Re: [asterisk-users] Balancing interrupts.

[Matthew J. Roth]

Steve Edwards wrote:
Should I be concerned that cpu1 is servicing only 700,000 interrupts 
from my te410p while cpu3 is servicing almost 90,000,000?


I thought this is what irqbalance was for...

Steve,

It was my experience that irqbalance used smp affinity to bind the 
interrupts from each ethernet device to their own CPU.  This led to 
uneven processor utilization on my Asterisk server, so after some 
research I turned off irqbalance.


If you choose to do so, you'll want to confirm that your kernel has been 
configured to do IRQ balancing.  For more details see:


Asterisk & SMP: Is irqbalance Redundant on 2.6 Kernels? - Resolved 



Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] Poor man's High Availability solution

[FailSafeVOIP]

On Sun, Apr 29, 2007 at 09:06:53PM +0200, Clayton Milos wrote:

Since a PRI is a physical connection as well as a logical one, if you can
get the server to shut down when it has a problem you could put a 4-pole
relay to change the PRI over to the other box.



The ISDN Guard is an excellant product from what I've seen of it,
and you would be well served by it.

We are in the process of releasing (Product is ready, working on sales 
channels)

a somewhat simpler product, the FSV-4PFS.
It will handle two asterisk server redundancy for significantly less cost.

http://www.failsafevoip.com/images/4PFS/FSV-4PFS-Datasheet.pdf

A demo of it in action:
http://www.failsafevoip.com/images/4PFS/FSV-4PFS_Demo.avi


FailSafeVOIP, Inc.
www.failsafevoip.com
Contact: [EMAIL PROTECTED] 


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Re: [asterisk-users] Poor man's High Availability solution

[FailSafeVOIP]

On Sun, Apr 29, 2007 at 09:06:53PM +0200, Clayton Milos wrote:
> Since a PRI is a physical connection as well as a logical one, if you can 
> get the server to shut down when it has a problem you could put a 4-pole 
> relay to change the PRI over to the other box.
>
The ISDN Guard is an excellant product from what I've seen of it,and you would 
be well served by it.We are in the process of releasing (Product is ready, 
working on sales channels) a somewhat simpler product, the FSV-4PFS.  It will 
handle two asterisk server redundancy,for significantly less 
cost.http://www.failsafevoip.com/images/4PFS/FSV-4PFS-Datasheet.pdfA demo of it 
in 
action:http://www.failsafevoip.com/images/4PFS/FSV-4PFS_Demo.aviFailSafeVOIP, 
Inc.www.failsafevoip.comContact for additional info: [EMAIL PROTECTED]___
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Re: [asterisk-users] SIP RealTime Friends

[Remco Post]

Forrest Beck wrote:
> I setup sip realtime.  Is it possible to use a type of friend?  User
> and Peer seem to work fine.
> 

have you tried? If so, what went wrong? (*hint* ;-) )

-- 
Met vriendelijke groeten,

Remco Post

SARA - Reken- en Netwerkdiensten  http://www.sara.nl
High Performance Computing  Tel. +31 20 592 3000Fax. +31 20 668 3167
PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16  B3F6 048A 02BF DC93 94EC

"I really didn't foresee the Internet. But then, neither did the
computer industry. Not that that tells us very much of course - the
computer industry didn't even foresee that the century was going to
end." -- Douglas Adams
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Re: [asterisk-users] Connections rejected in DUNDi requests

[Remco Post]

Chris Bagnall wrote:
> Greetings list,
> 
> Wondering if anyone's come across this before.
> 
> I've configured a couple of our servers with a "privatedundi" context to 
> allow calls to still flow between extensions even if they're registered to 
> different servers . The DUNDi lookups seem to work fine, evidenced by the 
> following on the originating server:
> -- Called private:@/[EMAIL PROTECTED]
> 

shouldn't that be 'private:@/minotaur-201'? I guess you
have a mistake in your dundi mapping

> However, on the destination server, I have the following:
> 
> May  4 03:50:45 NOTICE[1149]: chan_iax2.c:7354 socket_read: Rejected connect 
> attempt from 80.68.80.210, request '[EMAIL PROTECTED]' does not exist
> 
> I then performed the following:
> 
> cronus*CLI> show dialplan privatedundi 
> [ Context 'privatedundi' created by 'pbx_config' ]
>   '_minotaur-2XX' => 1. NoOp(Connected to ${EXTEN})
> [pbx_config]
> 2. Goto(minotaur|${EXTEN:9}|1)[pbx_config]
> 
> Unless I'm missing something, "[EMAIL PROTECTED]" definitely *does* exist. 
> I've tried manually specifying minotaur-201 in full rather than as a pattern 
> match - which works correctly. I'm having exactly the same problem the other 
> way around (origination and target servers reversed).
> 
> What's particularly strange is that other entries in [privatedundi] such as 
> _clienta-2XX, _clientb-2XX are working fine between the same servers.
> 
>  So, what's special about _minotaur-2XX vs. _somethingelse-2XX that causes 
> pattern matching to fail?
> 
> If anyone can shed some light on this I'd be most grateful.
> 
> Regards,
> 
> Chris


-- 
Met vriendelijke groeten,

Remco Post

SARA - Reken- en Netwerkdiensten  http://www.sara.nl
High Performance Computing  Tel. +31 20 592 3000Fax. +31 20 668 3167
PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16  B3F6 048A 02BF DC93 94EC

"I really didn't foresee the Internet. But then, neither did the
computer industry. Not that that tells us very much of course - the
computer industry didn't even foresee that the century was going to
end." -- Douglas Adams
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Re: [asterisk-users] SIP RealTime Friends

[0xception]

yes you can use the type friend

On 5/3/07, Forrest Beck <[EMAIL PROTECTED]> wrote:


I setup sip realtime.  Is it possible to use a type of friend?  User
and Peer seem to work fine.

--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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[asterisk-users] Call interruption

[Andre Wangler]

Hello all

Could someone tell me what happens with running calls when reloading the 
whole asterisk config files? I think SIP-calls are not interrupted because 
of the protocol architecture (signalling vs. media) but what's with other 
kind of calls like h323 or over analogue interfaces? are they interrupted?

I'm quite new with asterisk, so excuse this probably trivial question...

Andre 


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Re: [asterisk-users] OT - robo dialer

[Doug Crompton]

Thanks Chris,

 Looks like a lot of capability. I can deal with Perl but I still was
hoping for something a little more turnkey.

Doug

On Fri, 4 May 2007, Chris Bennett wrote:

> Hi Doug,
>
> > Can anyone suggest a source for a free robot dialer or examples? I need to
> > do some non-commercial auto dialing using Asterisk. Simple phone numbers
> > in a file, line by line format.
> >
> > I found one called AstAutoDiaker but I was not able to get it to work and
> > it appears to not be supported - no email response from author.
>
> If you are comfortable with Perl (programming language), or have
> access to somebody who is, you could get something working with the
> CPAN module Net::SIP ..
>   http://search.cpan.org/~sullr/Net-SIP-0.26/
>
> Amongst other things, it can be a 'phone' with the module
> Net::SIP::Endpoint.
>   http://search.cpan.org/~sullr/Net-SIP-0.26/lib/Net/SIP/Endpoint.pod
>
> I havn't used it myself as yet but intend to in the next few months.
> If your requirements are for unattended dialling then you might find
> this option more flexible since you can make it do exactly what you
> want.
>
> Good luck with it anyhow! :)
>
> Regards,
>
> Chris Bennett
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"Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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Re: [asterisk-users] Large dial plans and variables

[Ira]

At 09:29 AM 5/3/2007, you wrote:

I guess we are. I propose we add functions or procedures!


How about we just start over and make it a proper language?

Not that I don't love it, but the hoops that one has to jump through 
because it's not cause my hair to disappear at an ever faster rate.


Ira 


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Re: [asterisk-users] OT - robo dialer

[Chris Bennett]

Hi Doug,

> Can anyone suggest a source for a free robot dialer or examples? I need to
> do some non-commercial auto dialing using Asterisk. Simple phone numbers
> in a file, line by line format.
> 
> I found one called AstAutoDiaker but I was not able to get it to work and
> it appears to not be supported - no email response from author.

If you are comfortable with Perl (programming language), or have
access to somebody who is, you could get something working with the
CPAN module Net::SIP ..
  http://search.cpan.org/~sullr/Net-SIP-0.26/

Amongst other things, it can be a 'phone' with the module
Net::SIP::Endpoint. 
  http://search.cpan.org/~sullr/Net-SIP-0.26/lib/Net/SIP/Endpoint.pod

I havn't used it myself as yet but intend to in the next few months.
If your requirements are for unattended dialling then you might find
this option more flexible since you can make it do exactly what you
want.

Good luck with it anyhow! :)

Regards,

Chris Bennett
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[asterisk-users] OT - robo dialer

[Doug Crompton]

Can anyone suggest a source for a free robot dialer or examples? I need to
do some non-commercial auto dialing using Asterisk. Simple phone numbers
in a file, line by line format.

I found one called AstAutoDiaker but I was not able to get it to work and
it appears to not be supported - no email response from author.


Doug

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RE: [asterisk-users] Called party identification - where to takecalledname?

[Yehavi Bourvine +972-8-9489444]

>>Yehavi wrote:
>> >  I am trying to apply the "called party identification"
>> > patch (patch 8824) and managed to make it work with a
>> > static data. Where do I take the name of the called person
>> > (the "equivalent" of CALLERID, but the other way...)?

Asnwering myself: I am using realtime extensions, so I've added call to
MYSQL() application to get the called user callerid field.

  __Yehavi:
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Re: [asterisk-users] T1/E1 Configuration

[Forrest Beck]

You can remove the extra context in zapata.

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 98
(remove this line) context=default

The way zapata works is you define options for the channels, then
specify the channels the options are for.  Context=default will just
add this option to the next channel below.  Also, If you don't want to
use a group on the channel, just eliminate the line all together
instead of group=0

Does ztcfg run without errors?

What does "zap show channels" display in the asterisk CLI?  Are you
running asterisk as root using asterisk -vvc and it still fails to
load?

Does lsmod show the driver modules loaded?



On 5/3/07, Nitesh Divecha <[EMAIL PROTECTED]> wrote:

Hello All,

Can anyone please post their working T1/E1 configuration...

Both '/etc/zaptel.conf' and '/etc/asterisk/zapata.conf'. I believe if
you run 'genzaptelconf' it created '/etc/asterisk/zapata-channels.conf',
so please post that one also.

Here is my configuration which is failing Asterisk to load... I have two
cards TE405P and TDM400P: -
===
/etc/zaptel.conf
===
# T1 Configuration
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

span=2,1,0,esf,b8zs
bchan=25-47
dchan=48

span=3,1,0,esf,b8zs
bchan=49-71
dchan=72

span=4,1,0,esf,b8zs
bchan=73-95
dchan=96

fxsks=97
fxsks=98
fxsks=99
fxsks=100

# Global data

loadzone= us
defaultzone = us


/etc/asterisk/zapata-channels.conf

group = 1
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 1-23

group = 2
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 25-47

group = 3
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 49-71

group = 4
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 73-95

signalling=fxs_ks
callerid=asreceived
group=0
context=from-zaptel
channel => 97
; context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 98
context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 99
context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 100
context=default


Thanking in advance...

Cheers,
Nitesh


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--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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[asterisk-users] T1/E1 Configuration

[Nitesh Divecha]

Hello All,

Can anyone please post their working T1/E1 configuration...

Both '/etc/zaptel.conf' and '/etc/asterisk/zapata.conf'. I believe if 
you run 'genzaptelconf' it created '/etc/asterisk/zapata-channels.conf', 
so please post that one also.


Here is my configuration which is failing Asterisk to load... I have two 
cards TE405P and TDM400P: -

===
/etc/zaptel.conf
===
# T1 Configuration
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

span=2,1,0,esf,b8zs
bchan=25-47
dchan=48

span=3,1,0,esf,b8zs
bchan=49-71
dchan=72

span=4,1,0,esf,b8zs
bchan=73-95
dchan=96

fxsks=97
fxsks=98
fxsks=99
fxsks=100

# Global data

loadzone= us
defaultzone = us


/etc/asterisk/zapata-channels.conf

group = 1
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 1-23

group = 2
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 25-47

group = 3
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 49-71

group = 4
switchtype = national
signalling = pri_cpe
context = from-zaptel
channel => 73-95

signalling=fxs_ks
callerid=asreceived
group=0
context=from-zaptel
channel => 97
; context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 98
context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 99
context=default

signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 100
context=default


Thanking in advance...

Cheers,
Nitesh


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[asterisk-users] Connections rejected in DUNDi requests

[Chris Bagnall]

Greetings list,

Wondering if anyone's come across this before.

I've configured a couple of our servers with a "privatedundi" context to allow 
calls to still flow between extensions even if they're registered to different 
servers . The DUNDi lookups seem to work fine, evidenced by the following on 
the originating server:
-- Called private:@/[EMAIL PROTECTED]

However, on the destination server, I have the following:

May  4 03:50:45 NOTICE[1149]: chan_iax2.c:7354 socket_read: Rejected connect 
attempt from 80.68.80.210, request '[EMAIL PROTECTED]' does not exist

I then performed the following:

cronus*CLI> show dialplan privatedundi 
[ Context 'privatedundi' created by 'pbx_config' ]
  '_minotaur-2XX' => 1. NoOp(Connected to ${EXTEN})[pbx_config]
2. Goto(minotaur|${EXTEN:9}|1)[pbx_config]

Unless I'm missing something, "[EMAIL PROTECTED]" definitely *does* exist. I've 
tried manually specifying minotaur-201 in full rather than as a pattern match - 
which works correctly. I'm having exactly the same problem the other way around 
(origination and target servers reversed).

What's particularly strange is that other entries in [privatedundi] such as 
_clienta-2XX, _clientb-2XX are working fine between the same servers.

 So, what's special about _minotaur-2XX vs. _somethingelse-2XX that causes 
pattern matching to fail?

If anyone can shed some light on this I'd be most grateful.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons


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RE: [asterisk-users] Unable to Execute System Command From DialPlan

[Salvatore Giudice]

Could it be permissions? What uid is your Asterisk running under and what
are the perms for your sounds directory? sounds is this by default on my
server since I built from source:

drwxr-xr-x  13 root root 110592 Apr 14 01:13 sounds

 

Try putting:

 

/bin/mkdir -p /var/lib/asterisk/sounds/1234 > /tmp/logfile 2>&1

 

See if that generates a log for you at least.

 

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of vcomp
Sent: Thursday, May 03, 2007 8:38 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Unable to Execute System Command From DialPlan

 

Hello,

I have scoured the mailing lists and forums to no avail. Does anyone have
tips on how to use the system command within a dialplan (1.2.7.1). I am very
familiar with dialplan scripting but new to the system command.

I am attempting to create a directory. I put both of the lines below in my
dialplan but neither executes, although they do not generate errors. The
first line was added just for kicks to see if system is working properly.

exten => s,n,System(/bin/pwd > location.out)
exten => s,n,System(/bin/mkdir -p /var/lib/asterisk/sounds/1234)

Any assistance would be greatly appreciated.

Thanks,

Victor

 

P.S.  I received a suggestion to change System(/bin/pwd... to
System(!/bin/pwd ... but it did not work, with or without a space.

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RE: [asterisk-users] Asterisk 1.4 and Cisco Phones 7940

[Salvatore Giudice]

The features you mentioned will work fine, but you'll need to also have to
maintain a tftp server to provision the phones and ensure a quick boot. In
my experience, the only annoying thing about sip loads on Cisco phones is
that they don't support sidecars for admin.

 

http://www.voip-info.org/wiki-Setup+SiP+on+7940+-+7960

 

Make sure you remove any callmanager related info from your DHCP scope
before you deploy Asterisk if they previously had  a callmanager installed.
When completing these types of conversions, you run the risk of the phones
going to an unprovisioned state if they start trying to access a callmanager
that has been removed from the network. It sucks to get called back to a job
a few weeks later when the customer's phone gets whacked after it was
unplugged and rebooted.

 

 

Good luck

 

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez
Sent: Thursday, May 03, 2007 8:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.4 and Cisco Phones 7940

 

I have read the wiki and several other internet documents. Can anyone make a
comment as to what kind of functionality will you loose if you use Cisco
7940 phones with asterisk 1.4
things like: MWI, call transfer, conference,etc,etc. 
I have a customer with 6 of those phones that he like to use with the
asteirsk PBX.

thanks,


-- 

Erick Perez
 

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RE: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!

[Salvatore Giudice]

That phone sounds like a real pain in the ass. Besides, it looks a little
junky from the photos. I guess you really can't complain too much about a
$150 light office SIP phone. BTW, how does the phone 'feel'? When you pick
up the handset do you immediately get the feeling that it's a cheap phone?



--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Garstang
Sent: Thursday, May 03, 2007 6:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk-Polycom HEPPP

The polycom lets you do either attended or unattended transfers. If you 
want an unattended transfer, you press the 'blind' soft key. It's been a 
few months since I've looked at this, so a bit fuzzy on the details.

Jason Adams wrote:
>
> Isn't that the function of an attended transfer? User3 hears User1 to 
> see if they want to take the call or not. User1 should then hit the 
> transfer key again to finalize the call.
>
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Jim Suber
> *Sent:* Thursday, May 03, 2007 12:54 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Asterisk-Polycom HEPPP
>
> PBX:
>
> Asterisk 1.4
>
> Phones:
>
> PSTN phone connected to TDM400
>
> X-Ten Lite
>
> Polycom 430
>
> Scenario
>
> Polycom 430 = User1
>
> User2 calls User1(Polycom 430) asks to be transfered to User3
>
> User1 does an attended transfer using the trnsfr button on the polycom
>
> User2 is placed in music-on-hold
>
> User3s phone rings.
>
> (So far so good Right?)
>
> User3 picks up the phone to answer User2 only to find that he is 
> talking to User1
>
> User2 is stuck in music-on-hold. FOREVER!
>
> The other two phones work exactly as they should using the # key
>
> Using the # key on the Polycom only allow the dialing of 1 number 
> before Alice announces
>
> That there is no such extension.
>
> HELP
>
> Thanks in advance
>
> Jim
>
> 
>
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RE: [asterisk-users] Get Asterisk to redirect a SIP INVITE

[Salvatore Giudice]

I don't think you can do that. You can easily issue a 302 with something
like SER or OpenSER. I believe the only thing Asterisk can do is receive a
call on the initial URI and open a channel to the destination and connect
them. Media could pass directly between those two points but your Asterisk
box would still have to participate in the signaling. Think of Asterisk as a
B2BUA instead of a SIP call router/response system.

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU
Sent: Thursday, May 03, 2007 6:18 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Get Asterisk to redirect a SIP INVITE

>From: "CSB" <[EMAIL PROTECTED]>
>Date: Thu, 3 May 2007 21:51:02 +1200
>
>I want to get Asterisk to redirect an incoming SIP INVITE to another SIP 
>URI. I was looking at the Transfer application but it seems to

You may want to elaborate the requirement.  How is the incoming INVITE 
initiated?  Is the originator a user in your system?  Does the other URI 
represent a peer? etc.

Yuan Liu

>be broken (http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9483). 
>Is there an alternative way to do this on Asterisk 1.2.18?
>
>Regards
>
>Cameron


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[asterisk-users] [Announce] Web-MeetMe 3.0.2 and 2.2.2 Released

[Dan Austin]

Basic bug fix releases

Both have updates to app_cbmysql to be thread-safe,
reconnect to the database in case of timeout and to
detect missing/mis-configured conference app/conference
participant counting apps.

The last one has caused Asterisk to crash.  Now
If it does not find MeetMe or MeetMeCount (the
defaults) it posts a warning and exits back to
the dialplan.

Web-MeetMe 2.2.2 also has an a couple of small PHP
Updates (2.2.1 shipped with a copy of one PHP file
from the 3.X tree that broke the conference monitoring
page)

The new releases can be found at:  
http://sourceforge.net/projects/web-meetme/


Thanks go out to the users and testers who found these
issues and who kept after me until I found and fixed them.

Special thanks to hadefix on SF for identifying the 
threading issue and providing hints about the fix.

Thanks,
The Web-MeetMe development team
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[asterisk-users] Unable to Execute System Command From DialPlan

[vcomp]

Hello,

I have scoured the mailing lists and forums to no avail. Does anyone have
tips on how to use the system command within a dialplan (1.2.7.1). I am very
familiar with dialplan scripting but new to the system command.

I am attempting to create a directory. I put both of the lines below in my
dialplan but neither executes, although they do not generate errors. The
first line was added just for kicks to see if system is working properly.

exten => s,n,System(/bin/pwd > location.out)
exten => s,n,System(/bin/mkdir -p /var/lib/asterisk/sounds/1234)

Any assistance would be greatly appreciated.

Thanks,

Victor


P.S.  I received a suggestion to change System(/bin/pwd... to
System(!/bin/pwd ... but it did not work, with or without a space.
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[asterisk-users] SIP RealTime Friends

[Forrest Beck]

I setup sip realtime.  Is it possible to use a type of friend?  User
and Peer seem to work fine.

--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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[asterisk-users] SIP peer / Maximum retries exceeded on transmission

[chris]

Hi Everyone,

I was hoping someone might know why I am experiencing a problem with
Asterisk logging the event:

[May  3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on 
transmission [EMAIL PROTECTED] for seqno 669371069 (Critical Response)

This is happening after:
  - call is setup, 2 way audio
  - call can function correctly for up to 5 minutes, with the external
provider re-inviting every 1 minute

When the problem happens
  - external peer re-invites asterisk
  - asterisk sends 200 OK
  - external peer sends ACK
  - asterisk retransmits 200 OK
  - external peer sends ack
  - ..
  - asterisk retransmits 200 OK (Retransmitting #6)
  - external peer sends ack
  - Asterisk logs the above message about maximum retries exceeded,
and sends BYE to the inside SIP UA.


The network configuration is as follows:
  phone <--> alternative SIP server <--> Asterisk <-NAT-> External peer

The alternative SIP server is not a B2BUA, just SIP proxy.  Now,
sometimes a call can work without any problems, but not as often as
when the above symptoms are experienced.

The references I've found online about this type of problem suggest
NAT as being the culprit, but in this case, Asterisk is logging it's
reception of the ACK but deciding to ignore it and retransmit the
200 OK anyhow.  I'm guessing in other cases people suspect is' NAT
because they believe SIP isn't getting back trhough after a period of
time.

I was using 1.4.2, but found this changelog today for 1.4.3:

ftp://ftp.digium.com/pub/asterisk/releases/ChangeLog-1.4.3

2006-09-30 16:12 + [r44068-44078] Paul Cadach <[EMAIL PROTECTED]>
  * channels/chan_sip.c: Found some buggy SIP clients (phones Planet
VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which
sends ACK not on OK message only (when remote party answers) but
on RINGING message too, so when we send 200 OK message, we get
unidentified ACK message (because INVITE acknowledged on RINGING
message already), so 200 OK retransmits within its retransmission
interval then call gets dropped. If someone else knows how to
provide workaround for such cases, please, fix it in correct way.
Thanks to ssh from #asteriskru for provide access to his box to
study and fix this case.

I've upgraded to 1.4.4 but the problem still persists.  The above
changelog doesn't sound exactly like what I"m experiencing but maybe
it's related.

Attached is my sip.conf, extensions.conf, and (debug = 10) logs for
one example.  I don't know what else might be needed to help anyone
assist me in this problem - let me know if I missed something.

It *feels* like an Asterisk bug but maybe a SIP expert can spot the
problem in signalling/RFC conformance..

Thanks in advance,

Chris Bennett
[general]
context=default 
allowoverlap=no 
bindport=5060   
bindaddr=0.0.0.0
srvlookup=yes   

domain=proxy.myhostname

disallow=all
allow=alaw
sipdebug = yes  
recordhistory=yes   
dumphistory=yes 
register => @sip.externalpeer.com
  
externhost=proxy.myhostname

localnet=192.168.0.0/255.255.0.0
localnet=10.0.0.0/255.0.0.0 
localnet=172.16.0.0/12  
localnet=169.254.0.0/255.255.0.0 
nat=never   
canreinvite=no  

[authentication]
auth = @sip.externalpeer.com

[provider]
type=peer
username=
secret=
fromuser=
fromdomain=sip.externalpeer.com
host=sip.externalpeer.com
nat=never
canreinvite=no

[]
type=friend
username=
secret=
host=dynamic
context=tutorial
nat=never   
insecure=invite
qualify=yes
[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
CONSOLE=Console/dsp 

[tutorial]
exten => _XXX.,1,Dial(SIP/[EMAIL PROTECTED],,r)


asterisk.logs.example1.txt.bz2
Description: BZip2 compressed data
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[asterisk-users] Asterisk 1.4 and Cisco Phones 7940

[Erick Perez]

I have read the wiki and several other internet documents. Can anyone make a
comment as to what kind of functionality will you loose if you use Cisco
7940 phones with asterisk 1.4
things like: MWI, call transfer, conference,etc,etc.
I have a customer with 6 of those phones that he like to use with the
asteirsk PBX.

thanks,


--

Erick Perez

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RE: [asterisk-users] OT: USB T1/E1 Interface?

[Michael Collins]

> How about PCMCIA and 2 T1/E1/J1 interfaces?
> http://www.utelsystems.com/instruments/hardware/pist-2mp-pro.php
> 

Nice, but less portable than a USB - most desktops and servers don't
have a PCMCIA slot.  I'm thinking about the 'U' in USB.  If I'm going to
have something be portable, why not make it work with as many different
systems as possible?

-MC
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RE: [asterisk-users] VoiceXML + Nuance

[wendell hamilton]

I've done considerable work with the voxeo Prophecy platform, and it's
been successful, albeit challenging at times. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Rousse
Sent: Thursday, May 03, 2007 2:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] VoiceXML + Nuance

Hello,

Is there anyone who has ever done a setup of VoiceXML combined with some

licenses from Nuance for the ASR/TTS engine within Asterisk ?
I'm currently working with VoiceGenie, for the VoiceXML + ASR/TTS 
engine, but we are having a couple of issues which I guess are caused by

VoiceGenie.

If there's an alternative, it would be very interesting for us.

Thanks,

-- 
Eric Rousse
System Administrator
514.380.2992
450.655.1001
1.888.641.5800

Telmatik inc.
204 Montarville, suite 250
Boucherville, QC, Canada
J4B 6S2

www.telmatik.com


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This message is confidential. It may also be privileged or otherwise protected 
by work product immunity or other legal rules. If you have received it by 
mistake, please let us know by e-mail reply and delete it from your system; you 
may not copy this message or disclose its contents to anyone. Please send us by 
fax any message containing deadlines as incoming e-mails are not screened for 
response deadlines. The integrity and security of this message cannot be 
guaranteed on the Internet.

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Re: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!

[Eric \"ManxPower\" Wieling]

Brandon Kruse wrote:

Do not use the transfer key on the Polycom.

Use /etc/asterisk/features.conf and setup blind and attended transfers for 
asterisk.


It just works better in my opinion.


In my opinion DTMF transfers are an ugly hack for phones that are too 
cheap to have a transfer button.

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[asterisk-users] Restricting membership to a queue?

[Fernando Gleiser]

I'm having some weird problem with asterisk (trixbox).

I've set it up as a pbx for a small call center with 4
main queues. It usually works great, until for some
reason I still don't fully understand the extension
the users dial in to register to a queue becomes a
queue member itself :/

So what happens after that is when all the operators
are busy taking calls and a new call gets in, the
caller gets the message asking them to register into
the queue (because the wrong extension gets the call
since it's treated as a free agent)

Here's an example:

Let's suppose the queue is "123" and the users dial in
to "123*" in order to become members of such a queue.
When that weird thing happens you see things like
this:

asterisk1*CLI> show queue 123
123  has 0 calls (max unlimited) in
'leastrecent' strategy (4s holdtime), W:0, C:4, A:0,
SL:0.0% within 0s
   Members:
  Local/[EMAIL PROTECTED]/n (dynamic) (Unknown) 
  Local/[EMAIL PROTECTED]/n (dynamic) (Unknown)
   No Callers

Is there a way to prevent this from happening? Is
there a way to set up a whitelist so only good known
extensions can become a member?

Any help would be greatly apreciated


Fer


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RE: [asterisk-users] OT: USB T1/E1 Interface?

[Michael Collins]

> Maybe we could interest the guy thats building his own open telco
> hardware:
> 
>   http://www.rowetel.com/ucasterisk/pr1.html
> 
> He seems to have the skills :)
> 

I'm working on it right now! :)

-MC
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Re: [asterisk-users] IAX Trunk

[Steve Kennedy]

On Thu, May 03, 2007 at 03:23:16PM -0300, Ronaldo wrote:

> OK Steve,
> Just one more question. Using this configuration can I make more than 
> one call at the same time?

The whole point of trunking is to support multiple "calls" down the same
IAX trunk (well actually down the same packets).


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation

[Andreas van dem Helge]

Here's something similar for Linux: http://sourceforge.net/projects/vgps/

Note I do not support nor endorse Voicepulse. Actually let's get it
straight, I detest Voicepulse.

On 5/3/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:

Mats Karlsson wrote:
> Take a look here:
> http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html

Ugh. This is a Win32 app, isn't it?

-Stephen-
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RE: [asterisk-users] Called party identification - where to takecalledname?

[Yuan LIU]

From: "Dan Austin" <[EMAIL PROTECTED]>
Date: Thu, 3 May 2007 10:01:25 -0700

Yehavi wrote:
>  I am trying to apply the "called party identification"
> patch (patch 8824) and managed to make it work with a
> static data. Where do I take the name of the called person
> (the "equivalent" of CALLERID, but the other way...)?
Short answer is that you cannot.

Longer answer is that it is possible, but requires new
functionality to be added to the core and a new API call
be added that can check if the called party is a local
endpoint and retrieve the caller-id values.


It will depend on actual application.  For some small sites, manually 
setting up an AstDB family should suffice.  This can even be semi automated.


Yuan Liu


At least that was what I found when working on the patch.
If anyone knows a way to lookup a peer/friend from the
dialplan and collect such details, it would be possible to
use the existing patch without any more changes in the core.

> BTW, one note to the above patch: To make it work the device
> should have the parameter sendrpid set to true.

Dan



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Re: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!

[Doug Garstang]

The polycom lets you do either attended or unattended transfers. If you 
want an unattended transfer, you press the 'blind' soft key. It's been a 
few months since I've looked at this, so a bit fuzzy on the details.


Jason Adams wrote:


Isn’t that the function of an attended transfer? User3 hears User1 to 
see if they want to take the call or not. User1 should then hit the 
transfer key again to finalize the call.


*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Jim Suber

*Sent:* Thursday, May 03, 2007 12:54 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Asterisk-Polycom HEPPP

PBX:

Asterisk 1.4

Phones:

PSTN phone connected to TDM400

X-Ten Lite

Polycom 430

Scenario

Polycom 430 = User1

User2 calls User1(Polycom 430) asks to be transfered to User3

User1 does an attended transfer using the trnsfr button on the polycom

User2 is placed in music-on-hold

User3s phone rings.

(So far so good Right?)

User3 picks up the phone to answer User2 only to find that he is 
talking to User1


User2 is stuck in music-on-hold. FOREVER!

The other two phones work exactly as they should using the # key

Using the # key on the Polycom only allow the dialing of 1 number 
before Alice announces


That there is no such extension.

HELP

Thanks in advance

Jim



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RE: [asterisk-users] Get Asterisk to redirect a SIP INVITE

[Yuan LIU]

From: "CSB" <[EMAIL PROTECTED]>
Date: Thu, 3 May 2007 21:51:02 +1200

I want to get Asterisk to redirect an incoming SIP INVITE to another SIP 
URI. I was looking at the Transfer application but it seems to


You may want to elaborate the requirement.  How is the incoming INVITE 
initiated?  Is the originator a user in your system?  Does the other URI 
represent a peer? etc.


Yuan Liu

be broken (http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9483). 
Is there an alternative way to do this on Asterisk 1.2.18?


Regards

Cameron



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Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation

[Tom Rymes]

On May 3, 2007, at 12:20 PM, Stephen Bosch wrote:


Mats Karlsson wrote:

Take a look here:
http://www.voip.com.sg/voip_products/ 
voip_ip_phone_provisioning_tool.html


Ugh. This is a Win32 app, isn't it?


Wow,

The guy makes a useful application and provides it to the community  
for free and you have the cojones to bitch and moan b/c it's a  
windows app? Talk about looking a gift horse in the mouth!


Tom
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Re: [asterisk-users] Reinvite after DTMF?

[Yuan LIU]

From: "Wilson Pickett" <[EMAIL PROTECTED]>
Date: Thu, 3 May 2007 09:19:25 +0200

On 5/2/07, Yuan LIU <[EMAIL PROTECTED]> wrote:

>From: "Wilson Pickett" <[EMAIL PROTECTED]>
>Date: Wed, 2 May 2007 15:30:21 +0200
>
>Is there a way to do the following scenario?
>
>1) my asterisk box receives an incoming call from a toll free number
>provider such as nufone, voicepulse, etc.
>2) It then dials a number  via SIP and outputs a  DTMF  sequence.

At this point, I assume, the destination SIP has not been invited?  The
purpose of the DTMF is either determine which SIP destination to invite or
to perform some other dial plan functions.

>ok, that part we do every day.
>
>3) After DTMF though, is it possible to get the two SIP channels
>(original SIP caller plus SIP called) hooked together and have my pbx
>no longer in the call at all?
>
>tia

If the above is true, then there shouldn't be a problem if all other
conditions for reinvite are satisfied, because Asterisk will only execute
Dial at this point, and that Dial could follow with reinvite. (I assume 
that

the original SIP caller is in fact the toll free provider.)


So what is in the dialplan once the DTMF is sent? The two channels are
already bridged, how can asterisk then bow out? I don't see a way,


Maybe I missed something here.  In my understanding, the only parties in the 
call at DTMF stage are the originator and Asterisk.  The destination is not 
in the picture yet.  Is this correct?  What is the purpose of the said DTMF 
sequence?  Do you have a sample dial plan?


Yuan Liu


but
I thought I'd ask if someone else did?



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RE: [asterisk-users] IAX Trunk

[Salvatore Giudice]

Yes of course. If you want to limit it, I think you have to set
'incominglimit' and/or 'outgoinglimit'.

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo
Sent: Thursday, May 03, 2007 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX Trunk

OK Steve,

Just one more question. Using this configuration can I make more than 
one call at the same time?

Thanks.

Steve Kennedy wrote:
> On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote:
>
>   
>> Can you suggest me any documentation about using IAX trunking?
>> Thank you.
>> 
>
> There are examples in the iax.conf files I think, but basically just put
> something like
>
> [iax-toremote]
> type=friend
> username=whatever
> secret=somesecret
> auth=plaintext
> host=somewhere.com
> peercontext=some-context
> qualify=yes
> trunk=yes
>
> then you dial with Dial(iax2/iax-toremote/number)
>
>
> Steve
>
>   

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RE: [asterisk-users] Linseed

[Salvatore Giudice]

Interesting play for maybe a smartphone,  pda , or internet enabled
microwave oven .

 

It's a little expensive for that price. I just picked up a new half-ebx
Advantech pcm-9381 with 1GB DDR 333 and a Pentium M 745 1.8Ghz cpu for $250.
That's smaller than the evaluation board they picture in that article. Plus,
it features 2x18 bit LVDS and pc/104. On the downside it will likely
generate more heat. I need to find a 2GB compact flash card and a 10.4"
touch screen lcd so I can start building a phone prototype. I just wish I
could find a chassis and a small power supply for it. MBOX was making them
for Advantech gear, but I can't seem to find one for this board model. =(

 

I think these kind of soft chips may end up being popular in PDA's, but only
if manufacturers can get them for less than $15. The majority of $400 - 500
cell phones on the market cost less than $30 to manufacture with the lcd
being the most expensive component.

 

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Thursday, May 03, 2007 2:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Linseed

 

http://www.linuxdevices.com/news/NS3013985136.html

 

Ok so who's going to be the first to install Asterisk on it?

 

 

Regards,

Dean Collins
Cognation Pty Ltd
  [EMAIL PROTECTED]
+1-212-203-4357 Ph

  Call Button
 

 

 

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[asterisk-users] Error messages in console : sipsock_read: SIP MESSAGE JUST IGNORED:

[Don Fletcher]

I have googled this, and have come up with blanks.  Someone told me it 
was a bug, and I read in one forum, that these errors were because of 
poorly configured .conf files, but, it gave no clue as to what are to 
look in.


Any help would be appreciated.

Thanks

Don Fletcher
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[asterisk-users] VoiceXML + Nuance

[Eric Rousse]

Hello,

Is there anyone who has ever done a setup of VoiceXML combined with some 
licenses from Nuance for the ASR/TTS engine within Asterisk ?
I'm currently working with VoiceGenie, for the VoiceXML + ASR/TTS 
engine, but we are having a couple of issues which I guess are caused by 
VoiceGenie.


If there's an alternative, it would be very interesting for us.

Thanks,

--
Eric Rousse
System Administrator
514.380.2992
450.655.1001
1.888.641.5800

Telmatik inc.
204 Montarville, suite 250
Boucherville, QC, Canada
J4B 6S2

www.telmatik.com


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Re: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!

[Brandon Kruse]

Do not use the transfer key on the Polycom.

Use /etc/asterisk/features.conf and setup blind and attended transfers for 
asterisk.


It just works better in my opinion.

-bk

- Original Message -
From: "Jim Suber" <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: Thursday, May 3, 2007 9:53:58 AM (GMT-0800) America/Tijuana
Subject: [asterisk-users] Asterisk-PolycomHEPPP

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RE: [asterisk-users] OT: USB T1/E1 Interface?

[Michael Collins]

> http://www.gl.com/laptopt1.html
> 

That's the first item I found when I did a Google search.  It prompted
me to ask the question - is there something more generic than this?  I
was quoted a price of US$8000 for this, which is way more than I'm
willing to pay for an item which would be used for tinkering, learning,
FOSS-type fun, etc.

-MC
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RE: [asterisk-users] IAX and SETLANGUAGE delays

[Jonathan Barratt]

Hi Joel,

As I thought, your _XX5 wildcard is causing the problem. If these are
all in the same context then anytime someone presses a digit like "1"
asterisk has to wait 6 seconds to see if they're going to end up
entering "115, 125" or anything else that might match the pattern. 

Just rearrange your extensions so there's no potential crossover between
your wildcards and your explicit extensions and the problem should go
away. I don't know what reason you need the _XX5 entries for so I'll
leave it up to you to find the best arrangement...

Good luck!
Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Hill
Sent: Wednesday, May 02, 2007 11:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] IAX and SETLANGUAGE delays

Hi Jonathon,

Here's the relevant part (I hope!)

exten => _XX5,1,Answer
exten => _XX5,2,Set(COUNT=0)
exten => _XX5,3,Wait(1)
exten => _XX5,4,Background(ST1000-001-1) ;english
exten => _XX5,5,Background(STS1000-001-1);spanish
exten => _XX5,6,Background(STG1000-001-1);greek
exten => _XX5,7,Background(STI1000-001-1);italian
exten => _XX5,8,WaitExten(1)

exten => 1,1,Set(LANGUAGE()=english); english
exten => 1,2,Goto(STE1050,s,1)
exten => 2,1,Set(LANGUAGE()=spanish);spanish
exten => 2,2,Goto(STE1050,s,1)
exten => 3,1,Set(LANGUAGE()=greek)  ;greek
exten => 3,2,Goto(STE1050,s,1)
exten => 4,1,Set(LANGUAGE()=italian);italian
exten => 4,2,Goto(STE1050,s,1)

exten => 7,1,Goto(incoming,_XX5,1)

exten => 8,1,Set(COUNT=$[${COUNT} + 1]) ; after pressing 8 2 times then
goes to consultant
exten => 8,2,GotoIf($[${COUNT} = 3]?4:3)
exten => 8,3,Goto(incoming,_XX5,4)
exten => 8,4,Goto(STE1850,s,1)

exten => 9,1,Playback(STE-thankyou) ; hangs up after plays thank you
for calling msg

exten => 0,1,Goto(STE1850,s,1)  ; sends to consultant


Cheers,

Joel

On Wed, 2007-05-02 at 22:49 -0400, Jonathan Barratt wrote:
> Hi Joel,
> 
> 6 seconds sounds suspiciously like Asterisk's dialplan timeout value.
> Perhaps you have a wildcard extension that it's waiting to match
> against. Either post the relevant section of dial plan or send it to
me
> off-list, as you prefer, and we'll see what we can find...
> 
> Best,
> Jonathan
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Joel
Hill
> Sent: Wednesday, May 02, 2007 9:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] IAX and SETLANGUAGE delays
> 
>   Hi all,
> 
> I'm having some trouble. I've got an IVR with 4 different languages
> using the SETLANGUAGE command and I'm getting a 6 second delay when I
> make the first selection, after that all is fine. There's nothing in
my
> dial plan that I can find that would be causing it. And the delay is
> driving me nuts!
>  I have an IAX connection from a provider coming in. Could this be the
> cause? Has anyone experienced anything similar.
> 
> Thanks,
> 
> Joel.
> 
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[asterisk-users] Balancing interrupts.

[Steve Edwards]

I see the following on one of my new servers:

-ts10::sedwards:~$ cat /proc/interrupts
CPU0   CPU1   CPU2   CPU3
   0:29790452988620   87780075   87779501IO-APIC-edge  timer
   1:  1  3  2  3IO-APIC-edge  i8042
   8:  0  0  0  1IO-APIC-edge  rtc
   9:  0  0  0  0   IO-APIC-level  acpi
  12: 13 13 15 17IO-APIC-edge  i8042
  14:909  53719 814685 763055IO-APIC-edge  ide0
169:  0  0  0  0   IO-APIC-level  uhci_hcd, 
uhci_hcd
177:  0  0  0  0   IO-APIC-level  uhci_hcd
185:  0  0  0  0   IO-APIC-level  uhci_hcd
193:  0  0  0  0   IO-APIC-level  ehci_hcd
201:  665255896107123122   IO-APIC-level  eth0
217:   4298   2495 238514 233273   IO-APIC-level  cciss0
225:4611916 681023   84732445   89903138   IO-APIC-level  wct4xxp
NMI:  0  0  0  0
LOC:  181534588  181534654  181534653  181534652
ERR:  0
MIS:  0

-ts10::sedwards:~$ ps -e | grep bal
  2633 ?00:00:00 irqbalance

Should I be concerned that cpu1 is servicing only 700,000 interrupts from 
my te410p while cpu3 is servicing almost 90,000,000?


I thought this is what irqbalance was for...

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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RE: [asterisk-users] OT: USB T1/E1 Interface?

[John Treble]

How about PCMCIA and 2 T1/E1/J1 interfaces?
http://www.utelsystems.com/instruments/hardware/pist-2mp-pro.php

The linux drivers for this card support SS7 + Wireshark.


John Treble
Ottawa, Ontario, Canada


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[asterisk-users] Re: Asterisk-1.4 with agent snmp

[Everton Goularth]

Hi Matt Gibson,

First of all, congratulations for your how-to..
I selected res_snmp in my asterisk by the "menu select" and I complied 
the asterisk-1.4. After the compilation, I edited the res_snmp.conf and 
turn on snmp in asterisk as you said in your how-to.


In my last tests I received "Warning: Failed to connect to the agentx 
master agent ([NIL]):" in the asterisk cli.


And when I try the command that you say in your how-to (after to copy 
asterisk-mib.txt and digium-mib.txt to /usr/share/snmp/mibs):


# export MIBS=+ASTERISK-MIB
# snmpwalk -v 1 -c private localhost asterisk

I receive nothing.

What about my configuration??

/etc/snmp/snmpd.conf

com2sec local   localhost   private
com2sec mynet   192.168.0.0/24  public
com2sec public  default public
group   mygroup v1  mynet
group   mygroup v2c mynet
group   mygroup usm mynet
group   local   v1  local
group   local   v2c local
group   local   usm local
group   public  v1  public
group   public  v2c public
group   public  usm public
view allincluded  .1   80
access  mygroup ""  any   noauthexact  allall  all
access  public  ""  any   noauthexact  all   allall
access  local   ""  any   noauthexact  allall  all
master agentx
agentXPerms 0660 0550 root root
agentXSocket tcp:localhost:161


/etc/asterisk/res_snmp.conf

[general]
subagent = yes
enabled = yes

And I use this command to run snmpd and snmptrapd:

snmpd -a -d -V -x 127.0.0.1
snmptrapd -a -Lf /var/log/snmptrapd.log


Thank's for your help

Everton Goularth
Uberlandia - MG - Brasil


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Re: [asterisk-users] OT: USB T1/E1 Interface?

[Steve Edwards]

Way cool product.

Way too cool for my neighborhood -- the interface box is $7k. Software 
will set you back $3k to $30k. And then I would have no clue what to do 
with it.


Maybe we could interest the guy thats building his own open telco 
hardware:


http://www.rowetel.com/ucasterisk/pr1.html

He seems to have the skills :)

On Thu, 3 May 2007, Jorge Mendoza wrote:


http://www.gl.com/laptopt1.html

Jorge

Michael Collins wrote:

Why? There used to be a saying 'usb is for mice, firewire is for men',
though USB has grown a bit in bandwidth since then, it is still not


very


well suited for a high sustained bandwidth. NOw T1/E1 is not that big,


I


suspect a lack of demand. Havng a E1 termintae in your laptop is quite
useless, and a server usually has plenty of slots (if not, buy a


bigger


server ;-).




Just for fun.  I'm a telecom geek and having a USB T1 interface would be
a fun toy to tinker with.  Besides, it might lead to some useful
products.

-MC
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Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Linksys SPA3012 inbound FXO problems

[Drew Gibson]

http://www.sipura.com/support/index.htm

lenz wrote:


Thanks a lot, that did the trick! Wish there was an half-decent manual 
on the site at least :-(

l.


In data Thu, 03 May 2007 18:34:34 +0200, Dave Cotton 
<[EMAIL PROTECTED]> ha scritto:



On Thu, 2007-05-03 at 17:56 +0200, lenz wrote:

Hello list,
hope someone can help me - I'm going crazy using the FXO port a 
SPA3012.
I would like the SPA 3012 to act as a simple FXO port to an 
Asterisk, that

is, once it detects a call, it should simply send it over to the local
Asterisk server. No intelligent routing, PIN, anything else

I configured it like this:

PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: yes
PSTN Caller Auth Method: none
PSTN Ring Thru Line 1: no
PSTN Caller Default DP: 1

Then I configured the dialplan #1 as:
Dial Plan 1: (S0<:@gw1>)


And that's where it started to go wrong.

(S0<:[EMAIL PROTECTED]:5060>)

will do what you want.






--Home of QueueMetrics - http://queuemetrics.com

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--
Drew Gibson

Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com

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[asterisk-users] Double DTMF digits

[Ken Leland III]

When dtmfmode is set to inband for SIP, and i originate a call from sip 
out to the PSTN, I can hear the DTMF digit twice in the audio stream. 
Once very briefly and once for normal duration.


Our Theory: While Asterisk is parsing the DTMF, for a fraction of a 
second, while the end user generated DTMF is being detected, the DTMF is 
passed inband. Once the DTMF is detected Asterisk silences it and 
regenerates it. Sensitive machines like auto attendants pick up both the 
brief end user generated tone as well as the full length asterisk 
generated tone and ultimately perceive each digit twice.


Is anyone else experiencing this?

I have reproduced this in an environment
   * with one asterisk server that is both the feature server and the 
media gateway, and is timing off of network T1s
   * with two servers, one feature server (timing off of ztdummy) and 
one media gateway (timing off of network T1s) using IAX as the inter 
asterisk protocol


It is pretty easy to reproduce:
-Dial a PSTN number(like your cell) from a sip phone using inband DTMF, 
and configured in asterisk sip.conf with dtmfmode=inband.

-Answer the PSTN end.
-Press and hold a digit on the sip phone. On the PSTN phone you will 
hear a very brief, end user generated, tone.
-Let go of the digit on the sip phone. On the PSTN phone you will hear 
the asterisk generated tone.


Can anyone else hear the brief initial tone?  Any help is greatly 
appreciated!


There is a good wiki article on DTMF:
http://www.voip-info.org/wiki/view/Asterisk+DTMF

and there has been a thread about it on this mailing list:
http://lists.digium.com/pipermail/asterisk-users/2007-March/181461.html

We have opened a bug report:
http://bugs.digium.com/view.php?id=9382

Some System Specifications:
Asterisk 1.2.14
Digium Wildcard TE405P/TE410P
CentOS release 4.4 (Final)

--
Ken W. Leland  III
Engineering
[EMAIL PROTECTED]
Monmouth Telecom
10 Drs. James Parker Blvd., Suite 110
Red Bank, NJ  07701
732-704-1000  X283
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RE: [asterisk-users] Answer A Ringing Queue By Dialing An Extension

[Brandon Comouche]

This would work - I had never noticed this feature. It is very useful
now that I have looked in to it.

I would still like to find a way to directly pick up the Queue if there
is one. Any other thoughts out there?

--
Thanks,
  Brandon Comouche

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Bagnall
Sent: Thursday, May 03, 2007 6:55 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Answer A Ringing Queue By Dialing An
Extension

Stick the lobby phones into a call group and put your other phones in
that pickup group. Then you can hit *8 to pick up those calls (or, if
you have speed dial/BLF/softkeys, program one of those as *8 for an
immediately accessible button).

In sip.conf for the lobby phones:
callgroup=1

in sip.conf for the other phones:
pickupgroup=1

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons


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Re: [asterisk-users] Poor man's High Availability solution

[Stephen Davies]

On 29/04/07, Noah Miller <[EMAIL PROTECTED]> wrote:

> > I've heard of a device that acts as a failover for a PRI line so you
> > can plug a PRI into two different devices and have the PRI failover if
> > one device fails.  Unfortunately nothing like this is commercially
> > available today.
> Sounds like the ISDNguard:
> http://www.junghanns.net/en/ISDNguard_produkt.html

Aha!  Thank You!  I've wanted something like this for quite some time.
 A question:  Does this require BRIStuff?


Yes, but no.

Installing BRIstuff gets you the heartbeat stuff.

But it took me 5 minutes to extract that from BRIstuff and use it by itself.

Steve
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RE: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!

[Jason Adams]

Isn't that the function of an attended transfer?  User3 hears User1 to
see if they want to take the call or not.  User1 should then hit the
transfer key again to finalize the call.

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Suber
Sent: Thursday, May 03, 2007 12:54 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk-Polycom HEPPP

 

PBX:

Asterisk 1.4

 

Phones:

PSTN phone connected to TDM400

X-Ten Lite

Polycom 430

 

Scenario

Polycom 430 = User1

 

User2 calls User1(Polycom 430)  asks to be transfered to User3

 

User1 does an attended transfer using the trnsfr button on the polycom

User2 is placed in music-on-hold

User3s phone rings.

(So far so good Right?)

 

User3 picks up the phone to answer User2 only to find that he is talking
to User1

User2 is stuck in music-on-hold. FOREVER!

 

The other two phones work exactly as they should using the # key

Using the # key on the Polycom only allow the dialing of 1 number before
Alice announces

That there is no such extension.

 

HELP

 

Thanks in advance

Jim

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Re: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!

[Doug Lytle]

Jim Suber wrote:


PBX:

Asterisk 1.4

 


Phones:

PSTN phone connected to TDM400

X-Ten Lite

Polycom 430

 


Scenario

Polycom 430 = User1

 


User2 calls User1(Polycom 430)  asks to be transfered to User3

 


User1 does an attended transfer using the trnsfr button on the polycom



I'm not sure about the 430, but on the 501, you'd do a transfer, talk to 
user3, if user3 wants the call, you'd press transfer again to complete 
the call.  If not, you'd press the resume to get the call back.


Doug


--

Ben Franklin quote:

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


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Re: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!

[Stephen Bosch]

Whoa. Calm down.

Jim Suber wrote:
> PBX:
> 
> Asterisk 1.4
> 
>  
> 
> Phones:
> 
> PSTN phone connected to TDM400
> 
> X-Ten Lite
> 
> Polycom 430
> 
>  
> 
> Scenario
> 
> Polycom 430 = User1
> 
>  
> 
> User2 calls User1(Polycom 430)  asks to be transfered to User3
> 
>  
> 
> User1 does an attended transfer using the trnsfr button on the polycom
> 
> User2 is placed in music-on-hold
> 
> User3s phone rings.
> 
> (So far so good Right?)
> 
>  
> 
> User3 picks up the phone to answer User2 only to find that he is talking
> to User1
> 
> User2 is stuck in music-on-hold. FOREVER!

That's because you didn't finish transferring User2. EVER!

I can tell that you haven't bothered to read the Polycom user's guide,
which is a bit annoying, but I will do you this one favour.

Attended transfers on the Polycom phones work like this:

1. User A calls you.
2. You answer the phone.
3. You press 'Transfer' -- at this point User A hears music -- and dial
User B, then press 'Send'
4. You will hear User B's line ring. YOU AREN'T DONE YET.
5. Now press 'Transfer' again. NOW you're done. Hang up.

Note -- you are not done with a transfer procedure until your phone is
in the idle state again. That is, once you've transferred a call, the
call will disappear from the phone's display. If you still see User A's
call on your phone's display, something is wrong.

Now that I have helped you, go read the Polycom documentation. The list
likes to help people who have tried to help themselves first.

-Stephen-
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RE: [asterisk-users] Semi-OT: useful things to do with XML browsersinphones

[Dean Collins]

Neat, not something I would be interested in but I can certainly see how
people would use that.

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Dan Austin
> Sent: Thursday, 3 May 2007 1:06 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Semi-OT: useful things to do with XML
> browsersinphones
> 
> Chris wrote:
> > It seems that more and more phones these days are
> > coming with XML mini-browsers. I'd like to have a
> > go at developing something useful to use on them,
> > but in all honesty, most of our customers use their
> > phones to make and take calls and very little else.
> 
> > So I'm open to suggestions.
> 
> > What useful applications are you developing for these
> > mini-browsers? What sort of things do your customers
> > want to use on them?
> 
> I've been planning to write to app for joining scheduled
> conferences.  It would be bundled with the Web-MeetMe
> suite.  Users of the app would see a list of conferences
> scheduled for the current time and have one-button access
> to the conference (assuming no PINs)
> 

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Re: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!

[Gordon Henderson]

On Thu, 3 May 2007, Jim Suber wrote:


PBX:

Asterisk 1.4

Phones:

PSTN phone connected to TDM400

X-Ten Lite

Polycom 430

Scenario

Polycom 430 = User1

User2 calls User1(Polycom 430)  asks to be transfered to User3

User1 does an attended transfer using the trnsfr button on the polycom

User2 is placed in music-on-hold

User3s phone rings.

(So far so good Right?)


Yes ...


User3 picks up the phone to answer User2 only to find that he is talking to
User1


This is as expected. It's an ATTENDED transfer -

  User 2 (external?) calls in. U1 answers.
  User 1 has placed User 2 on hold
  User 1 calls U3
  User 1 speaks to U3, asks U3 if they want to accept the call from U2.
 and if yes, then U1 has to do whatever is neccessary to transfer U2
to U1.
  If no, then U1 has to retrieve U2 and say "sorry they are not at their
desk" (or whatever)

Maybe you're confusing it wuth UNattended transfer? I've no knowlege of 
the Polycoms, but on the Grandstreams, for unattended xfer, you hit the 
transfer button, dial the extension, then hang up - U2 gets ringing and 
U1's phone rings. (Attended transfer is slightly more complicated)


Gordon
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Re: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!

[Bruce Reeves]

Jim,

What happens in your first senario is an attended transfer, after User1 and
3 have initiadted their call, User1 should press transfer again to complete
the transfer. At which point User1 will be disconnected and Users 2 & 3 will
talk.

The second issue is the limit of digits and is likely due to a very short
timeout in features.conf, check the entry transferdigittimeout.

On 5/3/07, Jim Suber <[EMAIL PROTECTED]> wrote:


 PBX:

Asterisk 1.4



Phones:

PSTN phone connected to TDM400

X-Ten Lite

Polycom 430



Scenario

Polycom 430 = User1



User2 calls User1(Polycom 430)  asks to be transfered to User3



User1 does an attended transfer using the trnsfr button on the polycom

User2 is placed in music-on-hold

User3s phone rings.

(So far so good Right?)



User3 picks up the phone to answer User2 only to find that he is talking
to User1

User2 is stuck in music-on-hold. FOREVER!



The other two phones work exactly as they should using the # key

Using the # key on the Polycom only allow the dialing of 1 number before
Alice announces

That there is no such extension.



HELP



Thanks in advance

Jim

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--
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Nortex Networks
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[asterisk-users] FXO recommendation

[Kyle Gordon]

Hi all,

With the gamut of FXO cards out there, I'm looking for a recommendation for 
home use. I have a nicely working Asterisk 1.4 system that just requires an 
FXO card to connect my NTL PSTN to it. My previous X101P clone seems to have 
kicked the bucket. 

Any suggestions would be greatly appreciated.

Regards

Kyle
-- 
Kyle Gordon
[EMAIL PROTECTED]
http://lodge.glasgownet.com
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RE : [asterisk-users] Wildcard TE410P problem

[f6hqz-m]

Autocorrection mode :

pri_cpe / pri_net rather than TE / NT  ;-)

-Message d'origine-
De : Francois BERGERET [mailto:[EMAIL PROTECTED] De la part de
'[EMAIL PROTECTED]'
Envoyé : jeudi 3 mai 2007 21:03
À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : RE : [asterisk-users] Wildcard TE410P problem


Hi Alexander and the list,

Have you well checked your E1 cable ?
Sometime, you must use a crossed E1 cable (not an Ethernet one)... Check
also without the crc check.

How is your zapata.conf file ?
Have you checked with a loop (crossed E1 cable) between two spans (one in TE
the second in NT, of course) ?

Best Regards,
Francois BERGERET,
France.

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RE : [asterisk-users] Wildcard TE410P problem

[f6hqz-m]

Hi Alexander and the list,

Have you well checked your E1 cable ?
Sometime, you must use a crossed E1 cable (not an Ethernet one)...
Check also without the crc check.

How is your zapata.conf file ?
Have you checked with a loop (crossed E1 cable) between two spans (one in TE
the second in NT, of course) ?

Best Regards,
Francois BERGERET,
France.

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[asterisk-users] Strange noise - Polycom

[Rob Schall]

I'm not sure if this is a problem with our polycom 501 phones or with a
setting in asterisk.

When you set the "forward" option on the phone and have it point to an
outside number (a cell phone) we see the following problem... The call
does forward, but while its doing so and while its ringing, you hear
this irritating loud pulsing sound. When we call from a non-polycom
phone and dial into the phone with the forward, this noise isn't there.

I couldn't find a polycom mailing list, otherwise I'd try this question
in there as well. Has anyone ever seen a problem like this?
Rob
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[asterisk-users] Linseed

[Dean Collins]

http://www.linuxdevices.com/news/NS3013985136.html

 

Ok so who's going to be the first to install Asterisk on it?

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph

    
 
 

 

 

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Re: [asterisk-users] IAX Trunk

[Ronaldo]

OK Steve,

Just one more question. Using this configuration can I make more than 
one call at the same time?


Thanks.

Steve Kennedy wrote:

On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote:

  

Can you suggest me any documentation about using IAX trunking?
Thank you.



There are examples in the iax.conf files I think, but basically just put
something like

[iax-toremote]
type=friend
username=whatever
secret=somesecret
auth=plaintext
host=somewhere.com
peercontext=some-context
qualify=yes
trunk=yes

then you dial with Dial(iax2/iax-toremote/number)


Steve

  


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Re: [asterisk-users] Linksys SPA3012 inbound FXO problems

[lenz]

Thanks a lot, that did the trick! Wish there was an half-decent manual on  
the site at least :-(

l.


In data Thu, 03 May 2007 18:34:34 +0200, Dave Cotton  
<[EMAIL PROTECTED]> ha scritto:



On Thu, 2007-05-03 at 17:56 +0200, lenz wrote:

Hello list,
hope someone can help me - I'm going crazy using the FXO port a SPA3012.
I would like the SPA 3012 to act as a simple FXO port to an Asterisk,  
that

is, once it detects a call, it should simply send it over to the local
Asterisk server. No intelligent routing, PIN, anything else

I configured it like this:

PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: yes
PSTN Caller Auth Method: none
PSTN Ring Thru Line 1: no
PSTN Caller Default DP: 1

Then I configured the dialplan #1 as:
Dial Plan 1: (S0<:@gw1>)


And that's where it started to go wrong.

(S0<:[EMAIL PROTECTED]:5060>)

will do what you want.






--
Home of QueueMetrics - http://queuemetrics.com

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RE: [asterisk-users] OT: USB T1/E1 Interface?

[Michael Collins]

> I can well understand the idea of having USB T1 adapters since that
way
> you can colocate 1U Asterisk systems  ;-) which at least doubles you
> density in a rack...
> 
> Frank

I'm glad I asked the question!  I was just thinking to myself that it
would be cool to have a USB T1 adapter so that I could tinker, but you
guys have already come up with several real-world applications!

I think I will research this some more and let you all know if anything
interesting pops up.

-MC
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Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation

[Erik Anderson]

On 5/3/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:


Ugh. This is a Win32 app, isn't it?


Yup.
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[asterisk-users] ast_parse_allow_disallow: Cannot disallow unknown format ''

[Dima]

Hello, everyone. I've installed asterisk SVN-branch-1.4-r62942 and every
time I reload asterisk I get this in CLI:

-- Reloading module 'app_playback.so' (Sound File Playback Application)
[May  3 20:04:26] NOTICE[13892]: app_playback.c:455 reload: Reloading
say.conf
  == Parsing '/etc/asterisk/say.conf': Found
[May  3 20:04:26] WARNING[13879]: frame.c:1289 ast_parse_allow_disallow:
Cannot disallow unknown format ''
[May  3 20:04:26] WARNING[13879]: frame.c:1289 ast_parse_allow_disallow:
Cannot disallow unknown format ''
[May  3 20:04:26] WARNING[13879]: frame.c:1289 ast_parse_allow_disallow:
Cannot disallow unknown format ''
[May  3 20:04:27] WARNING[13879]: frame.c:1289 ast_parse_allow_disallow:
Cannot disallow unknown format ''

Is that something to worry about?
Thanks

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Re: [asterisk-users] OT: USB T1/E1 Interface?

[Jorge Mendoza]

http://www.gl.com/laptopt1.html

Jorge

Michael Collins wrote:

Why? There used to be a saying 'usb is for mice, firewire is for men',
though USB has grown a bit in bandwidth since then, it is still not


very
  

well suited for a high sustained bandwidth. NOw T1/E1 is not that big,


I
  

suspect a lack of demand. Havng a E1 termintae in your laptop is quite
useless, and a server usually has plenty of slots (if not, buy a


bigger
  

server ;-).




Just for fun.  I'm a telecom geek and having a USB T1 interface would be
a fun toy to tinker with.  Besides, it might lead to some useful
products.

-MC
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Re: [asterisk-users] OT: USB T1/E1 Interface?

[Steve Edwards]

On Thu, 3 May 2007, [EMAIL PROTECTED] wrote:


Why? There used to be a saying 'usb is for mice, firewire is for men',
though USB has grown a bit in bandwidth since then, it is still not very
well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I
suspect a lack of demand. Havng a E1 termintae in your laptop is quite
useless, and a server usually has plenty of slots (if not, buy a bigger
server ;-).


I can well understand the idea of having USB T1 adapters since that way
you can colocate 1U Asterisk systems  ;-) which at least doubles you
density in a rack...


) Impress the hell out of a client -- "Our PBX smoked and this guy is
running our telecoms on his laptop!"

) Inline/passthrough diagnosis and monitoring -- kind of like a network
sniffer.

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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RE: [asterisk-users] IAX Trunk

[Salvatore Giudice]

Good luck. Try these.

http://www.voip-info.org/wiki-IAX

http://www.voip-info.org/wiki-IAX+versus+SIP

http://www.voip-info.org/wiki/view/IAX+encryption

http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf

http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels



--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo
Sent: Thursday, May 03, 2007 12:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX Trunk

Hi Bruce,

Can you suggest me any documentation about using IAX truking?
Thank you.

Ronaldo.

Bruce Reeves wrote:
> Yes it is.
>
> On 5/3/07, *Ronaldo* <[EMAIL PROTECTED] 
> > wrote:
>
> Hi all,
>
> Is it possible to have something like this:
>
> SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone
>
> I want a IAX trunk between two asterisks and on each tip I have SIP
> clients that need to talk to each other.
>
> Thansk.
>
> Ronaldo
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>
>
>
> -- 
> Bruce Reeves
> Nortex Networks
> 
>
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Re: [asterisk-users] chan_sip seems to be hanging

[Jaswinder Singh]

try soft hangup 

On 02/05/07, Ken Williams <[EMAIL PROTECTED]> wrote:



I posted about this problem last week and thought it was a combination of
SIP/ZAP causing issues in Asterisk.  Since then I've realized it's only the
SIP channel that's hanging.  When this happens a call can still come in and
hit the IVR, but no one can connect to the server from a SIP client.

I tried reloading chan_sip.so today when this occurred, and I tried
unloading chan_sip.so but was told the channel was in use.  How can I clear
SIP connections?  With ZAP channels I can use ZAP DESTROY CHANNEL, but I
don't see the equivalent for SIP.

Any suggestions for tracking down what's causing SIP to hang?  My only
option as it stands is to shutdown asterisk & restart it, I included a piece
of the log last week and am willing to do so again if needed.  If I can see
which SIP channels the server thinks are open when the channel hangs I'm
hoping this will allow me to find if it's a common phone or perhaps some
dialplan logic gone bad.

Thanks,
Ken
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Re: [asterisk-users] zttranscode crashes server

[Tzafrir Cohen]

On Thu, May 03, 2007 at 10:52:51AM -0400, Forrest Beck wrote:
> So is anyone not using the zaptel init script to load modules?  

The fix I mentioned was about unloading rather than about loading.

> Anyone
> using modules.conf?  How an I load them at boot without using the init
> script?  Do I just remove --ignore-install from modprobe?

If you use the init.d script, you don;t need any special "install"
commands: just use the init.d script, and it will run ztcfg when
everything is loaded.

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

[Bruce Reeves]

The wiki has a decent page about it.

http://www.voip-info.org/wiki-IAX

What you are trying to setup sounds simple enoug, you mainly will have an
extension or pattern match that executes a dial command  from box A to box B
and passes the remote extension.

On 5/3/07, Ronaldo <[EMAIL PROTECTED]> wrote:


Hi Bruce,

Can you suggest me any documentation about using IAX truking?
Thank you.

Ronaldo.

Bruce Reeves wrote:
> Yes it is.
>
> On 5/3/07, *Ronaldo* <[EMAIL PROTECTED]
> > wrote:
>
> Hi all,
>
> Is it possible to have something like this:
>
> SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)->
SoftPhone
>
> I want a IAX trunk between two asterisks and on each tip I have SIP
> clients that need to talk to each other.
>
> Thansk.
>
> Ronaldo
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>
>
>
> --
> Bruce Reeves
> Nortex Networks
> 
>
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RE: [asterisk-users] Semi-OT: useful things to do with XML browsers inphones

[Dan Austin]

Chris wrote:
> It seems that more and more phones these days are 
> coming with XML mini-browsers. I'd like to have a 
> go at developing something useful to use on them, 
> but in all honesty, most of our customers use their 
> phones to make and take calls and very little else.

> So I'm open to suggestions.

> What useful applications are you developing for these
> mini-browsers? What sort of things do your customers 
> want to use on them?

I've been planning to write to app for joining scheduled
conferences.  It would be bundled with the Web-MeetMe
suite.  Users of the app would see a list of conferences
scheduled for the current time and have one-button access
to the conference (assuming no PINs)


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[asterisk-users] Asterisk-Polycom HELLLLPPP!!!!

[Jim Suber]

PBX:

Asterisk 1.4

 

Phones:

PSTN phone connected to TDM400

X-Ten Lite

Polycom 430

 

Scenario

Polycom 430 = User1

 

User2 calls User1(Polycom 430)  asks to be transfered to User3

 

User1 does an attended transfer using the trnsfr button on the polycom

User2 is placed in music-on-hold

User3s phone rings.

(So far so good Right?)

 

User3 picks up the phone to answer User2 only to find that he is talking to
User1

User2 is stuck in music-on-hold. FOREVER!

 

The other two phones work exactly as they should using the # key

Using the # key on the Polycom only allow the dialing of 1 number before
Alice announces

That there is no such extension.

 

HELP

 

Thanks in advance

Jim

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RE: [asterisk-users] Called party identification - where to take calledname?

[Dan Austin]

Yehavi wrote:
>  I am trying to apply the "called party identification"
> patch (patch 8824) and managed to make it work with a 
> static data. Where do I take the name of the called person
> (the "equivalent" of CALLERID, but the other way...)?
Short answer is that you cannot.

Longer answer is that it is possible, but requires new
functionality to be added to the core and a new API call
be added that can check if the called party is a local 
endpoint and retrieve the caller-id values.

At least that was what I found when working on the patch.
If anyone knows a way to lookup a peer/friend from the
dialplan and collect such details, it would be possible to
use the existing patch without any more changes in the core.

> BTW, one note to the above patch: To make it work the device
> should have the parameter sendrpid set to true.

Dan
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Re: [asterisk-users] using Playback() to play a random sound file

[Steve Edwards]

Steve Edwards wrote:
On Tue, 1 May 2007, Jay Austad wrote:


I've got a directory under /var/lib/asterisk/sounds which contains a bunch
of sound files.  I would like to call the Playback command to play the
files, but I need it to select a file to play randomly.  Is there any way
to do this?


I do this with an AGI.


On Wed, 2 May 2007, dave cantera wrote:


here is a way that I solved a similar problem...  have a shell script that
runs and indexes all the files in the directory into an ascii flat file with
a format of
 filename
0001 directory/tt-weasels
0002 directory/tt-monkeys

in your dialplan use the rand() to pick a number, pass it to the shell script
as an arg[], then the shells script grep()'s and cut()'s the filename puts it
in a db varaible, the dialplan picks it up and plays it...  as you can see, I
haven't done it yet :) but, in theory it works...  you could skip the
dialplan rand() and just use linux rand based on the minutes or seconds value
for current time...

you don't have to zero fill the index either, I seem to like nicely formated
files, they are easier for humans to read.
daveC


Sounds like a lot of effort to avoid writing an AGI. If you have the 
skills to write the script described above, you have the skills to write 
an AGI -- you can write AGI's in shell scripts, btw.


AGI's accept stuff from Asterisk on stdin and send stuff back to Asterisk 
on stdout -- very simple and elegant actually. Take your script and 
rewrite the "reading arguments" bits to read from stdin and change the 
"write db" bits to write to stdout (set a channel variable) and you have 
an AGI and a much cleaner dialplan.


I write AGI's in C for speed and flexibility. No interpreter (bash, perl, 
php, etc.) to fire up, full access to anything you want to do.


In C, I call ftw() ("ftw - traverse (walk) a file tree"). If I get more 
than 1 file, I choose one randomly.


Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Autologoff

[Ed Nuñez]

I am having an issue with the autologoff fuction in agents.conf

 

I am running Asterisk BE and I am testing with agent 1656.  I log in, and then 
make a call into the queue.  The agent's phone rings, and if I answer it, all's 
fine/  I am trying to have this agent automatically be logged off if he does 
not answer the queue callback within 5 seconds, however the agents extension 
keeps ringing until the call eventually goes to the extension's voice mail, 
which I am also trying to avoid.

 

Here is my agents.conf

 

[general]

 

persistentagents=yes

 

 

[agents]

 

autologoff=5

multiplelogin=no

recordagencalls=yes

monitor-join=yes

createlink=yes

updatecdr=yes

musiconhold=>default

recordformat=wav49

urlprefix=http://64.211.222.226/calls/

savecallsin=/var/www/html/calls

 

agent => 1650,1650,Tareq Tujjar

agent => 1656,1656,Ed Nuñez

 

 

Here is my queues.conf

 

[general]

persistentmembers=yes

 

 

[noi-cust-serv-spanish]

strategy = leastrecent

announce-frequency = 90

announce-holdtime = yes

announce-round-seconds = 10

timeout=180

monitor-format=wav49

monitor-join=yes

joinwhenempty = strict

leavewhenempty = yes

musiconhold = default

eventwhencalled = yes

queue-youarenext = queue-youarenext;   ("You are now first in 
line.")

queue-thereare = queue-thereare;   ("There are")

queue-callswaiting = queue-callswaiting;   ("calls waiting.")

queue-holdtime = queue-holdtime;   ("The current est. 
holdtime is")

queue-minutes = queue-minutes  ;   ("minutes.")

queue-seconds = queue-seconds  ;   ("seconds.")

queue-thankyou = queue-thankyou;   ("Thank you for your 
patience.")

queue-lessthan = queue-less-than   ;   ("less than")

 

member => Agent/1656

 

 

 

 

 

 

 

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RE: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released

[Dan Austin]

Ondrej wrote:
> I finally got some time to test the SVN branches and 
> here are my comments:
Cool.

>>> One thing that does not work for sure - I had some problems to
>>> terminate the running conference from within the web page - I 
>>> just clicked the button and nothing happened.
>>> 
>> This is likely a manager.conf security issue, but it could be
>> a problem in the php code.  I just tested branches/3.0 and
>> trunk against 1.4.1 and it worked as expected.
>> If you set core verbose to 10 and click on 'End Now' the console
>> should display a message like this:
>>  app_meetme.c:941 meetme_cmd: Cmdline: 41251|k|1
>>
>> At this point this would be a topic better suited for the support
>> forums on SF.
>>   
> I browsed the php sources trying to understand and from what 
> I see, the "End Now" button does nothing else than kicking all
> attendees - not exactly what I would expect. I would expect this
>  action to terminate the conference immediately so, it could be
> seen in the "Past conferences" list. Also, some javascript popup
> dialog confirming this action would be nice. The same is valid for
> the "Extend" button - it works, but from the user prospective, 
> nothing happens - I would expect some dialog box like "The
> conference # has been extended by 10 minutes". This is the 
> only missing piece, I would say - thanks :-)

Oh!  Those are great ideas and fairly easy to add.
I'm about to be offline for two weeks, and need to get updated
releases of 2.X and 3.X out before I go.  Your ideas are now
on the ToDo list and I'll try to get them integrated and
released by mid-June.

Dan
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Re: [asterisk-users] IAX Trunk

[Gordon Henderson]

On Thu, 3 May 2007, Ronaldo wrote:


Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP clients 
that need to talk to each other.


Absolutely.

Have a look at this: http://astrecipes.net/index.php?n=204

Gordon
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Re: [asterisk-users] IAX Trunk

[Steve Kennedy]

On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote:

> Can you suggest me any documentation about using IAX trunking?
> Thank you.

There are examples in the iax.conf files I think, but basically just put
something like

[iax-toremote]
type=friend
username=whatever
secret=somesecret
auth=plaintext
host=somewhere.com
peercontext=some-context
qualify=yes
trunk=yes

then you dial with Dial(iax2/iax-toremote/number)


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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Re: [asterisk-users] CDR(accountcode) empty in * 1.4.4 (for local chan)

[Steve Murphy]

On Thu, 2007-05-03 at 16:47 +0500, Rizwan Hisham wrote:
> Hi all,
> i just updated to asterisk 1.4.4 from 1.4.2. i was doing this to
> forward an unanswered call in 1.4.2
> 
> exten=> 1,1,Dial(SIP/123,,Ttg)
> exten=> 1,2,Gotoif($[${DIALSTATUS}=ANSWERED]?:10)
> exten=> 1,3,Hangup 
> 
> exten=> 1,10,Dial(Local/2,,Ttg)
> exten=> 1,11,Hangup
> 
> exten=> 2,1,Dial(SIP/234,,Ttg)
> exten=> 2,2,Hangup
> 
> All the CDR variables for the first channel (SIP/123) are fine. but
> when local channel initiates, it does not copy the CDR(accountcode)
> variable from the first channel (in asterisk 1.4.4), whereas it did in
> 1.4.2. so the CDR(accountcode) variable for local channel is empty in
> 1.4.4. This is a big problem for me as i have to charge the forwarded
> calls also and all calls are charged based on account code. If
> accountcode is empty, i cant make a decision how to charge the call. 
> 
> Can anybody fix this for me or do i have to jump back to asterisk
> 1.4.2?
> 
> -- 
> Regards
> Rizwan Hisham
> Software Engineer 

Riswan--

This could easily be my fault. I've attached a fix, that I can commit to
the source, if it works for you.

Here the instructions:

1. save the attachment to a file.
2. cd to your 1.4-source/channels directory
3. patch -p0 < localfix
4. cd ..
5. make
6. make install

test

If there's no differences, you still have the same problem, you'd best
restore the source to it's previous condition:

1. cd 1.4-sourcedir/channels
2. mv chan_local.c.orig chan_local.c
3. cd ..
4. make
5. make install

This patch will properly set the accountcode amaflag from the local
channel's owner at channel creation time, and therefore, the local
channels' CDR as well.


-- 
Steve Murphy
Software Developer
Digium
Index: chan_local.c
===
--- chan_local.c	(revision 62984)
+++ chan_local.c	(working copy)
@@ -594,10 +594,22 @@
 {
 	struct ast_channel *tmp = NULL, *tmp2 = NULL;
 	int randnum = ast_random() & 0x, fmt = 0;
+	const char *t;
+	int ama;
 
 	/* Allocate two new Asterisk channels */
-	if (!(tmp = ast_channel_alloc(1, state, 0, 0, "", p->exten, p->context, 0, "Local/[EMAIL PROTECTED],1", p->exten, p->context, randnum)) 
-			|| !(tmp2 = ast_channel_alloc(1, AST_STATE_RING, 0, 0, "", p->exten, p->context, 0, "Local/[EMAIL PROTECTED],2", p->exten, p->context, randnum))) {
+	/* safe accountcode */
+	if (p->owner && p->owner->accountcode)
+		t = p->owner->accountcode;
+	else
+		t = "";
+
+	if (p->owner)
+		ama = p->owner->amaflags;
+	else
+		ama = 0;
+	if (!(tmp = ast_channel_alloc(1, state, 0, 0, t, p->exten, p->context, ama, "Local/[EMAIL PROTECTED],1", p->exten, p->context, randnum)) 
+			|| !(tmp2 = ast_channel_alloc(1, AST_STATE_RING, 0, 0, t, p->exten, p->context, ama, "Local/[EMAIL PROTECTED],2", p->exten, p->context, randnum))) {
 		if (tmp)
 			ast_channel_free(tmp);
 		if (tmp2)


smime.p7s
Description: S/MIME cryptographic signature
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RE: [asterisk-users] "you have been kicked my this conference"

[Salvatore Giudice]

Replace it with a pause sound byte.

--
Salvatore Giudice
[EMAIL PROTECTED]

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Thursday, May 03, 2007 11:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] "you have been kicked my this conference" 

How do I stop the "you have been kicked by this conference" message
 from speaking?

I first had MeetMe(conf, l) and I get the kicked message.

I tried Meetme(CONF, lq) and I still get he kicked message.

and it still says it.

Thanks,

Jerry
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Re: [asterisk-users] Linksys SPA3012 inbound FXO problems

[Dave Cotton]

On Thu, 2007-05-03 at 17:56 +0200, lenz wrote:
> Hello list,
> hope someone can help me - I'm going crazy using the FXO port a SPA3012.
> I would like the SPA 3012 to act as a simple FXO port to an Asterisk, that  
> is, once it detects a call, it should simply send it over to the local  
> Asterisk server. No intelligent routing, PIN, anything else
> 
> I configured it like this:
> 
> PSTN-To-VoIP Gateway Setup
> PSTN-To-VoIP Gateway Enable: yes
> PSTN Caller Auth Method: none
> PSTN Ring Thru Line 1: no
> PSTN Caller Default DP: 1
> 
> Then I configured the dialplan #1 as:
> Dial Plan 1: (S0<:@gw1>)

And that's where it started to go wrong.

(S0<:[EMAIL PROTECTED]:5060>)

will do what you want.


-- 
Dave Cotton <[EMAIL PROTECTED]>


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Re: [asterisk-users] OT: USB T1/E1 Interface?

[Frank]

> Why? There used to be a saying 'usb is for mice, firewire is for men',
> though USB has grown a bit in bandwidth since then, it is still not very
> well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I
> suspect a lack of demand. Havng a E1 termintae in your laptop is quite
> useless, and a server usually has plenty of slots (if not, buy a bigger
> server ;-).

I can well understand the idea of having USB T1 adapters since that way 
you can colocate 1U Asterisk systems  ;-) which at least doubles you 
density in a rack...

Frank


Frank Gorgas-Waller
Explido Software USA Inc.
Phone +1-863-248-1195  &  Fax +1-863-248-1155
EMail  [EMAIL PROTECTED]  &  ICQ 7733546
--QQ-
We teach penguin to fly http://www.explido.us

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Re: [asterisk-users] Large dial plans and variables

[Doug Garstang]

Andreas Sikkema wrote:

You're so right!

I thought about having just a catchall _. extension in the
dialplan and doing everything else in a "real" language via AGI -
PHP, Perl, ... whichever you like. It would make the programming
part much easier as the scope of variables is just as you
expect it to be.



Well, they're called macro's for a reason You guys are 
proposing adding functions or procedures. 

My first step in any macro would be to copy incoming 
variables, be it arguments or even asterisk defined stuff 
to local variables. But that is just me and my coding 
convention.


  

I guess we are. I propose we add functions or procedures!

Until that time though, it seems best practices are to prefix every 
single variable in macros, including copying ARG parameters to 
variables, with the name of the function, to avoid stepping on yourself.



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Re: [asterisk-users] Digital Phones

[Stephen Bosch]

Jason Fuermann wrote:
> I've used these gateways and never experienced any of these problems. I
> could imagine me missing the popping noise but I do know that MWI did
> work just fine.

What he said was that he couldn't turn stutter dialtone off, not that
the MWI didn't work.

Not hearing the DTMF would be a show stopper for me too.

-Stephen-
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Re: [asterisk-users] Daemontools and holidays macro

[Stephen Bosch]

Vicente Aguilar wrote:
> El jue, 03-05-2007 a las 07:04 -0500, William Moore escribió:
>> You may want to consider renaming "daemontools" as it is also the name
>> of a windows program that allows you to mount CD/DVD ISOs, so there
>> could be some confusion.
> 
> daemontools is not the name of my scripts, but the name of a program
> they use to ensure Asterisk is running 100% of the time:
> 
> http://cr.yp.to/daemontools.html
> 
> *This* daemontools spawn a process and monitor it, re-running it if it
> dies.
> 
> Quite useful for high-availability services.

I'm surprised nobody mentioned using daemontools for Asterisk sooner.

-Stephen-
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Re: [asterisk-users] Daemontools and holidays macro

[Stephen Bosch]

William Moore wrote:
> You may want to consider renaming "daemontools" as it is also the name
> of a windows program that allows you to mount CD/DVD ISOs, so there
> could be some confusion.

Uh... This is Daniel Bernstein's 'daemontools' -- and he's not going to
rename it, especially since his software predates the Windows program.

-Stephen-
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RE: [asterisk-users] Linksys SPA3012 inbound FXO problems

[Cosmin Prund]

I've managed to configure a SPA3012 to do that a few days ago. I
remamber using something like "S0:<[EMAIL PROTECTED]>" for the
#1 dial plan. Unfortunately I no longer have access to the SPA because I
"shiped" it to an co-worker and this co-worker didn't manage to install
it yet.

I also remamber an odd thing: the extension really needs to exist in the
correct context, it doesn't fall back to the "s" extension and there's
no worning on the CLI ither!

Also an googling tip: most configuration for the SPA3012 is the same as
that for SIPURA 3000, so google for that too. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of lenz
Sent: Thursday, May 03, 2007 6:57 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Linksys SPA3012 inbound FXO problems

Hello list,
hope someone can help me - I'm going crazy using the FXO port a SPA3012.
I would like the SPA 3012 to act as a simple FXO port to an Asterisk,
that  
is, once it detects a call, it should simply send it over to the local  
Asterisk server. No intelligent routing, PIN, anything else

I configured it like this:

PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: yes
PSTN Caller Auth Method: none
PSTN Ring Thru Line 1: no
PSTN Caller Default DP: 1

Then I configured the dialplan #1 as:
Dial Plan 1: (S0<:@gw1>)

And I configured gateway 1 as:

Gateway Accounts
Gateway 1: my.asterisk.server   
GW1 NAT Mapping Enable:  no
GW1 Auth ID:   --my-sip-login--
GW1 Password:  --my-sip-password--

But it seems to simply ignore incoming calls at all
Anybody's got a pointer to get me started?
Thanks in advance,
l.



-- 
Home of QueueMetrics - http://queuemetrics.com

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Re: [asterisk-users] IAX Trunk

[Ronaldo]

Hi Bruce,

Can you suggest me any documentation about using IAX truking?
Thank you.

Ronaldo.

Bruce Reeves wrote:

Yes it is.

On 5/3/07, *Ronaldo* <[EMAIL PROTECTED] 
> wrote:


Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.

Thansk.

Ronaldo
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--
Bruce Reeves
Nortex Networks


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Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation

[Stephen Bosch]

Mats Karlsson wrote:
> Take a look here:
> http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html

Ugh. This is a Win32 app, isn't it?

-Stephen-
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Re: [asterisk-users] IAX Trunk

[Ronaldo]

Hi Dean,

Can you suggest me any documentation about using IAX trunking?
Thank you.

Ronaldo.

Dean Collins wrote:
Yes it is. 


Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph


  

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ronaldo
Sent: Thursday, 3 May 2007 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX Trunk

Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.

Thansk.

Ronaldo
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Re: [asterisk-users] UK - 2 port ISDN2e cards ...

[Stephen Bosch]

Gordon Henderson wrote:
> On Thu, 3 May 2007, Per Jessen wrote:
>> (aren't you guys getting rid of ISDN anyway? :-)
> 
> H... Some people would like to think so, but it's going to be here
> for a long time yet! BT have/are dumping the "consumer" versions of
> ISDN2 - "home highway" which went a while back, but business highway is
> going soon if it's not already gone, which is a real shame as they had
> almost all the functionality a small business needed for less price than
> the full ISDN2e...

Another revenue generating feature!

-Stephen-
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RE: [asterisk-users] OT: USB T1/E1 Interface?

[Michael Collins]

> Why? There used to be a saying 'usb is for mice, firewire is for men',
> though USB has grown a bit in bandwidth since then, it is still not
very
> well suited for a high sustained bandwidth. NOw T1/E1 is not that big,
I
> suspect a lack of demand. Havng a E1 termintae in your laptop is quite
> useless, and a server usually has plenty of slots (if not, buy a
bigger
> server ;-).


Just for fun.  I'm a telecom geek and having a USB T1 interface would be
a fun toy to tinker with.  Besides, it might lead to some useful
products.

-MC
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Re: [asterisk-users] IAX Trunk

[William Moore]

On 5/3/07, Ronaldo <[EMAIL PROTECTED]> wrote:

Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.


Yes, Asterisk will do the conversion from SIP to IAX and back again if
necessary.
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Re: [asterisk-users] IAX Trunk

[Bruce Reeves]

Yes it is

On 5/3/07, Ronaldo <[EMAIL PROTECTED]> wrote:


Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.

Thansk.

Ronaldo
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--
Bruce Reeves
Nortex Networks
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Re: [asterisk-users] IAX Trunk

[Bruce Reeves]

Yes it is.

On 5/3/07, Ronaldo <[EMAIL PROTECTED]> wrote:


Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP
clients that need to talk to each other.

Thansk.

Ronaldo
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--
Bruce Reeves
Nortex Networks
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RE: [asterisk-users] IAX Trunk

[Dean Collins]

Yes it is. 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ronaldo
> Sent: Thursday, 3 May 2007 12:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] IAX Trunk
> 
> Hi all,
> 
> Is it possible to have something like this:
> 
> SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone
> 
> I want a IAX trunk between two asterisks and on each tip I have SIP
> clients that need to talk to each other.
> 
> Thansk.
> 
> Ronaldo
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> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] IAX Trunk

[Ronaldo]

Hi all,

Is it possible to have something like this:

SoftPhone -(SIP)-> Asterisk -(IAX trunk)-> Asterisk -(SIP)-> SoftPhone

I want a IAX trunk between two asterisks and on each tip I have SIP 
clients that need to talk to each other.


Thansk.

Ronaldo
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