after I hung up, quite a delayed reaction.  Also
could not access voicemail.  When the connection came back up for a
minute and crapped back out again, I was suddenly able to access
voicemail and make a call.  Shortly after that, I'd dial a number and
it'd connect after 10 seconds or so.  After that, it wouldn't try to
connect until after the phone received a fast busy.

A workaround was to backup my sip.conf and iax.conf files, then edit
them taking out every single host reference that wasn't an ip address.
 If I left them in and tried to restart asterisk, it would hang on the
first host trying to resolve.  A minute or so later it'd give up and
move on to the second.  Obviously very bad news if you have several
hosts that it needs to resolve (side note, why can't asterisk try to
resolve multiple hosts at once; say one every 5 seconds, so it doesn't
flood your network with dns requests, but also if one host hangs it
can try resolving other hosts while waiting?).

I've looked in dns.c and dnsmgr.c and can't see where I can set a
timeout.  Perhaps it's somewhere else?  Maybe hiding in several files?
 Any ideas?  I'd like to set it to five seconds, this should give most
hosts that aren't down plenty of time to respond.  Perhaps even
better, I could cache dns results and save them to a file?  Run a
background application to query dns servers, if it hangs then asterisk
uses the last good values (and if it's not reachable, no big deal,
asterisk will just move on).

I promise I searched on google before posting here.  The closest thing
I could find is this:

http://bugs.digium.com/view.php?id=3946

Doesn't seem to have a real solution.

Joseph Tanner
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[Asterisk-Users] List Problems

[Dovid Bender]

Is anyone with a yahoo account having problems
recieving emails from the list. I have not recieved
any emails in about 8 hours and I posted something
about 3 hours ago. If anyone knows please email to
asteriskdigium _AT_ yahoo.com

Thanks

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[Asterisk-Users] Calls forwarding to numbers only in user's context

[Bartosz Piec]

Hello,

I'm trying to do call forwarding based on this: 
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding


In the extensions.conf file I have several context defined (local, 
longdistance, mobile, international and so on). Each user can be 
associated with different context (so can make only i.e. local calls). 
How to set calls forwarding only to numbers that are available in user's 
context (so if he has only locals calls he cannot set calls forwarding 
for mobile phones)?


I'm using this for forwarding:

[forwarding] ; available for all users
; Unconditional Call Forward
exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
exten => _*21*X.,2,Background(auth-thankyou)
exten => _*21*X.,3,Hangup
exten => #21#,1,DBdel(CFIM/${CALLERIDNUM})
exten => #21#,2,Background(auth-thankyou)
exten => #21#,3,Hangup

; Call Forward on Busy or Unavailable
exten => _*61*X.,1,DBput(CFBS/${CALLERIDNUM}=${EXTEN:4})
exten => _*61*X.,2,Background(auth-thankyou)
exten => _*61*X.,3,Hangup
exten => #61#,1,DBdel(CFBS/${CALLERIDNUM})
exten => #61#,2,Background(auth-thankyou)
exten => #61#,3,Hangup

[macro-call-forwarding]
exten => s,1,Set(temp=${DB(CFIM/${ARG1})})
exten => s,n,GotoIf(${temp}?cfim:nocfim)
exten => s,n(cfim),Dial(SIP/[EMAIL PROTECTED])   ; Unconditional forward
exten => s,n(nocfim),NoOp

exten => s,n,Dial(SIP/${ARG1},20,tTwW)

exten => s,n,Set(temp=${DB(CFBS/${ARG1})})
exten => s,n,GotoIf(${temp}?cfbs:nocfbs)
exten => s,n(cfbs),Dial(SIP/[EMAIL PROTECTED]) ; Forward on busy or unavailable
exten => s,n(nocfbs),Goto(s-${DIALSTATUS},1) ; 
NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER


...

[incoming]
;
; Incoming calls.
;

exten => XYY,1,Macro(call-forwarding,YY)

--
Best regards,
Bartosz Piec
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[Asterisk-Users] More 7940 Questions

[Aaron Daniel]

Does anyone know why putting an outbound proxy in the SIP.cnf file 
causes the phone to not pull it's logo from logo_url?


Aaron
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[Asterisk-Users] Cisco 7960 SIP - Displaying Time

[Ben Blakely]

Is there a way to display the time of the 7960 running
firmware 7.4? Im unable to find any information.

 

Thanks,

 

Ben Blakely






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Re: [Asterisk-Users] Calls between Asterisk servers using SIP?What about IAX (got it working w/ IAX but I have questions)

[Gabriel Afana]

Does anybody have any experience with capabilities 
here?  I need to know if IAX is able to handle more than that.  I 
think I might just benchmark this with a bunch of .call files between servers to 
see how they are handled.
 
Any input?
 
- Gabriel Afana
 

  - Original Message - 
  From: 
  Umair Bari 
  
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Tuesday, March 07, 2006 3:30 
  AM
  Subject: Re: [Asterisk-Users] Calls 
  between Asterisk servers using SIP?What about IAX (got it working w/ IAX but I 
  have questions)
  
  Hello Gabriel,
   
  IMHO, using IAX between * servers is a good choice, I dont see any 
  problem in it. Actually I used it for sometime and never encounter any issue, 
  but i had max 5 concurrent connections. 
  regards,
   
  Umair bari 
  On 3/7/06, Gabriel 
  Afana <[EMAIL PROTECTED]> 
  wrote: 
  Hi 
everyone,   I just spend the last two hours trying to get two 
asterisk boxes totransfer calls between eachother using 
SIP.  I dont know why but I *could not* get the calls to 
authenticate!  I think I got everything setup.   
There was Server A and Server B.  I was trying to place a call 
from ausers registered on Server A to a user regsitered on Server 
B.  I setup the registration info for Server A and even had 
Server A registeringsuccessfully to Server B.  However, 
whenever I would hand off the calls fromserver A to Server B, it would 
*always* say it failed to authenticate (passwords did not 
match).  Here was my setup:SERVER A:register => serga:[EMAIL PROTECTED][to_80]username=sergatype=friendsecret=test 
host=216.152.244.81disallow=allallow=ulawuser=phoneusereqphone=yescanreinvite=yesregseconds=0cancallforward=yesdtmfmode=rfc2833disallow=allallow=ulaw 
insecure=verytrunk=yesSERVER 
B:[serga]type=friendusername=sergatrunk=yesnotransfer=yessecret=testcontext=302host=dynamicqualify=yesDIALPLAN 
ON SERVER A: exten => 302,1,Dial(SIP/to_80/[EMAIL PROTECTED],30,r)It 
always says authentication failed.  However I always noticed it 
showedthe user as [EMAIL PROTECTED].  This is 
the extension of the phone I am calling from.  It seems it is 
trying to authenticate the actual phone I amcalling from on Server A, 
and not Server A itself.  Was I doing somethingwrong?I 
tried doing this with IAX and within 5 minutes I had it all 
working!!  I feel it was too easy :-)   However, 
this brings up a big question.IsIAX very reliable for 
this?  I've heard from people that I should not useIAX under 
any condition because it really is not 
veryreliable/thourough/consistant...etc.  I am trying to start 
a VOBB company and will obviosly need a reliable setup.  I am 
thinking to have all phonesregister to the servers via SIP and maybe 
just have all the servers transfercalls between eachother via 
IAX.  Does this sound like a correct setup? - 
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[Asterisk-Users] Zap not installing

[Curt Shaffer]

I have decided to move on from [EMAIL PROTECTED] and start
compiling asterisk myself now. I got a dedicated box and put my X100P in it. I
installed the server version of CentOS 4.2 (2.6.9-22.EL kernel) barebones. The
box is a dell GX270 workstation with 1GB of RAM. I got a fresh copy of
O’Reilly’s Asterisk the future of technology and begun. I
downloaded the zaptel-1.2.4.tar.gz, libpri-1.0.9, and asterisk-1.2.5. I started
compiling the zaptel (make && make install && make clean) when
I try to start zaptel -> /etc/init.d/zaptel start I get the following error:

 

Loading zaptel framework:  FATAL: Module zaptel not
found 

Unable to open /dev/zap/ctl: No such file or directory

 

Below are the only things I have declared in my
/etc/zaptel.conf

 

ks=1

loadzone=us

defaultzone=us

fxoks=1 ( I have tried fxsks=1 as well, because the book had
a section that read the following):

 

"...a physical FXO port will be defined in
configuration with FXS signaling..an FXO card connects to a central office(CO),
which means it will need to behave like a station that use FXS signaling"

 

I tried this both in /etc/udev/rules.d/50-udev.rules and
/etc/udev/rules.d/zaptel.rules (rebooting after each change)

 

Zaptel devices

KERNEL="zapctl",
NAME="zap/ctl"

KERNEL="zaptimer",  
NAME="zap/timer"

KERNEL="zapchannel", NAME="zap/channel"

KERNEL="zappseudo", 
NAME="zap/pseudo"

KERNEL="zap[0-9]*",  NAME="zap/%n"

 

When I run ztcfg I get the following error:

 

line 0: Unable to open master device '/dev/zap/ctl'

 

When I run zttool I get the following error:

 

Unable to open /dev/zap/ctl: No such file or directory

 

I have started from scratch multiple times and I get the
same result. 

 

I get no errors when compiling and the card can be removed
and put back in the old system and work properly. Also Linux does notice the
device when I install and boot into the OS.

 

Any help would be appreciated.

 

Curt

 

 



 



 






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Re: [Asterisk-Users] HW Echo Cancellers

[Matt]

> > Tellabs looks a little too up-scale for what I need :). $1k for a
> > single port orion unit might be worth considering for really stubborn
> > installs though.
> >
>
> Why? they go for around $100.00 on eBay.

What goes for $100 on eBay?  Tellabs?  or Orion?  I can't find any
Orion equipment on eBay.  What model Tellabs am I looking for?
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[Asterisk-Users] Faxing with MFC/r2

[Carlos Chavez]

 I am having a problem when trying to send a receive faxes on an E1
running with unicall on an asterisk 1.2.4 x64 server.  The same server has a
TDM02 card and if I send or receive faxes through there there is usually no
problem.  I am afraid that my customer insists that he wants to use the DID on
the E1 for faxes so I need to fix this. 

 The fax is connected to a Linksys PAP2 adapter but I have also tried
rxfax and I get the same results when trying to use the E1 connection.  Is
there a setting or modification that can be done to unicall?

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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[Asterisk-Users] Softphone for Windows CE 3.0

[Matt]

Hi,
I've found several softphones for Windows Mobile 2003, but does anyone
know of a softphone (or older version of a current softphone) that
will run on Windows CE 3.0?

~ Matt
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[Asterisk-Users] System Design

[Jason Adams]

Hey 
Everyone,
 
We are in the works 
of planning a new * installation for our company.  We have 20 users in our 
main office and 5 users in a remote office a couple of states away.  Our 
call volume for the main office will be anywhere from 5-10 concurrent 
calls.  The remote office will have about 3 heavy users with two users 
making calls occasionally.
 
Right now we have an 
existing PBX.  We have a T-1/PRI coming into the main office and a DSL 
connection at the remote office.  We have a Cisco 2610/PIX 501 at the main 
office a cheesy linksys router at the remote site.
 
We are planning on 
purchasing new Cisco IP phones for everyone.
 
My main question is 
this:  What type of hardware/network design would be best for this 
situation?  Would a full T-1 at the remote site work with a VPN between the 
offices?  Or would a higher bandwidth DSL work with a VPN?  Or should 
we move to a Point-to-Point connection?  What type of hardware would be 
best for the end-to-end communication in regards to QoS?  I know the PIX 
501 doesn't support it.
Would it be best to 
have two * servers in each office or for that call volume at the remote office 
does it make sense?  I was thinking of a Dell Power Edge server with 4GB of 
ram and a dual processor.. is that enough?
 
Sorry for all the 
questions!
 
Jason 
AdamsSumo Systems 57 E. Wilson Bridge 
RdSuite 
200Worthington, 
OH 
43085 
Phone | 614.433.9906 
ext: 102Fax | 
614.433.9931 
E-mail | [EMAIL PROTECTED] 
 
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Re: [Asterisk-Users] Setting Vaaibles

[Dovid Bender]

Figured it out. It was simple  had to add Answer and hangupDovid Bender <[EMAIL PROTECTED]> wrote:  Helo List,First I would like to apologize for my bad spelling aswell as that I did not search the wiki first. I onlyhave email access at the moment.I am having trouble setting both variables and globalvariables thru an extension.I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4with an Xlite softphone. I have two xlite phones ondiffent computers. One logs in as xlite1 and the otheras SNOM.My dial plan is as followsExten => 200,1,Dial(${OnCall},10)Exten => 201,1,Set(>Exten => 202,1,Set(>(I have tried Set and SetGlobalVar).When I use Set I get the following in the CLI-- Executing Set("SIP/snom-a6
 45",
 ">in new stack== Auto fallthrough, cahnnel 'SIP/snom\a645 status is'UNKNOWN'If I dial ext. 201 or 202 I get call failed: 603declined on the xlite phone. When I dail 200 I get anerrorIf I use SetGlobalVar the output from the CLI is-- Executing SetGlobalVar("SIP/snom-24f8"," in new stack= Setting global variable 'OnCall' to 'SIP/SNOM'== Auto fallthrough, channel 'SIP/snom-24f8' status is'UNKNOWN'When I use SetGlobalVar I get the same error in thexlite phone. However when I dial ext. 200 it works.I tried dialing 201 and 202 from both softphones and Igot the same errors.Thanks a lot.Dovid __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easyn
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[Asterisk-Users] No DTMF

[Dovid Bender]

Some one was on my server making changes to my sip.conf files. I am now having trouble with DTMF. No matter what I use (inband,auto,rfc2833) the dtmf tones seem to not come thru. I compared it to the wiki and all the configs seem to be in order.     Here is my sip.conf     [general]disallow=all;allow=g729 ; requires license for g729allow=ulawport = 5060nat=yescontext=from-sipbindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)maxexpirey=4800 ; Maximum expiration for registrationsdefaultexpirey=1800 ; Default expiration for registrationscanreinvite=no ; Allow clients to directly connect if set to yes. Set to no if behind NAT.tos=reliabilitysrvlookup=yes ; Enable DNS SRV lookups on outbound callsvideosupport=no ; Turn on support for SIP videodtmfmode=rfc2833 ;rfc2833 ;inband ;rfc2833 ; DTMF inband need to be set
 here.pedantic=noexternip=..XXX  ;Sip Mediaregister => XX:[EMAIL PROTECTED]/7322761368  [sipmedia6]type=frienduser=XX ;(Phone Number)username=XX ;(Phone Number)fromuser=XX ;(Phone Number)authname=XX ;(Phone Number)secret= ;(SIP Password)host=sip.sipmedia.com disallow=allallow=ulawcontext=ServerHighwayrealm=sip1.xchangetele.comfromdomain=sip.sipmedia.comdtmfmode=rfc2833canreinvite=no insecure=very     Here is my extensions.conf  [general]static=yeswriteprotect=yes  [ServerHighway];Play Server Highway IVR  Exten => s,1,Background(server-highway-ivr)Exten => s,2,Background(blank-file-10)  Exten => 1,1,Ringing()Exten =>
 1,2,Wait(15)Exten => 1,3,Macro(stdexten,9511,9511)Exten => 2,1,Ringing()Exten => 2,2,Wait(15)Exten => 2,3,Macro(stdexten,9512,9512)Exten => 3,1,Ringing()Exten => 3,2,Wait(15)Exten => 3,3,Macro(stdexten,9513,9513)Exten => 4,1,Ringing()Exten => 4,2,Wait(15)Exten => 4,3,Macro(stdexten,9514,9514)Exten => i,1,Background(invalid)Exten => i,2,Goto(s,1)  Exten => t,1,Goto(s,1)  exten => 9,1,Goto(s,1);Extension To Record Main IVR Messageexten => 500,1,Authenticate(XXX)exten => 500,2,Record(ServerHighwayIvr:gsm)
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Re: [Asterisk-Users] Polling Asterisk for Life

[Matt]

> > Are you guys perchance using Local/[EMAIL PROTECTED] in your installations?
> >
> > --
> > Cheers,
> >
> > Matt Riddell
> > ___
> >
>
> Is there a known issue when using the Local/[EMAIL PROTECTED]
>
> thanks,

This is how I would read it.. but yes.. can someone give more
information on this apparently huge bug!
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[Asterisk-Users] Memory Problems

[Dumpolid Exeplish]

Hello,This is not a question directly related to asterisk.I am currently rinning ansterisk on  a Debian server and i just upgraded my memory from 1GB to 2GB. However, my linux OS does not recognise the memory upgrade. The BIOS does, but the Debian Linux refuses to use the entier memory, currently, it registered only 900MB.
Can anyone tell me why thi is and a solution to this??My Debian version is "Linux asterisk 2.6.12.3 #1 Mon Aug 1 19:33:51 WAT 2005 i686 GNU/Linux"The server is currently routing calls from SIP internal users through an E1 card (TE410)
OUTPUT FROM dmesg command009dc00 (usable) BIOS-e820: 0009dc00 - 000a (reserved) BIOS-e820: 000f - 0010 (reserved) BIOS-e820: 0010 - 7fee (usable)
 BIOS-e820: 7fee - 7fee3000 (ACPI NVS) BIOS-e820: 7fee3000 - 7fef (ACPI data) BIOS-e820: 7fef - 7ff0 (reserved) BIOS-e820: fec0 - 0001 (reserved)
Warning only 896MB will be used.Use a HIGHMEM enabled kernel.896MB LOWMEM available.found SMP MP-table at 000f5a20On node 0 totalpages: 229376  DMA zone: 4096 pages, LIFO batch:1
  Normal zone: 225280 pages, LIFO batch:31  HighMem zone: 0 pages, LIFO batch:1-END
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RE: [Asterisk-Users] Receiving Multiple calls on asterisk at home

[Jeff Herring]

Do you have call-limit parameter set to 3 in sip.conf or possibly
sip_additional.conf on AAH?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rolf
Brusletto
Sent: Tuesday, March 07, 2006 1:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Receiving Multiple calls on asterisk at home


All - I've been muddling around with this for a few days now.. and I'm
trying to figure out why I am not receiving more than one phone call on
each polycom 501 phone. I can make more than one phone call out, but not
receive another one in, while on a call. Has anybody seen this behaivior
before, or is there something simple in the config i'm missing, like..
maxcalls.. or something.

Thanks!

Rolf Brusletto

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[Asterisk-Users] impact of qualify=yes

[Damon Estep]

Anyone have any information on the performance impact of
using qualify=yes for hundreds (500ish) of SIP UAs?

 

I have seen tidbits on qualifyspreading=yes, but not enough
to understand what it does. I assume lessens the peak load of qualify sip
options queries?

 

Thx!






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RE: [Asterisk-Users] Receiving Multiple calls on asterisk at home

[Jeff Herring]

Do you have the phone specific config file for the polycom set to something
like this?



  
  
  
  



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rolf
Brusletto
Sent: Tuesday, March 07, 2006 1:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Receiving Multiple calls on asterisk at home


All - I've been muddling around with this for a few days now.. and I'm
trying to figure out why I am not receiving more than one phone call on
each polycom 501 phone. I can make more than one phone call out, but not
receive another one in, while on a call. Has anybody seen this behaivior
before, or is there something simple in the config i'm missing, like..
maxcalls.. or something.

Thanks!

Rolf Brusletto

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Re: [Asterisk-Users] Changing REINVITE status of the channel dynamically

[Luki]

> I'd like to know if it's possible to set the REINVITE on or off dynamically,
> based on the extension being dialed.

Define two peers in sip.conf, one with canreinvite=yes and the second
with canreinvite=no. Then you can route your calls with or without
reinvites depending on the dialed number. Like:

[provider-reinvite]
type=peer
host=external_sip_server.com
canreinvite=yes
...

[provider-noreinvite]
trype=peer
host=external_sip_server.com
canreinvite=no
...

exten => _1[0-4]X0.,1,Dial(SIP/{EXTEN:[EMAIL PROTECTED])
exten => _1[5-9]X0.,1,Dial(SIP/{EXTEN:[EMAIL PROTECTED])

--Luki
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[Asterisk-Users] Putting caller in queue and dialing an extension simultaneously

[Zach A]

Hi,

 

Is it possible to do this in extensions.conf to put a caller
in queue and dial an agent’s extension so that he knows that somebody is
in queue waiting to be answered. This agent will be a remote agent and
extension will dial his cell phone.

 

Thanks

 

Zach A.

 






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[Asterisk-Users] What port mpg123 uses for MoH?

[Zach A]

Hi,

What port does mpg123 uses to play music on when it starts MoH after
asterisk has put called on hold?

Zach A

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RES: [Asterisk-Users] Inserting access codes as prefixes to CID

[Filipe Mordhorst]

There’s the
SetCallerID cmd that you should read about.

 

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

 

It has others links to clarify
your ideas.

 

Tell us if you get
something.

 

 





Filipe Mordhorst  
Brazil-SC











De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de AR Tarzi
Enviada em: domingo, 5 de março de
2006 12:58
Para: Asterisk Users Mailing List
- Non-Commercial Discussion
Assunto: [Asterisk-Users]
Inserting access codes as prefixes to CID



 



When I receive a call from fwd, I'd like to insert a prefix prior to
the caller ID - 1) to be able to look it up in a database of identified
numbers and 2) for the receiver to be able to dial it back.





So what I need is to identify the DID and based on that, insert the
prefix.





 





Any pointers to documentation would be appreciated





 










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[Asterisk-Users] Upgrading Asterisk witk G729 license installed

[Álvaro Palma]

I've an Asterisk 1.2.4 installation, where I've also installed the G729 
codec license. I'd like to upgrade that installation to 1.2.5, but I'm 
not sure if I'll lost the license in the process (and if I'll be able to 
recover it later!!!).


Is there any special consideration I've to keep in mind in this case, or 
should I just run the typical "make + make install" and it will take 
care of keeping the license information?


Thanks a lot for your attention.

--
Atly.
Alvaro Palma

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[Asterisk-Users] PAP2 won't make two g729 calls at the same time

[Warren Burstein]

I have a Linksys PAP2.  Identical setups for the two channels in both 
the unit and in Asterisk.  In particular, both channels enable g729 and 
set it as the preferred codec, and have disallow=all and allow=g729 in 
sip.conf.


If we make a call on one channel, it works (and uses g729), but if we 
make a call on the other channel when the first one is still connected, 
it fails.  We have three g729 licenses, and no others were in use at the 
times this happened, but even if we didn't have enough, how would the 
PAP2 know that?


Here's a good, and a bad INVITE message, from the log file with sip 
debug enabled.  Has anyone seen anything like this?


INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3e521aa
From: PAP 220 ;tag=6b66e68deef168b2o0
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PAP 220 
Expires: 240
User-Agent: Linksys/PAP2-3.1.3(LS)
Content-Length: 246
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 261305180 261305180 IN IP4 192.168.254.44
s=-
c=IN IP4 192.168.254.44
t=0 0
m=audio 16392 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3fda1e15
From: PAP 220 ;tag=b8b86be991749af5o0
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PAP 220 
Expires: 240
User-Agent: Linksys/PAP2-3.1.3(LS)
Content-Length: 267
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 261589835 261589835 IN IP4 192.168.254.44
s=-
c=IN IP4 192.168.254.44
t=0 0
m=audio 16400 RTP/AVP 0 8 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv




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RES: [Asterisk-Users] pap2 Dial plan

[Filipe Mordhorst]

You’re almost right.

The PAP2 has some features
that are factory default. I don’t remember the section in the web
interface, but here’s what you going to do:

 

Find the section that
contains a lot of features name with values like this *56 or *78.

Erase all of them. Letting’
this filled you’ll not be able to implement your asterisk features, cause’
they are conflicting with the (factory defaults) PAP2 commands.

 

About the long time
waiting for start to call, the problem is that the PAP2 waits 10 or 15 (I don’t
remember de default) seconds after a digit is pressed to start the “send”
procedure.

To change these settings,
go to Regional/Control Timer Values/ Interdigit Long Times and change the value
to any other (this is expressed in seconds).

 

Hope it helps.

 

 

Regards,





Filipe Mordhorst

Brazil-SC













De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Giridhar Bandi
Enviada em: terça-feira, 7 de
março de 2006 14:47
Para:
asterisk-users@lists.digium.com
Assunto: [Asterisk-Users] pap2
Dial plan



 

Hi 

i am using pap2 phone adaptors as clients to connect to asterisk server 
i am able to make calls but i cannot access voice mail using phone 
or start recording while call is in progress 

and when i place a call to local sip extension there is a long pause ( 15 sec )

before the call gets dialled 

i assume that the problem would be due to the dial plan in PAP2 

if so please help me changing it 

thanks 
Giridhar Bandi 








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[Asterisk-Users] Location of MeetMe Recordings

[Gavin Adams]

In Asterisk 1.2.4 is love being able to recording conferences. However,
using the default variables, the files are being written to
/var/lib/asterisk/sounds instead of /var/spool/asterisk/meetme.

If I change MEETME_RECORDINGFILE variable to something different in works,
bit I lose the ability to define CONFNO as part of the file name, which is
handy when sorting for users to review. I call meetme using (,r,) so the
conference number is not defined yet.

My /etc/asterisk/asterisk.conf file is set to point to /var/spool/asterisk
for recording related bits, and voicemail and general recordings are being
stored in the appropriate subdirectories. It's only meetme that is going to
a different place.


Regards,

--- Gavin Adams
VP Operations
PARC Inc.

E-mail: [EMAIL PROTECTED]
Office: +1 678.281.6402
   Fax: +1 678.281.6401
Mobile: +1 404.933.8183
 Skype: gadams999

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Re: [Asterisk-Users] grandstream handytone 286 sometimes dials out wrong number

[Giorgio Incantalupo]

Hi Martin,

I have 3 choices on my ATA webpage and I chose SIP INFO:
/Send DTMF: / in-audio via RTP (RFC2833) via SIP INFO

This is the only point I can make changes since it is connected to my 
asterisk box through a TDM400P:
asterisk box <--->TDM400P <-(telephone cable)-> HT-288 <---> LAN <---> 
Internet <---> Messagenet VoIP provider


We examined Messagenet provider logs and, I do not why, we lose 1 call 
on 30 made...our customer loses 1 call on 2 (50%).

We think it is the ATA sending bad DTMF sometime.
Seems strange anybody else but me hadn't had problems like this...I 
found nothing on internet...



TIA

Giorgio Incantalupo


Martin Joseph wrote:


On Mar 6, 2006, at 6:46 AM, Giorgio Incantalupo wrote:


Hi,
I have an asterisk 1.2.1 (on a debian sarge) box with a TDM400P card. 
I connected the TDM400P to a grandstream 286 to use a VoIP provider.
It seems all right except for a little problem: one call every 30 is 
made to a wrong number.

Is there anybody who had the same problem and solved it?

Usually this is DTMF issue? So make sure the extensions and the HT286 
have the correct DTMF config. I have some experience with the HT-488 
FXS and that needed to have dtmfmode=rfc2833 in the extensions and the 
configuration on the HT-488 set the same.


Hope this helps,
Marty

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[Asterisk-Users] pickup last ringing phone

[erkan kolemen]

Hello,I am using pickup, i can pickup an extension from outside of the queue, but i cannot pickup any call comes to queue.queue strategy=ringallWhat is the problem with queue?Is there anyway to pickup last ringing phone?-erkaN
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[Asterisk-Users] Cisco Call Manager SIP trunk + Asterisk

[Chris HARIGA]

Hi,

 

I setup a SIP trunk in a brand new Cisco Call Manager and I
try to place the calls using Asterisk… but I get error:

 

“<-- SIP read from 192.168.11.10:5060:

SIP/2.0 400 Bad Request - 'Malformed/Missing URL'

Via: SIP/2.0/UDP
192.168.10.199:5060;branch=z9hG4bK2e7ca9c9;rport

From: "asterisk"
;tag=as56c7728f

To: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

Content-Length: 0”

 

Question: How I can setup asterisk to get the sip call
without authentication? I check on voip-info.org but I didn’t find a
sip.conf sample L

 

Best regards,

 

Chris HARIGA

 






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