[asterisk-users] Re: OT: Patch to OSS app for CDP without a Cisco switch - TESTERS WANTED

[Tomislav Parcina]

Kristian Kielhofner wrote:

Hey everyone,

 I came across a situation where I needed to use CDP to advertise a
voice vlan to Polycom/Cisco (and other CDP capable phones) without a
Cisco switch.


Hi Kristian!

Thank you for your work. I'm not able to test this right now, but I'll 
sourly need this sometimes.



--
Tomislav Parcina
[EMAIL PROTECTED]

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[asterisk-users] Re: 1.4 lost internet internal phones loose registration

[Tomislav Parcina]

Thomas Kenyon wrote:
Asterisk also seems to barf if it makes a registration/renewal request 
and it doesn't receive a reply in a timely fashion which will obviously 
happen if your internet connection disappears. (all versions I've used).


That's why people should use dnsmasq.


--
Tomislav Parcina
[EMAIL PROTECTED]

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[asterisk-users] Re: Registrations, how many is too many?

[Tomislav Parcina]

voiplist wrote:

We do not use dyndns for anything, not sure what we would even use it for.

We do have lots of hostnames to different systems in our sip.conf, I
have changed them all to IP to see if this helps.

So, you think that maybe when DNS gets hosed up that it could cause
SIP to just tank on a high volume system?


Of course you need to have DNS server installed on Asterisk machine. So, 
Asterisk will ask that machine for DNS records. If that machine doesn't 
know the answer (because Internet connection is down), at least Asterisk 
will get fast answer so it won't stop responding.



--
Tomislav Parcina
[EMAIL PROTECTED]

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[asterisk-users] app_queue not using exit context?

[Steve Edwards]

Before I report this as a bug (and get whacked with more bad karma), I'd
like to make sure I'm understanding this "feature."

I'm defining a queue with a couple of SIP phones as the memebers -- not 
agents.


queue.conf allows you to set an exit context such that if set (and you use 
the "T" or "t" option) allows the caller or callee to transfer the call to 
that context/extension.


This doesn't appear to be working for me.

If I set "T," app_queue looks for the extension in the caller's current 
context -- where the queue() call is.


If I set "t," app_queue looks for the extension in the callee's context -- 
in my case, the context specified in sip.conf.


Am I misunderstanding how this is supposed to work?

Is it working for you?

Relevant snippet from my queue.conf:

[customer-service]
context = customer-service
member  = sip/pap2-000F66A83C90-line-1

Relevant snippet from my sip.conf:

[pap2-000F66A83C90-line-1]
context = inside
type= friend

Relevant snippet from my extensions.conf:

[first-time-caller](h,s)
exten = s,n,
set(CALLERID(number)=${CARD-NUMBER:0:4}${CARD-NUMBER:-4})
exten = s,n,queue(customer-service|nrt)

[customer-service](h,i,s)
exten = s,n,hangup
exten = 3,1,goto(redirect,s,1)
exten = 7,1,goto(theme,s,1)
exten = 8,1,goto(enter-card-number,s,1)

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] IAX2, DTMF and x86_64.

[William F. Acker WB2FLW +1-303-722-7209]

Hi all,

 I'm just starting to play with 1.4.  I installed 1.4.1 on an Ia32 
machine, and can't find any problems.  So, I decided to upgrade my home 
pbx.  All went well until I tried using my S101 to talk to the IVR.  Some 
times, the first one or two digits get through, but eventually a digit 
will get stuck, playing continuously until the call is terminated.  I 
have confirmed this on another x86_64 machine that I connect with.


 Also, when I reloaded IAX2, Asterisk crashed with a message about a 
double linklist and an ugly trace.  Unfortunately, the crash didn't make 
it into the logs.


  Any ideas?


--
Bill in Denver
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[asterisk-users] server generated outbound conference calls?

[Dean Collins]

Is anyone currently generating asterisk server outbound conference calls
via some form of desktop application or IM client?

 

What I mean by this is can I currently initiate an event on my asterisk
server where it dials me first as a conference initiator and then 4 of
my contacts by me either;

 

1.  Right clicking in Outlook on their names and highlighting join
conference command?
or
2.  Dropping and dragging multiple Outlook icons onto a desktop
application (such as hud-light)
or
3.  Using some form of IM (either Windows Messenger, Jive, Gtalk or
Jabber client) initiate conference calls between multiple buddies? 

 

 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).

 

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Re: [asterisk-users] How to disable MOH completely?

[David Thomas]

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

Could be its trying but does it actualy play the music?


It's not actually playing anything. I guess it just seems odd that
Asterisk re-invites the media back to itself when a call is put on
hold (when MOH is disabled), instead of simply disconnecting the media
until the call is retrieved. I guess I was hoping for a config option
that would simply turn MOH off to achieve this behavior.

Does such a config option exist in 1.4?

Regards,
David
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[asterisk-users] Polycom Questions

[»Steven Ringwald«]

Any Polycom gurus out there? If so, I have a few config file questions.

First off, does anyone have the daylight savings time rules written for 
this Sunday's big change?


Secondly, if there any way in the config file to tell the phone not to 
display the number of missed calls? I don't mind it keeping the missed 
calls list, I just don't want that running count.


Lastly, I am trying to get the dialplan to work, but have had no luck so 
far. I have tried defining it in the sip.cfg and/or the phone1.cfg, but 
have had no luck getting the phone to latch onto the numbers, and 
immediately dial. I am running with the 2.0.1 firmware, if that matters.


from sip.cfg:

  dialplan.removeEndOfDial="1">
 dialplan.digitmap="9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx" 
dialplan.digitmap.timeOut="3"/>

 
dialplan.routing.server.1.port="5060"/>
dialplan.routing.emergency.1.server.1="1"/>

 
  

from phone1.cfg:

dialplan.1.removeEndOfDial="1" dialplan.2.impossibleMatchHandling="0" 
dialplan.2.removeEndOfDial="1" dialplan.3.impossibleMatchHandling="0" 
dialplan.3.removeEndOfDial="1" dialplan.4.impossibleMatchHandling="0" 
dialplan.4.removeEndOfDial="1" dialplan.5.impossibleMatchHandling="0" 
dialplan.5.removeEndOfDial="1" dialplan.6.impossibleMatchHandling="0" 
dialplan.6.removeEndOfDial="1">
 dialplan.1.digitmap="9[2-9]xx[2-9]xx|91[2-9]xx[2-9]xx" 
dialplan.1.digitmap.timeOut="3" dialplan.2.digitmap="" 
dialplan.2.digitmap.timeOut="" dialplan.3.digitmap="" 
dialplan.3.digitmap.timeOut="" dialplan.4.digitmap="" 
dialplan.4.digitmap.timeOut="" dialplan.5.digitmap="" 
dialplan.5.digitmap.timeOut="" dialplan.6.digitmap="" 
dialplan.6.digitmap.timeOut=""/>

 
dialplan.1.routing.server.1.port="5060" 
dialplan.2.routing.server.1.address="" 
dialplan.2.routing.server.1.port="" 
dialplan.3.routing.server.1.address="" 
dialplan.3.routing.server.1.port="" 
dialplan.4.routing.server.1.address="" 
dialplan.4.routing.server.1.port="" 
dialplan.5.routing.server.1.address="" 
dialplan.5.routing.server.1.port="" 
dialplan.6.routing.server.1.address="" dialplan.6.routing.server.1.port=""/>
dialplan.1.routing.emergency.1.server.1="" 
dialplan.2.routing.emergency.1.value="" 
dialplan.2.routing.emergency.1.server.1="" 
dialplan.3.routing.emergency.1.value="" 
dialplan.3.routing.emergency.1.server.1="" 
dialplan.4.routing.emergency.1.value="" 
dialplan.4.routing.emergency.1.server.1="" 
dialplan.5.routing.emergency.1.value="" 
dialplan.5.routing.emergency.1.server.1="" 
dialplan.6.routing.emergency.1.value="" 
dialplan.6.routing.emergency.1.server.1=""/>

 
  



Thanks in advance!
Steve

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Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com

[Andres]

Rehan Ahmed



Come on Rehan... Do you think we're really going to fall for that
trick. We all know you represent virtualphoneline.com.


he is so clueless I can't believe his companies are still in business



Regards,
David
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Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com

[Dovid B]

On Tue, 06 Mar 2007 05:12:03, [EMAIL PROTECTED] <[EMAIL PROTECTED]> 
wrote:


Dear Asterisk Users Mailing List - Non-Commercial Discussion,

I joined VirtualPhoneLine.Com service and am really enjoying the use of 
it.


VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in 
the world and then forwards it to my Mobile Number, Regular Phone, MSN 
Messenger, Google Talk or an IP Phone.


Have a look at the http://www.virtualphoneline.com/faq and 
http://www.virtualphoneline.com/did for current available numbers.


Follow this link 
http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129b&OID=10


Let me know how it goes for you,

Rehan Ahmed


Come on Rehan... Do you think we're really going to fall for that
trick. We all know you represent virtualphoneline.com.

Regards,
David


Now Linda has to go on a new PR campaign.. This is what happens when out 
source for cheaper  rates 



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Re: [asterisk-users] How to disable MOH completely?

[C F]

Could be its trying but does it actualy play the music?

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

On 3/5/07, C F <[EMAIL PROTECTED]> wrote:
> Just comment everything in your musiconhold.conf
>

Funny thing is, I don't have a musiconhold.conf and res_musiconhold.so
is not loaded, however when I press flash or hold on my phone
(connected to an ATA), on the CLI I see Asterisk try to execute music
on hold.

Regards,
David
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[asterisk-users] extra-sounds 1.4.5 timestapmed newer than 1.4.6 ???

[Mr. James W. Laferriere]

	Hello All ,  I'd usually just take the latest timestamped tarballs & use 
them ,  But this has gotten me a tad setback .
	I want to build astersik-1.4.1 & I am not sure which of these is going 
to work correctly .  Anyone else have a better idea than me ?

Rsvp ,  Tia ,  JimL


-rw-r--r--1 00 9397296 Feb 21 01:07 
asterisk-core-sounds-es-g722-1.4.6.tar.gz
-rw-r--r--1 00 2129399 Feb 21 01:07 
asterisk-core-sounds-es-gsm-1.4.6.tar.gz
-rw-r--r--1 0012219353 Feb 21 01:07 
asterisk-core-sounds-en-alaw-1.4.6.tar.gz
-rw-r--r--1 00 7167005 Feb 21 01:07 
asterisk-core-sounds-fr-g722-1.4.6.tar.gz
-rw-r--r--1 00 6874032 Feb 21 01:07 
asterisk-core-sounds-en-g729-1.4.6.tar.gz
-rw-r--r--1 00 1623436 Feb 21 01:07 
asterisk-core-sounds-fr-gsm-1.4.6.tar.gz
-rw-r--r--1 0018603291 Feb 21 01:07 
asterisk-core-sounds-es-wav-1.4.6.tar.gz
-rw-r--r--1 0012278506 Feb 21 01:07 
asterisk-core-sounds-en-ulaw-1.4.6.tar.gz
drwxr-xr-x3 008192 Feb 22 00:32 .
-rw-r--r--1 0027839721 Feb 22 00:32 
asterisk-extra-sounds-en-wav-1.4.5.tar.gz
-rw-r--r--1 001375 Feb 22 00:32 
asterisk-extra-sounds-en-ulaw-1.4.5.tar.gz
-rw-r--r--1 0013675929 Feb 22 00:32 
asterisk-extra-sounds-en-g722-1.4.5.tar.gz
-rw-r--r--1 00 3235653 Feb 22 00:32 
asterisk-extra-sounds-en-gsm-1.4.5.tar.gz
-rw-r--r--1 0013473844 Feb 22 00:32 
asterisk-extra-sounds-en-alaw-1.4.5.tar.gz
-rw-r--r--1 00 2017747 Feb 22 00:32 
asterisk-extra-sounds-en-g729-1.4.5.tar.gz
drwxr-xr-x4 004096 Feb 22 00:40 ..
drwxr-xr-x6 004096 Mar 06 00:50 .svn
ncftp ...ephony/sounds/releases > dir -alrt

--
+-+
| James   W.   Laferriere | System   Techniques | Give me VMS |
| NetworkEngineer | 663  Beaumont  Blvd |  Give me Linux  |
| [EMAIL PROTECTED] | Pacifica, CA. 94044 |   only  on  AXP |
+-+
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Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com

[David Thomas]

On Tue, 06 Mar 2007 05:12:03, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:


Dear Asterisk Users Mailing List - Non-Commercial Discussion,

I joined VirtualPhoneLine.Com service and am really enjoying the use of it.

VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in the 
world and then forwards it to my Mobile Number, Regular Phone, MSN Messenger, 
Google Talk or an IP Phone.

Have a look at the http://www.virtualphoneline.com/faq and 
http://www.virtualphoneline.com/did for current available numbers.

Follow this link 
http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129b&OID=10

Let me know how it goes for you,

Rehan Ahmed


Come on Rehan... Do you think we're really going to fall for that
trick. We all know you represent virtualphoneline.com.

Regards,
David
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[asterisk-users] [Announce] Web-MeetMe V3.0.1 released

[Dan Austin]

Minor bug-fix release, no new functionality.

Bugs fixed:
*  app_cbmysql would fail to load
*  Incorrect handling of recurring conferences that
spanned a DST transition

Minor cleanup:
*  A couple image files were duplicated with 
both upper and lowercase names.  The 
uppercase variants were deleted and the
HTML code cleaned up to use just the
remaining files.

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

We do have a volunteer developer who will be maintaining the
2.X.X chain for Asterisk 1.2.X compatibility, so bug fixes and
features that are not Asterisk version dependant will still be
made available for older installations.

The 2.X.X chain does not have the problem with app_cbmysql,
but may suffer from the DST transition bug.

Thanks,
The Web-MeetMe development team...
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[asterisk-users] Using Asterisk as Voicemail Server on a dinosaur Meridian System

[J French]

We have a dinosaur Meridian system (~version 2) with 4 digital lines going
to a Repartee Voicemail server.  The Repartee got smoked by lightning two
days ago and I'm itching to get Asterisk installed in its place.  PRI is not
an option since the system is so old that it doesn't even support PRI.  I
need to figure out how to connect the old Meridian to Asterisk otherwise.

Any advice on getting Asterisk to work in its place is really appreciated.
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Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com

[Andrew Joakimsen]

Or that its not even a new service?

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

Or the fact that www.virtualphoneline.com is part of DIDXchange and of
course you love it since you work for supertec.com, didxchange.com,
and virtualphoneline.com

On 3/5/07, Singer Wang <[EMAIL PROTECTED]> wrote:
> Follow this link
> 
http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129b&OID=10
>
>
> non commerical eh? care to remove that Rferreal2= part?
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--
Bruce
Nortex Networks
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[asterisk-users] Voicemail question

[Hall, Eric M.]

Group

 In voicemail.conf I would like to having the following setup per
context not per-mailbox settings  

 

serveremail 

userscontext

fromstring

usedirectory

emailbody

pagerfromstring

dialout 

sendvoicemail

callback

review

operator

 volgain

nextaftercmd

forcename

forcegreetings

tempgreetwarn

 

Can this be done?

 

Thanks!

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Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com

[Bruce Reeves]

Or the fact that www.virtualphoneline.com is part of DIDXchange and of
course you love it since you work for supertec.com, didxchange.com,
and virtualphoneline.com

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

Follow this link
http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129b&OID=10


non commerical eh? care to remove that Rferreal2= part?
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--
Bruce
Nortex Networks
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[asterisk-users] Instant Messaging with SIP Softphone Eyebeam (was: SMS ON ASTERISK)

[Anselm Martin Hoffmeister]

Am Montag, den 05.03.2007, 09:01 -0300 schrieb Assis, Eduardo:
> We installed na Asterisk System whith 400 Softphone users (Eyebeam 1.5
> from Counterpath).
> 
> As far as we know, Asterisk don't support yet IM (Instante Message)
> feature,instead Eyebeam have this feature.

I cross-read their handbook and did not find anything about the
Instant-Messaging feature besides its existence and how to send a
message. I _guess_ it works with SIP messages (like SendText and
ReceiveText Asterisk commands, perhaps?), but it surely is NOT related
to the "SMS" application of Asterisk.

app_sms supports landline SMS over analogue and ISDN lines, meaning the
short message is encoded in a few-second 1200bps modem conversation. Of
course there are more appropriate ways to send messages if you are on an
IP network :-)

If I were you, I would try to fiddle with the SendText and ReceiveText
stuff (the latter might be available in AGI only - there are docs on the
voip-info.org website, and google is our friend).

SendText might only work while in a call, like
exten => 1,1,Answer
exten => 1,2,SendText(Lorem ipsum dolor)
(send text to the caller's display)

I cannot guarantee that it will work at all, there might still be other
means off messaging that I just do not have an idea of.

Did you test at all wether messages go anywhere, or what happens if you
send a message?

Best regards,
Anselm

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[asterisk-users] Setting Sip Headers From Dial App?

[Stuart Sheldon]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

This might sound strange, but is there anyway for Asterisk to set extra
sip headers based on a sip phone returning a 302 in a dialplan?

Example:

PSTN => Asterisk => SIP-Phone, SIP-Phone returns 302 Redirect, Asterisk
sets X-Something: Some_Value & X-Somethingelse: Some_Other_Value, then
sends the new invite with added headers.

Stu Sheldon
ACT USA

- --
Randomly Generated Fortune Tag:
Q:  How do you catch a unique rabbit?
A:  Unique up on it!

Q:  How do you catch a tame rabbit?
A:  The tame way!
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.6 (GNU/Linux)

iD8DBQFF7KIhK69Y+xPZrWYRAgmWAJ9FvL+BqRr5YzXSYlkn9vLu4mHq2ACfaKrc
LJts0IptsnzfawJzMNWibnM=
=Kjdf
-END PGP SIGNATURE-
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[asterisk-users] g.729 on solaris10/x86

[Juraj Bednar]

Hello,


  I'm looking for a way to have G.729 codec working on Solaris/x86.
Binary codec from Digium is not compiled for Solaris/x86 (only sparc).
Are there any alternative (free or commercial) G.729 implementations,
which would work?

  I saw something from Intel and got it to compile on Linux, but it
was only for evaluation purposes, so we upgraded to commercial codec
from Digium. I really don't care about the U.S. patent, it does not
apply here, only about copyright. If there's something with source
code (could be commercial), that I can make work on Solaris, it would
be great.


  Thank you,

Juraj.
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Re: [asterisk-users] How to disable MOH completely?

[David Thomas]

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

Just comment everything in your musiconhold.conf



Funny thing is, I don't have a musiconhold.conf and res_musiconhold.so
is not loaded, however when I press flash or hold on my phone
(connected to an ATA), on the CLI I see Asterisk try to execute music
on hold.

Regards,
David
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Re: [asterisk-users] A New Phone Service - www.virtualphoneline.com

[Singer Wang]

Follow this link 
http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129b&OID=10 



non commerical eh? care to remove that Rferreal2= part?
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Re: [asterisk-users] Re: Asterisk Java w/ Threads

[Stefan Reuter]

Eric "ManxPower" Wieling wrote:
> In the past, the Asterisk Manager Interface was prone to crashes if it
> had more than 1 client connected to it.  The proxy solved that issue.  I
> think this issue was resolved in 1.2.

Yes, this was indeed a problem with 1.0. I didn't encounter any problems
regarding this for 1.2 and 1.4.
Connecting from one application application to multiple Asterisk servers
(which was the question) has never been a problem though.

=Stefan

-- 
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Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

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Re: [asterisk-users] How to disable MOH completely?

[C F]

Just comment everything in your musiconhold.conf

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

I need to disable MOH completely. We are using all SIP extensions and
do not want Asterisk to invoke MOH when flash or hold is pressed on
the phone.

Anyone know how to configure this?

Thanks!
David
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[asterisk-users] Re: TDM400P/FXS in a HP DL380 G5

[James FitzGibbon]

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


Has anyone figured out a solution for this?  Something along the lines of
an external power brick whose output attaches to a backplane slot and gives
you a 12V connector inside the server?



Since i posted my original request I stumbled across this:

http://www.coolerguys.com/840556029977.html

They say it's custom made for them, and I certainly can't find anything else
like it after several hours of searching, but it seems to be what's
required.  I'll have to rig up a backplate with a cutout to get the 12V
connector into the case, but other than that I'm hoping it will do the
trick.

--
j.
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[asterisk-users] A New Phone Service - www.virtualphoneline.com

[rehan]

Dear Asterisk Users Mailing List - Non-Commercial Discussion,

I joined VirtualPhoneLine.Com service and am really enjoying the use of it.

VirtualPhoneLine.Com allows me to buy virtual numbers from anywhere in the 
world and then forwards it to my Mobile Number, Regular Phone, MSN Messenger, 
Google Talk or an IP Phone.

Have a look at the http://www.virtualphoneline.com/faq and 
http://www.virtualphoneline.com/did for current available numbers.

Follow this link 
http://www.virtualphoneline.com/signup/index.php?Referral2=vpl072625129b&OID=10
 

Let me know how it goes for you,

Rehan Ahmed

 EmailID: 25
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Re: [asterisk-users] TDM400P/FXS in a HP DL380 G5

[Hans Witvliet]

On Mon, 2007-03-05 at 12:54 -0500, James FitzGibbon wrote:
> The HP DL380 G5 (like many rack servers) has no AMP Mate-n-Lok
> connector available to attach to a card that needs more power than the
> PCI bus can provide, like the TDM400P when FXS modules are used.  HP
> has confirmed that there is no part they sell to give you such a
> connector, and Digium says their business edition folks got it to
> work, but only by doing nasty warranty-voiding things to the internal
> wiring. 
> 
> Has anyone figured out a solution for this?  Something along the lines
> of an external power brick whose output attaches to a backplane slot
> and gives you a 12V connector inside the server?
> 
> Or am I just SOL? 
> 
> Thanks

It was exactly because of this (voiding garantee on a bunch of DL380-G4)
that i had to advice to expand their Aterisk-configuration with either:
1) ordinary desktop with one or more TDM's
2) multiport ata's
3) T1 + channelbanks

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Re: [asterisk-users] Re: Asterisk Java w/ Threads

[Eric \"ManxPower\" Wieling]

Stefan Reuter wrote:

With Asterisk-Java the proposed solution to connect to multiple Asterisk
servers is to create multiple AsteriskManagerConnection obeject.
Each ManagerConnection handles its own thread so there is no need for
custom thread handing code.
All you have to do is to make sure is the EventListener objects you pass
to these connections synchronize access to shared data (if there are
such accesses).
I think this approach is rather simple for the user and don't see a
benefit in adding a proxy to that picture.


In the past, the Asterisk Manager Interface was prone to crashes if it 
had more than 1 client connected to it.  The proxy solved that issue.  I 
think this issue was resolved in 1.2.

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[asterisk-users] Re: Asterisk Java w/ Threads

[Stefan Reuter]

With Asterisk-Java the proposed solution to connect to multiple Asterisk
servers is to create multiple AsteriskManagerConnection obeject.
Each ManagerConnection handles its own thread so there is no need for
custom thread handing code.
All you have to do is to make sure is the EventListener objects you pass
to these connections synchronize access to shared data (if there are
such accesses).
I think this approach is rather simple for the user and don't see a
benefit in adding a proxy to that picture.

=Stefan

-- 
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail:  [EMAIL PROTECTED]
Jabber:  [EMAIL PROTECTED]

Steuernummern 215/5140/1791 USt-IdNr. DE220701760



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RE: [asterisk-users] 1.4 - SLA

[Bill Gibbs]

Sorry to reply to myself, once again onn the list, but since SLA is new
I figured I should answer my own question before anyone else gets
confused...I completely forgot about my -directory.xml defaults...so
that's where all these bogus SUBSCRIBE requests were coming from.

Bill

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs
Sent: Monday, March 05, 2007 2:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] 1.4 - SLA

Here is the debug output of the SUBSCRIBE request

I am sure it has something to do with the way I am attempting to setup
the Polycom for shared appearances...

Nat=yes is set in the peer.  I don't get these weird messages when
connecting with a "private" line appearance.

<--- SIP read from x.x.x.x:60671 --->
SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bKf2bac598B416612D
From: "Line 1" ;tag=259528A1-76B251C6
To: 
CSeq: 1 SUBSCRIBE
Call-ID: [EMAIL PROTECTED]
Contact: 
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<->
[Mar  5 14:25:02] VERBOSE[9835] logger.c: --- (13 headers 0 lines) ---

[Mar  5 14:25:02] VERBOSE[9835] logger.c: Sending to 192.168.1.116 :
5060 (no NAT)
[Mar  5 14:25:02] VERBOSE[9835] logger.c: Found no matching peer or user
for 'x.x.x.x:60671'
[Mar  5 14:25:02] VERBOSE[9835] logger.c: Looking for 103 in default
(domain x.x.x.x)
[Mar  5 14:25:02] VERBOSE[9835] logger.c:
<--- Transmitting (no NAT) to 192.168.1.116:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.1.116;branch=z9hG4bKf2bac598B416612D;received=x.x.x.x
From: "Line 1" ;tag=259528A1-76B251C6
To: ;tag=as4d77da56
Call-ID: [EMAIL PROTECTED]
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs
Sent: Monday, March 05, 2007 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] 1.4 - SLA

I have been using 2 Polycom 430s so far.  I can get incoming calls just
fine (both phones ring on line 1).  However it doesn't appear to seize
the line, so if a call is on the one phone, I can still pick up line 1
on the other and dial - and it's reflected in the connected call.  I
assume that's related to the hint/subscription issue Lacy indicated as
well.  "sip show subscriptions" shows nothing.

I just started playing with it this morning however...still playing
around w/ the configs.

One odd thing, I keep seeing some weirdness:
[Mar  5 13:10:52] VERBOSE[9381] logger.c: Looking for 105 in default
(domain x.x.x.x)
And also Looking for 103

Yet I have no idea where those values are coming from!

I am running 1.6.7.

Here is a snippet of the phone config from one of the phones:
mailto:[EMAIL PROTECTED] On Behalf Of Russell
Bryant
Sent: Friday, March 02, 2007 6:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.4 - SLA

Lacy Moore - Aspendora wrote:
> Russell, I don't have any specifics at this time.  I need to dig a
> little further.  I'm thinking the autocontext is what is giving me
> fits.  I can receive calls and place calls, but the hint status is not
> working.  It currently registers as a hint showing not in use.  It
> does not show in use.

If you aren't seeing any lights change on the phones when calls are 
going on, check "sip show subscriptions" at the CLI.  If the phones have

not properly subscribed to the right extensions, you won't see anything.

> I ended up using some of the config from the bottom of the sla.txt
> file.  The sample file may be missing the template section.  The
> sample config does not match the config in the sla.txt.  I couldn't
> get the sample config to work at all.  Again, hopefully over the
> weekend I'll be able to get more information.

You are correct.  The sample configuration is missing the template.  I 
will add it now.  However, I just made the tarballs for 1.4.1, so this 
config fix didn't make it in.

> Using the config in the sample file, the hint status was working.  I
> could see the line ringing, but I could not answer the lines or place
> calls.  Using the config from the sla.txt file, I could place calls
> and receive calls, but the hints never showed any activity, just
> always not in use.

As I noted earlier, check your "sip show subscriptions" to make sure the

phones are subscribed to the right thing.

Another helpful thing that you can use for debugging is to look at the 
output of "sla show stations".  You can see the state of each line 
appearance on each station.  This should correspond with what you see on

the phone  ... unless there is a problem, of course.

> If po

Re: [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P

[younss azzayani]

ok, here i see different config,
like indian T1 crossovercable
http://asterisk.pbx.in/digium-te110p-loopback-cable-india-howto
i'll try this
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RE: [asterisk-users] 1.4 - SLA

[Bill Gibbs]

Here is the debug output of the SUBSCRIBE request

I am sure it has something to do with the way I am attempting to setup
the Polycom for shared appearances...

Nat=yes is set in the peer.  I don't get these weird messages when
connecting with a "private" line appearance.

<--- SIP read from x.x.x.x:60671 --->
SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bKf2bac598B416612D
From: "Line 1" ;tag=259528A1-76B251C6
To: 
CSeq: 1 SUBSCRIBE
Call-ID: [EMAIL PROTECTED]
Contact: 
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094
Max-Forwards: 70
Expires: 3600
Content-Length: 0


<->
[Mar  5 14:25:02] VERBOSE[9835] logger.c: --- (13 headers 0 lines) ---

[Mar  5 14:25:02] VERBOSE[9835] logger.c: Sending to 192.168.1.116 :
5060 (no NAT)
[Mar  5 14:25:02] VERBOSE[9835] logger.c: Found no matching peer or user
for 'x.x.x.x:60671'
[Mar  5 14:25:02] VERBOSE[9835] logger.c: Looking for 103 in default
(domain x.x.x.x)
[Mar  5 14:25:02] VERBOSE[9835] logger.c:
<--- Transmitting (no NAT) to 192.168.1.116:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.1.116;branch=z9hG4bKf2bac598B416612D;received=x.x.x.x
From: "Line 1" ;tag=259528A1-76B251C6
To: ;tag=as4d77da56
Call-ID: [EMAIL PROTECTED]
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs
Sent: Monday, March 05, 2007 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] 1.4 - SLA

I have been using 2 Polycom 430s so far.  I can get incoming calls just
fine (both phones ring on line 1).  However it doesn't appear to seize
the line, so if a call is on the one phone, I can still pick up line 1
on the other and dial - and it's reflected in the connected call.  I
assume that's related to the hint/subscription issue Lacy indicated as
well.  "sip show subscriptions" shows nothing.

I just started playing with it this morning however...still playing
around w/ the configs.

One odd thing, I keep seeing some weirdness:
[Mar  5 13:10:52] VERBOSE[9381] logger.c: Looking for 105 in default
(domain x.x.x.x)
And also Looking for 103

Yet I have no idea where those values are coming from!

I am running 1.6.7.

Here is a snippet of the phone config from one of the phones:
mailto:[EMAIL PROTECTED] On Behalf Of Russell
Bryant
Sent: Friday, March 02, 2007 6:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.4 - SLA

Lacy Moore - Aspendora wrote:
> Russell, I don't have any specifics at this time.  I need to dig a
> little further.  I'm thinking the autocontext is what is giving me
> fits.  I can receive calls and place calls, but the hint status is not
> working.  It currently registers as a hint showing not in use.  It
> does not show in use.

If you aren't seeing any lights change on the phones when calls are 
going on, check "sip show subscriptions" at the CLI.  If the phones have

not properly subscribed to the right extensions, you won't see anything.

> I ended up using some of the config from the bottom of the sla.txt
> file.  The sample file may be missing the template section.  The
> sample config does not match the config in the sla.txt.  I couldn't
> get the sample config to work at all.  Again, hopefully over the
> weekend I'll be able to get more information.

You are correct.  The sample configuration is missing the template.  I 
will add it now.  However, I just made the tarballs for 1.4.1, so this 
config fix didn't make it in.

> Using the config in the sample file, the hint status was working.  I
> could see the line ringing, but I could not answer the lines or place
> calls.  Using the config from the sla.txt file, I could place calls
> and receive calls, but the hints never showed any activity, just
> always not in use.

As I noted earlier, check your "sip show subscriptions" to make sure the

phones are subscribed to the right thing.

Another helpful thing that you can use for debugging is to look at the 
output of "sla show stations".  You can see the state of each line 
appearance on each station.  This should correspond with what you see on

the phone  ... unless there is a problem, of course.

> If possible, could you provide the config that you've used for
> testing?  I'm testing using Polycom phones to try to keep things
> simple.  I'm assuming you are using a Polycom.

I have been testing with a variety of different phones.  I have not 
tested all of the Polycom models, yet.  This is one of the things we're 
going to have to work through.  I would like to document issues with 
specific phones in sla.txt as we come across them.

The configuration I'm using for testing looks just like the stuff in 
configs/sl

RE: [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P

[Michael Collins]

> Hi everybody,
> i need someone to tell me the spins numbers of Rx+,Rx-,Tx+ and Tx- of
> TE110P
> and also if you can tell me have to made a cable like that??
> 
> Modem Teleco <---Self Crosscable>Asterisk

You might check this out for a quick reference:
http://www.voip-info.org/wiki/view/crossover+T1+cable

-MC
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Re: [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P

[Eric \"ManxPower\" Wieling]

younss azzayani wrote:

Hi everybody,
i need someone to tell me the spins numbers of Rx+,Rx-,Tx+ and Tx- of 
TE110P

and also if you can tell me have to made a cable like that??

Modem Teleco <---Self Crosscable>Asterisk
Rx+ <--> 
Tx+
Rx-  
<--> Tx-
Tx+ 
<-->  Rx+
Tx-  
<-->  Rx-

i'm fear that i'm gonna to damage the G723 modem card or TE110P card


http://www.voip-info.org/wiki/view/crossover+T1+cable

http://www.cisco.com/en/US/products/hw/routers/ps233/products_tech_note09186a00800a3f09.shtml

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[asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P

[younss azzayani]

Hi everybody,
i need someone to tell me the spins numbers of Rx+,Rx-,Tx+ and Tx- of TE110P
and also if you can tell me have to made a cable like that??

Modem Teleco <---Self Crosscable>Asterisk
Rx+ <--> Tx+
Rx-  <--> Tx-
Tx+ <-->  Rx+
Tx-  <-->  Rx-
i'm fear that i'm gonna to damage the G723 modem card or TE110P card

Thank you for your help
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RE: [asterisk-users] 1.4 - SLA

[Bill Gibbs]

I have been using 2 Polycom 430s so far.  I can get incoming calls just
fine (both phones ring on line 1).  However it doesn't appear to seize
the line, so if a call is on the one phone, I can still pick up line 1
on the other and dial - and it's reflected in the connected call.  I
assume that's related to the hint/subscription issue Lacy indicated as
well.  "sip show subscriptions" shows nothing.

I just started playing with it this morning however...still playing
around w/ the configs.

One odd thing, I keep seeing some weirdness:
[Mar  5 13:10:52] VERBOSE[9381] logger.c: Looking for 105 in default
(domain x.x.x.x)
And also Looking for 103

Yet I have no idea where those values are coming from!

I am running 1.6.7.

Here is a snippet of the phone config from one of the phones:
mailto:[EMAIL PROTECTED] On Behalf Of Russell
Bryant
Sent: Friday, March 02, 2007 6:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.4 - SLA

Lacy Moore - Aspendora wrote:
> Russell, I don't have any specifics at this time.  I need to dig a
> little further.  I'm thinking the autocontext is what is giving me
> fits.  I can receive calls and place calls, but the hint status is not
> working.  It currently registers as a hint showing not in use.  It
> does not show in use.

If you aren't seeing any lights change on the phones when calls are 
going on, check "sip show subscriptions" at the CLI.  If the phones have

not properly subscribed to the right extensions, you won't see anything.

> I ended up using some of the config from the bottom of the sla.txt
> file.  The sample file may be missing the template section.  The
> sample config does not match the config in the sla.txt.  I couldn't
> get the sample config to work at all.  Again, hopefully over the
> weekend I'll be able to get more information.

You are correct.  The sample configuration is missing the template.  I 
will add it now.  However, I just made the tarballs for 1.4.1, so this 
config fix didn't make it in.

> Using the config in the sample file, the hint status was working.  I
> could see the line ringing, but I could not answer the lines or place
> calls.  Using the config from the sla.txt file, I could place calls
> and receive calls, but the hints never showed any activity, just
> always not in use.

As I noted earlier, check your "sip show subscriptions" to make sure the

phones are subscribed to the right thing.

Another helpful thing that you can use for debugging is to look at the 
output of "sla show stations".  You can see the state of each line 
appearance on each station.  This should correspond with what you see on

the phone  ... unless there is a problem, of course.

> If possible, could you provide the config that you've used for
> testing?  I'm testing using Polycom phones to try to keep things
> simple.  I'm assuming you are using a Polycom.

I have been testing with a variety of different phones.  I have not 
tested all of the Polycom models, yet.  This is one of the things we're 
going to have to work through.  I would like to document issues with 
specific phones in sla.txt as we come across them.

The configuration I'm using for testing looks just like the stuff in 
configs/sla.conf.sample.  Essentially, it is:


[line1]
type=trunk
device=Zap/3
autocontext=line1

[line2]
type=trunk
device=Zap/4
autocontext=line2

[station](!)
type=station
autocontext=sla_stations
trunk=line1
trunk=line2

[station1] (station)
device=SIP/station1

[station2](station)
device=SIP/station2

[station3](station)
device=SIP/station3


Thanks for providing some feedback on this.  You are the first one to 
say anything about it.  :)  I am very eager to get everything working 
well so that everyone is happy.  Just please be patient as I work 
through the initial flood of reports since it is just now getting out in

the field.

-- 
Russell Bryant
Software Engineer
Digium, Inc.
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Re: [asterisk-users] Read() status?

[Doug Garstang]

Yuan LIU wrote:
Does application Read() return a status?  Console displays stuff, but 
show application read doesn't mention any status variable.


Yuan Liu
I know that read() on a non-existent sound file will cause dial plan 
execution to abruptly stop (unlike background())... which is very bad imho.


Doug.

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Re: [asterisk-users] running error: load_modules: No 'modules.conf' found - vesrion 1.4.1 from svn

[TZieleniewski]

Hi,
I solved this issue.
I run asterisk with -C parameter and it worked.
So it seems that running configure with parameters changes the contents 
of the asterisk.conf file but doesn't change the
directory path where asterisk searches for the asterisk.conf file during 
the start.


Bests
Tomasz

tzieleniewski napisał(a):

Hi,

I have just installed the fresh svn version of asterisk and when I run it I get 
the following errors:

[Mar 4 14:19:27] WARNING[24527]: loader.c:728 load_modules: No 'modules.conf' 
found, no modules will be loaded.
[Mar 4 14:19:27] NOTICE[24527]: manager.c:2681 init_manager: Unable to open 
management configuration manager.conf. Call management disabled.
[Mar 4 14:19:27] NOTICE[24527]: cdr.c:1093 do_reload: CDR simple logging 
enabled.

I get this errors although my astetcdir contains the above considered files??
Does asterisk searched for those files in other directory than astetcdir??

the contents of my asterisk.conf:

[directories]
astetcdir => /home/asterisk/asterisk/asterisk
astmoddir => /home/asterisk/asterisk/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk

Bests
Tomasz
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Re: [asterisk-users] Asterisk Java w/ Threads

[Doug Garstang]

Jesus Mogollon wrote:
The best option would be to use AstManProxy and connect your event 
manager to it.

I tried this. Two problems...

The Asterisk Manager Proxy sends out a banner of 'Asterisk Manager 
Proxy/1.2' whereas the Asterisk-Java interface expects to see 'Asterisk 
Call Manager 1.0' (or something similar, the point is that the banner is 
different).


So, I changed what asterisk proxy manager sends in the source. This 
allowed Asterisk-Java to connect... and then the Proxy manager went and 
core dumped .


It won't work anyway. The proxy manager prefixes the name of the system 
to each line of output. The Asterisk-Java interface is not expecting 
this and will barf.


Doug.

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Re: [asterisk-users] Asterisk Java w/ Threads

[Doug Garstang]

Stefan Reuter wrote:

Jesus Mogollon wrote:
  

The best option would be to use AstManProxy and connect your event
manager to it.



why would adding a new system in between be better than directly
connecting to multiple Asterisk servers?

=Stefan
  
Simple. With the manager proxy in between, it does all the hard work of 
managing all the connections. It's what it's good at, and if it's doing 
it, I don't have to re-write all the thread management stuff again.


Doug.

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[asterisk-users] TDM400P/FXS in a HP DL380 G5

[James FitzGibbon]

The HP DL380 G5 (like many rack servers) has no AMP Mate-n-Lok connector
available to attach to a card that needs more power than the PCI bus can
provide, like the TDM400P when FXS modules are used.  HP has confirmed that
there is no part they sell to give you such a connector, and Digium says
their business edition folks got it to work, but only by doing nasty
warranty-voiding things to the internal wiring.

Has anyone figured out a solution for this?  Something along the lines of an
external power brick whose output attaches to a backplane slot and gives you
a 12V connector inside the server?

Or am I just SOL?

Thanks

--
j.
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Re: [asterisk-users] rtsavesysname not working in 1.4

[David Thomas]

Thanks again Bruce!

That was indeed the problem. I added displaysystemname=yes and it
started working.

Regards,
David
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[asterisk-users] How to disable MOH completely?

[David Thomas]

I need to disable MOH completely. We are using all SIP extensions and
do not want Asterisk to invoke MOH when flash or hold is pressed on
the phone.

Anyone know how to configure this?

Thanks!
David
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Re: [asterisk-users] rtsavesysname not working in 1.4

[Bruce Reeves]

David,

Here is what is working on my system, I added the following coulmn to the
sip table regserver and it is varchar(20) and then set the following items
in conf files.

asterisk.conf
systemname => server1

sip.conf
displaysystemname=yes <- Olle told me about this
rtsavesysname=yes

I bet the displaysystemname=yes is the missing setting, I seem to remeber
not getting anywhere till I added that.

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


On 3/2/07, Bruce Reeves <[EMAIL PROTECTED]> wrote:
> Try renaming you column in the peers table to regserver

Thanks for the suggestion Bruce, unfortunately it did not help. Any
other thoughts?

Does the systemname in asterisk.conf and regserver in field mysql need
to be an IP address, FQDN, hostname, or what is the proper format?

Regards,
David
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--
Bruce
Nortex Networks
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Re: [asterisk-users] Configurations Files of TE110P

[younss azzayani]

Hi,
this is the spin config of the Teleco modem:
RX - :  2
RX + : 1
Tx - : 5
Tx+:  4
what about those of TE110P & what I've to do?
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Re: [asterisk-users] rtsavesysname not working in 1.4

[David Thomas]

On 3/2/07, Bruce Reeves <[EMAIL PROTECTED]> wrote:

Try renaming you column in the peers table to regserver


Thanks for the suggestion Bruce, unfortunately it did not help. Any
other thoughts?

Does the systemname in asterisk.conf and regserver in field mysql need
to be an IP address, FQDN, hostname, or what is the proper format?

Regards,
David
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Re: [asterisk-users] Configurations Files of TE110P

[younss azzayani]

i do it & it doesn't work
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[asterisk-users] TC400B

[Wai Wu]

 
Anyone tried the digium TC400B transcoding card? What are your opinions?

Thnx
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Re: [asterisk-users] Double DTMF digits sent on IAX native bridge

[Remi Quezada]

Ok that makes sense, but I'm still getting double digits.  It seems to
me that the DTMF digit is getting detected too late.  When the digit is
pressed it seems like asterisk is passing the DTMF digit for a fraction
of a second through the audio path and then sends the digit for however
long your toneduration is set for.  I can hear this happening when I
dial the digits myself, I hear some kind sound being cut off for a
fraction of a second and then hear the DTMF tone pass.  So I guess this
is why sometimes some answer machines are detecting double digits. 

Russell Bryant wrote:
> Remi Quezada wrote:
>> I have two asterisk servers one is connected to the PSTN and the other
>> one is connected to SIP users.  The two servers connect with each other
>> using IAX.  When I have an incoming call from PSTN to the asterisk
>> servers and have a forward to go back out to the PSTN the two IAX
>> channel bridge together.  Now every time I dial a DTMF digit, the
>> asterisk is sending two DTMF digits.  I enable debugging for iax and I
>> do see it sending the DTMF digits two times.  Here is what I see:
>
> The IAX debug that you show below only shows one of each digit.  For
> each one, it shows Receiving the digit from one leg of the call, and
> then transmitting it out the other.  I have spaced out your debug to
> separate each digit.
>
> Each one shows ...
>
><- digit 
> - ACK ->
> - digit --->
> < ACK --
>
> which is exactly what is supposed to happen.
>
>
>> Rx-Frame Retry[ No] -- OSeqno: 018 ISeqno: 021 Type: DTMFSubclass: 1
>>Timestamp: 51523ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
>> Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 019 Type: IAX Subclass:
>> ACK  Timestamp: 51523ms  SCall: 2  DCall: 3
>> [192.168.15.201:4569]
>> Tx-Frame Retry[000] -- OSeqno: 019 ISeqno: 022 Type: DTMFSubclass: 1
>>Timestamp: 51543ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
>> Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 020 Type: IAX Subclass:
>> ACK  Timestamp: 51543ms  SCall: 4  DCall: 16385
>> [192.168.15.201:4569]
>
>
>> Rx-Frame Retry[ No] -- OSeqno: 019 ISeqno: 021 Type: DTMFSubclass: 2
>>Timestamp: 52083ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
>> Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 020 Type: IAX Subclass:
>> ACK  Timestamp: 52083ms  SCall: 2  DCall: 3
>> [192.168.15.201:4569]
>> Tx-Frame Retry[000] -- OSeqno: 020 ISeqno: 022 Type: DTMFSubclass: 2
>>Timestamp: 52103ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
>> Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 021 Type: IAX Subclass:
>> ACK  Timestamp: 52103ms  SCall: 4  DCall: 16385
>> [192.168.15.201:4569]
>
>
>> Rx-Frame Retry[ No] -- OSeqno: 020 ISeqno: 021 Type: DTMFSubclass: 3
>>Timestamp: 52663ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
>> Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 021 Type: IAX Subclass:
>> ACK  Timestamp: 52663ms  SCall: 2  DCall: 3
>> [192.168.15.201:4569]
>> Tx-Frame Retry[000] -- OSeqno: 021 ISeqno: 022 Type: DTMFSubclass: 3
>>Timestamp: 52683ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
>> Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 022 Type: IAX Subclass:
>> ACK  Timestamp: 52683ms  SCall: 4  DCall: 16385
>> [192.168.15.201:4569]
>
>
>> Rx-Frame Retry[ No] -- OSeqno: 021 ISeqno: 021 Type: DTMFSubclass: 4
>>Timestamp: 53223ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
>> Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 022 Type: IAX Subclass:
>> ACK  Timestamp: 53223ms  SCall: 2  DCall: 3
>> [192.168.15.201:4569]
>> Tx-Frame Retry[000] -- OSeqno: 022 ISeqno: 022 Type: DTMFSubclass: 4
>>Timestamp: 53243ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
>> Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 023 Type: IAX Subclass:
>> ACK  Timestamp: 53243ms  SCall: 4  DCall: 16385
>> [192.168.15.201:4569]
>
>
>> Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 021 Type: DTMFSubclass: 5
>>Timestamp: 53703ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
>> Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 023 Type: IAX Subclass:
>> ACK  Timestamp: 53703ms  SCall: 2  DCall: 3
>> [192.168.15.201:4569]
>> Tx-Frame Retry[000] -- OSeqno: 023 ISeqno: 022 Type: DTMFSubclass: 5
>>Timestamp: 53723ms  SCall: 16385  DCall: 4 [192.168.15.201:4569]
>> Rx-Frame Retry[ No] -- OSeqno: 022 ISeqno: 024 Type: IAX Subclass:
>> ACK  Timestamp: 53723ms  SCall: 4  DCall: 16385
>> [192.168.15.201:4569]
>
>
>> Rx-Frame Retry[ No] -- OSeqno: 023 ISeqno: 021 Type: DTMFSubclass: 6
>>Timestamp: 54163ms  SCall: 3  DCall: 2 [192.168.15.201:4569]
>> Tx-Frame Retry[-01] -- OSeqno: 021 ISeqno: 024 Type: IAX Subclass:
>> ACK  Timestamp: 54163ms  SCall: 2  DCall: 3
>> [192.168.15.201:4569]
>> Tx-Frame Retry[000] -- OSeqno: 024 ISeqno: 022 Type: DTMF   

Re: [asterisk-users] Configurations Files of TE110P

[Eric \"ManxPower\" Wieling]

In my experience usually you want to use a straight-thru cable, not a 
crossover cable.  Try a standard ethernet cable between TE110P and telco 
box.


younss azzayani wrote:

Hi,
the telco has given me a across cable (witch i put it on the modem so
the led modem( LOS Tx became Off : that's mean that the connection is
on)
so i taked this cable & i put it in my TE110P digium card, so the led
came green
but when i relayed TE110P to the modem the (led Modem turn of :mean
ok) & the TE110P became RED (mean not ok :-(  )
the teleco sayed that the delta channel is number 16( im using just TE110P)
hdb3, euroisdn
so what's the problem :( i'm realy lost

2007/3/5, Ioan Indreias <[EMAIL PROTECTED]>:

Hello,

Use the cross-over schema for creating a "self cross" connector.
Meaning you will connect your TX pair to your RX pair. This will be the
test of the physical layer of your card and the flashing red light of
the led will have to turn in green. Otherwise something is not
working/configured properly in your card.

Best regards,
## nini @ www.modulo.ro ##



younss azzayani wrote:
>> http://www.austechpartnerships.com/forum/viewtopic.php?t=76
> from the link you give me i see "RJ45 pins used for E1 is 1,2,4,5 -
> which you may need to cross (note you can't use standard CAT5 cables
> here for that)." what's that mean, is it mean that i have to use cable
> CAT6 or what exactly
>
> Can someone Give me A good schema of how to cross this cable
> for me
> 14
> 25
> 41
> 52
> but this doesn't work ,,, :-( :-( :-((

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Re: [asterisk-users] Configurations Files of TE110P

[younss azzayani]

Hi,
the telco has given me a across cable (witch i put it on the modem so
the led modem( LOS Tx became Off : that's mean that the connection is
on)
so i taked this cable & i put it in my TE110P digium card, so the led
came green
but when i relayed TE110P to the modem the (led Modem turn of :mean
ok) & the TE110P became RED (mean not ok :-(  )
the teleco sayed that the delta channel is number 16( im using just TE110P)
hdb3, euroisdn
so what's the problem :( i'm realy lost

2007/3/5, Ioan Indreias <[EMAIL PROTECTED]>:

Hello,

Use the cross-over schema for creating a "self cross" connector.
Meaning you will connect your TX pair to your RX pair. This will be the
test of the physical layer of your card and the flashing red light of
the led will have to turn in green. Otherwise something is not
working/configured properly in your card.

Best regards,
## nini @ www.modulo.ro ##



younss azzayani wrote:
>> http://www.austechpartnerships.com/forum/viewtopic.php?t=76
> from the link you give me i see "RJ45 pins used for E1 is 1,2,4,5 -
> which you may need to cross (note you can't use standard CAT5 cables
> here for that)." what's that mean, is it mean that i have to use cable
> CAT6 or what exactly
>
> Can someone Give me A good schema of how to cross this cable
> for me
> 14
> 25
> 41
> 52
> but this doesn't work ,,, :-( :-( :-((
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Re: [asterisk-users] Read() status?

[C F]

Yes. Use show application read in the cli

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

Does application Read() return a status?  Console displays stuff, but show
application read doesn't mention any status variable.

Yuan Liu


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Re: [asterisk-users] Problem with TE212P

[C F]

I agree with Tzafrir on this one. i have a digium dual span card that
starts at channel nine because i have two TDM400s that load first and
i have had no problems whatsoever with the D channel.

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

On Mon, Mar 05, 2007 at 11:00:36AM +0100, Benito Camelas wrote:
> Problem solved.
>
> Chris Hozian from Digium related me the problem:
>
>
> This problem is occurring because Asterisk expects to see the
> d-channel on every 16th channel.  This is being offset because your
> TDM2400P is being loaded first.

HUH??

The dchannel needs to be no. 16 in the span, and unrelated to its
general channel number, right?

>
> In order to fix this problem, make sure you that you are loading the
> kernel module for the TE212P before the TDM2400P.  Then you will need
> to reconfigure your /etc/zaptel.conf and /etc/asterisk/zapata.conf
> accordingly.
>
> Your zaptel.conf should contain the following.  Please keep in mind
> that this is only a snippet of the configuration file.
>
> --
>
> zaptel.conf snippet:
>
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> dchan=16
>
> span=2,0,0,ccs,hdb3,crc4
> bchan=32-46,48-62
> dchan=47
>
> # the following is assuming your TDM2400P has all FXO modules
> fxsks=63-86
>
> --
>
> You will need to modify your zapata.conf to reflect these channel range
> changes.
>
> In addition, please verify that the jumpers on the TE212P are set for E1
> mode.
>
>
>
> Now it works ok.
>
> Special thanks to Chris Hozian from Digium and Ioan Indreias from Modulo
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--
   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] Sangoma A102d and Asterisk on Debian 3.1.

[Markus Monka]

Hi,

try to set the TDMV_DCHAN = 16 (E1) or 24 (T1).

I had the same problem while updating from 2.3.4-(2|3) to 2.3.4-7 .
I think, that wancfg did not set this value correctly.

In my setup i updated an running system to new version, so the LinkLayer
is still ok.

Best Regards,
Markus


On 1/16/07, Klaus Darilion <[EMAIL PROTECTED]> wrote:

Erik Forsen wrote:
>
> Did you find any solution to this problem? I have the exact same problem
> with a Sangoma A102d card on debian 3.1, 2.6.19 and wanpipe 2.3.4-4.
> I've followed several different guides, including the one on sangoma's
> wiki. When I try to make a call out, I get this error:
>
> Jan 16 13:17:28] WARNING[18084]: app_dial.c:1081 dial_exec_full: Unable
> to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion)
>
> Also got the same SABME errors as you do.

Maybe you have link-layer problems - maybe you are using a wrong cable.

You can test the card using a E1 cross-over cable between the 2 ports of
the sangoma card.

regards
klaus

--
Klaus Darilion
nic.at

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Re: [asterisk-users] CTI

[Nuria Fernandez]

I'm implementing a TAPI driver to use with CTI-TAPI application. If you are
interesting vist activa.sourceforge.net

2006/11/28, Matt Florell <[EMAIL PROTECTED]>:


Have you looked at QueueMetrics?
http://queuemetrics.loway.it/

There are also several call center packages for Asterisk out there
that have all of the reports built into them that you want:
http://www.voip-info.org/wiki/view/Predictive+dialer

MATT---

On 11/28/06, Hernany Oliveira <[EMAIL PROTECTED]> wrote:
> I need something like Call Manager.
> I need to know how many agents is logged in, how many calls are on
queues,
> transfer calls, hang up calls, reports and so on.
> Everything related to a Call Center operation.
>
> I have been looking for and I did not find anything.
>
>
> -Mensagem original-
> De: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Em nome de Matt Florell
> Enviada em: segunda-feira, 27 de novembro de 2006 17:59
> Para: Asterisk Users Mailing List - Non-Commercial Discussion
> Assunto: Re: [asterisk-users] CTI
>
> CTI is a pretty broad term, what exactly do you need to do?
>
> Do you need to connect with another system?
>
> What features do you need?
>
> MATT---
>
> On 11/27/06, Hernany Oliveira <[EMAIL PROTECTED]> wrote:
> > Is there any cti for asterisk ??
> > Where may I download it ??
> >
> > Thanks in advance
> >
> > Hernany
> >
> >
> > --
> > No virus found in this outgoing message.
> > Checked by AVG Free Edition.
> > Version: 7.5.430 / Virus Database: 268.14.17/553 - Release Date:
> 27/11/2006
> > 04:00
> >
> >
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Re: [asterisk-users] Re: Problem with TE212P

[Tzafrir Cohen]

On Mon, Mar 05, 2007 at 11:00:36AM +0100, Benito Camelas wrote:
> Problem solved.
> 
> Chris Hozian from Digium related me the problem:
> 
> 
> This problem is occurring because Asterisk expects to see the
> d-channel on every 16th channel.  This is being offset because your
> TDM2400P is being loaded first.

HUH??

The dchannel needs to be no. 16 in the span, and unrelated to its
general channel number, right?

> 
> In order to fix this problem, make sure you that you are loading the
> kernel module for the TE212P before the TDM2400P.  Then you will need
> to reconfigure your /etc/zaptel.conf and /etc/asterisk/zapata.conf
> accordingly.
> 
> Your zaptel.conf should contain the following.  Please keep in mind
> that this is only a snippet of the configuration file.
> 
> --
> 
> zaptel.conf snippet:
> 
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15,17-31
> dchan=16
> 
> span=2,0,0,ccs,hdb3,crc4
> bchan=32-46,48-62
> dchan=47
> 
> # the following is assuming your TDM2400P has all FXO modules
> fxsks=63-86
> 
> --
> 
> You will need to modify your zapata.conf to reflect these channel range 
> changes.
> 
> In addition, please verify that the jumpers on the TE212P are set for E1 
> mode.
> 
> 
> 
> Now it works ok.
> 
> Special thanks to Chris Hozian from Digium and Ioan Indreias from Modulo
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-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] SMS ON ASTERISK

[Assis, Eduardo]

We installed na Asterisk System whith 400 Softphone users (Eyebeam 1.5 from
Counterpath).

As far as we know, Asterisk don't support yet IM (Instante Message)
feature,instead Eyebeam have this feature.

Is that true? Is there any new version from Asterisk that supports IM?

> Eduardo R. Assis
> Soluziona Ltda
> Consultor Sênior - TELECOM
> Al. Tocantins, 125 - 290 andar - Alphaville
> Barueri - São Paulo-SP - CEP 06.455-020
> E-mail. [EMAIL PROTECTED]
> Tel. +55 11-4197-0654
> Fax. +55 11-4197-0660
> Cel. +55 11-8577-0950
> www.soluziona.com.br  
> 
> 
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[asterisk-users] HITBSecConf2007 - Malaysia: Call for Papers now Open

[Praburaajan]

The CFP for HITBSecConf2007 - Malaysia is now open. HITBSecConf -
Malaysia is the premier network security event for the region and the
largest gathering of hackers in Asia. Our 2007 event is expected to
attract over 700 attendees from around the world and will see 4 keynote
speakers in addition to 40 deep-knowledge technical researchers
presenting over two-days.

Being a deep-knowledge technical conference, talks that are more
technical or that discuss new and never before seen attack methods are
of more interest than a subject that has been covered several times
before. Summaries not exceeding 250 words should be submitted (in plain
text format) to [EMAIL PROTECTED] for review and possible inclusion
in the programme.

Submissions are due no later than 1st May 2007.

Topics of interest include, but are not limited to the following:

# 3G/4G Cellular Networks
# SS7/Backbone telephony networks
# Analysis of network and security vulnerabilities
# Firewall technologies
# Intrusion detection
# Data Recovery and Incident Response
# GPRS and CDMA Security
# Identification and Entity Authentication
# Network Protocol and Analysis
# Smart Card Security
# Virus and Worms
# WLAN and Bluetooth Security
# Analysis of malicious code
# Applications of cryptographic techniques
# Analysis of attacks against networks and machines
# File system security
# Security in heterogeneous and large-scale environments

PLEASE NOTE:

We do not accept product or vendor related pitches. If your talk
involves an advertisement for a new product or service your company is
offering, please do not submit.

Your submission should include:

# Name, title, address, email and phone/contact number
# Draft of the proposed presentation (in PDF or PowerPoint format),
proof of concept for tools and exploits, etc.
# Short biography, qualification, occupation, achievement and
affiliations (limit 150 words).
# Summary or abstract for your presentation (limit 250 words)
# Time (45-60 minutes including time for discussion and questions)
# Technical requirements (video, internet, wireless, audio, etc.)

Each non-resident speaker will receive accommodation for 3 nights. For
each non-resident speaker, HITB will cover travel expenses (through our
airline partners, Malaysia Airlines) up to USD 1,000.00.

HITBSecConf2007 - Malaysia: The Largest Network Security Event in Asia!
http://conference.hitb.org/hitbsecconf2007kl/





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[asterisk-users] HITBSecConf2007 - Malaysia: Call for Papers now Open

[Praburaajan]

The CFP for HITBSecConf2007 - Malaysia is now open. HITBSecConf -
Malaysia is the premier network security event for the region and the
largest gathering of hackers in Asia. Our 2007 event is expected to
attract over 700 attendees from around the world and will see 4 keynote
speakers in addition to 40 deep-knowledge technical researchers
presenting over two-days.

Being a deep-knowledge technical conference, talks that are more
technical or that discuss new and never before seen attack methods are
of more interest than a subject that has been covered several times
before. Summaries not exceeding 250 words should be submitted (in plain
text format) to [EMAIL PROTECTED] for review and possible inclusion
in the programme.

Submissions are due no later than 1st May 2007.

Topics of interest include, but are not limited to the following:

# 3G/4G Cellular Networks
# SS7/Backbone telephony networks
# Analysis of network and security vulnerabilities
# Firewall technologies
# Intrusion detection
# Data Recovery and Incident Response
# GPRS and CDMA Security
# Identification and Entity Authentication
# Network Protocol and Analysis
# Smart Card Security
# Virus and Worms
# WLAN and Bluetooth Security
# Analysis of malicious code
# Applications of cryptographic techniques
# Analysis of attacks against networks and machines
# File system security
# Security in heterogeneous and large-scale environments

PLEASE NOTE:

We do not accept product or vendor related pitches. If your talk
involves an advertisement for a new product or service your company is
offering, please do not submit.

Your submission should include:

# Name, title, address, email and phone/contact number
# Draft of the proposed presentation (in PDF or PowerPoint format),
proof of concept for tools and exploits, etc.
# Short biography, qualification, occupation, achievement and
affiliations (limit 150 words).
# Summary or abstract for your presentation (limit 250 words)
# Time (45-60 minutes including time for discussion and questions)
# Technical requirements (video, internet, wireless, audio, etc.)

Each non-resident speaker will receive accommodation for 3 nights. For
each non-resident speaker, HITB will cover travel expenses (through our
airline partners, Malaysia Airlines) up to USD 1,000.00.

HITBSecConf2007 - Malaysia: The Largest Network Security Event in Asia!
http://conference.hitb.org/hitbsecconf2007kl/



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Re: [asterisk-users] Is the 1.0.X branch vulnerable to the SIP issue?

[Kevin P. Fleming]

Thermal Wetland wrote:
> We are still using 1.0.7 and did not see any patches for the 1.0.X branch.

Yes, it is, but we no longer provide patches (even for security issues)
for the 1.0 series.
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[asterisk-users] Re: DTMF detection problems on PRI channels?

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Michelle Dupuis <[EMAIL PROTECTED]> wrote:
> Sounds like the DTMF tones are too far from spec, or noisy.  Is the DTMF
> being transcoded somewhere along the way?
> 
> If you have time to killtry to separate the two frequencies in your
> software (I don't know goldwave) - are both present and clean and same
> amplitude and on freq?  Remove the two frequencies and what's left?  If
> there's a lot of noise, then the other party is doing a bad job encoding the
> DTMF.  Otherwise we can start to chase your machine causes

In the few examples I looked at recently, the audio appeared to be clean
and well-formed. Notching the two frequencies left very little.

The system is using E1 TDM interfaces with aLaw encoding.

I see there is a define called OLD_DSP_ROUTINES which causes conditional
compilation in dsp.c, and is defined or not in the Makefile. Is it worth
trying the old routines? What are the main differences between the old
and the new?

Cheers
Tony

-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] Re: Problem with TE212P

[Benito Camelas]

Problem solved.

Chris Hozian from Digium related me the problem:


This problem is occurring because Asterisk expects to see the
d-channel on every 16th channel.  This is being offset because your
TDM2400P is being loaded first.

In order to fix this problem, make sure you that you are loading the
kernel module for the TE212P before the TDM2400P.  Then you will need
to reconfigure your /etc/zaptel.conf and /etc/asterisk/zapata.conf
accordingly.

Your zaptel.conf should contain the following.  Please keep in mind
that this is only a snippet of the configuration file.

--

zaptel.conf snippet:

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47

# the following is assuming your TDM2400P has all FXO modules
fxsks=63-86

--

You will need to modify your zapata.conf to reflect these channel range changes.

In addition, please verify that the jumpers on the TE212P are set for E1 mode.



Now it works ok.

Special thanks to Chris Hozian from Digium and Ioan Indreias from Modulo
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Re: [asterisk-users] Configurations Files of TE110P

[Ioan Indreias]

Hello,

Use the cross-over schema for creating a "self cross" connector.
Meaning you will connect your TX pair to your RX pair. This will be the 
test of the physical layer of your card and the flashing red light of 
the led will have to turn in green. Otherwise something is not 
working/configured properly in your card.


Best regards,
## nini @ www.modulo.ro ##



younss azzayani wrote:

http://www.austechpartnerships.com/forum/viewtopic.php?t=76

from the link you give me i see "RJ45 pins used for E1 is 1,2,4,5 -
which you may need to cross (note you can't use standard CAT5 cables
here for that)." what's that mean, is it mean that i have to use cable
CAT6 or what exactly

Can someone Give me A good schema of how to cross this cable
for me
14
25
41
52
but this doesn't work ,,, :-( :-( :-((
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Re: [asterisk-users] new kernel and zaptel

[Tzafrir Cohen]

On Mon, Mar 05, 2007 at 10:21:10AM +0200, Giedrius Augys wrote:
> Hi,
> My older kernel was 2.6.18. Now I have compiled new kernel (2.6.20). Is it
> necessary to re-build zaptel drivers (I'm just using ztdummy).
> Thanks

Yes. 

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] new kernel and zaptel

[Giedrius Augys]

Hi,
My older kernel was 2.6.18. Now I have compiled new kernel (2.6.20). Is it
necessary to re-build zaptel drivers (I'm just using ztdummy).
Thanks
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