Re: [asterisk-users] X100P clone dial problems.

[Tzafrir Cohen]

On Mon, Dec 11, 2006 at 06:53:19PM +1100, Klaverstyn, David C wrote:
 I have since added fxs_ks=1 and channel = 1
 
 This has not fixed the problem.  I do notice a warning on the reload of
 asterisk.
 
 WARNING[4296]: chan_zap.c:10874 setup_zap: Ignoring signalling

Right. reload of chan_zap will only change settings of channels, but not
add new channels or change their signalling type.

You need to restart asterisk for that, sadly.

  asterisk -rx 'restart now'

(BTW: I'm not saying that Howard's example is not good. I'm trying to
give you a direct answer to your question with minimal voodoo values).

-- 
   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] X100P clone dial problems.

[Klaverstyn, David C]

Thanks for your help.

 

This is my file.

 

 

 

[channels]

 

language=au

context=from-pstn

signalling=fxo_ks

 

;rxwink=300

 

usecallerid=yes

hidecallerid=no

callwaiting=yes

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=yes

transfer=yes

canpark=yes

cancallforward=yes

callreturn=yes

echocancel=yes

echocancelwhenbridged=yes

rxgain=0.0

txgain=0.0

callgroup=1

pickupgroup=1

immediate=no

 

channel = 1 

 

 

 

 

 

 

Upon reloading asterisk I get the following errors.

 

 

Dec 11 19:03:45 WARNING[5265]: chan_zap.c:10874 setup_zap: Ignoring
signalling

Dec 11 19:03:45 ERROR[5265]: chan_zap.c:10305 setup_zap: Unable to
reconfigure channel '1'

 

 

 

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard
Lowndes
Sent: Monday, 11 December 2006 6:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] X100P clone dial problems.

 

 

 

Klaverstyn, David C wrote:

 I have since added fxs_ks=1

 

 

This is meaningless.  Follow the example that I posted.

 

 

 

  and channel = 1

 

 This has not fixed the problem.  I do notice a warning on the reload
of

 asterisk.

 

 WARNING[4296]: chan_zap.c:10874 setup_zap: Ignoring signalling

 

 -Original Message-

 From: [EMAIL PROTECTED]

 [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir

 Cohen

 Sent: Monday, 11 December 2006 4:47 PM

 To: asterisk-users@lists.digium.com

 Subject: Re: [asterisk-users] X100P clone dial problems.

 

 On Mon, Dec 11, 2006 at 04:18:20PM +1100, Klaverstyn, David C wrote:

 I'm not sure if I have a configuration problem or not. I am unable to

 dial out. When I try to dial in I can hear the phone ring on the

 dialling phone but Asterisk does not register anything. 

 

  

 

  

 

 In zaptel.conf I have

 

  

 

 loadzone = au

 

 defaultzone=au

 

 fxsks=1

 

  

 

 In zapata.conf

 

  

 

 language=au

 

 context=from-pstn

 

 

 Those need to be in the section [channels] and be followed by a

 

   channel = 1

 

 to actually have any effect. You also must set signaling (signalling =

 fxs_ks; in your case).

 

 

-- 

Howard.

LANNet Computing Associates - Your Linux people http://lannetlinux.com

When you want a computer system that works, just choose Linux;

When you want a computer system that works, just, choose Microsoft.

--

Flatter government, not fatter government; abolish the Australian
states.

 

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Re: [asterisk-users] Recommendations for QoS, PoE Switches

[Angel Heart]

Hi,

I am using Procurve Switches by HP for PoE.

http://www.hp.com/rnd/products/switches/ProCurve_Switch_3500yl-5400zl_Series/overview.htm?jumpid=reg_R1002_USEN

Aside from being a LIFETIME WARRANTY, I found them very easy to configure and 
install. 

Regards,

Angel


- Original Message 
From: Cory Andrews [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Sunday, December 10, 2006 10:35:13 PM
Subject: RE: [asterisk-users] Recommendations for QoS, PoE Switches


Zeeshan - I really like the Adtran Netvanta and/or Cisco Catalyst 3560 series 
switches if your customer has the pocket depth for them.  You are going to want 
a good, VLAN capable managed switch.  There are cheaper alternatives from 
Linksys, Netgear and DLink as well.
 
Cory Andrews




From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan Zakaria
Sent: Sunday, December 10, 2006 2:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Recommendations for QoS, PoE Switches


Hi all,

For a top quality setup, I will need to install high quality VoIP switches with 
QoS and PoE. My potential customer should not have any problem with call 
quality. Experienced folks, Please advice me what switches to install and at 
what price. I may need it for upto 100 phones. What else should I consider so 
that phones work without problem along with the computers on the same network? 
Phones will use their bridged ethernet connections, so that both computer and 
phones can work on the same connections. 

Thanks

-- 
Zeeshan A Zakaria 
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Yahoo! Music Unlimited
Access over 1 million songs.
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[asterisk-users] ParkAndAnnounce + Paging

[stefano.giuffredi]

[Sorry I re-send this message as I couldn't see it in the list. I hope it
will not come two times].

 

Hi everybody.

 

It is possible to announce the parking position through a paging to a group
of extensions?

I would like that when someone parks a call, some phones will announce with
the speaker the position.

 

Something like:

 

exten =
s,1,ParkAndAnnounce(call-parked-at:PARKED|30|PAGE(LOCAL/[EMAIL PROTECTED]
pageLOCAL/[EMAIL PROTECTED]|)

 

Is there a way, maybe with a different approach?

 

Thanks,

   Pol Po

 

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[asterisk-users] Asterisk with IM

[Mochamad Susantok]

Hi all,
Howto configure asterisk 1.2.13 (debian-base) with support Instant
Messaging, especially using client Xlite v.3.

Thanks


-
This email was sent using Student EEPIS-Webmail.
http://student.eepis-its.edu/

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[asterisk-users] OPS Protocol on Asterisk

[Dumpolid Exeplish]

hello everyone,
i have been researching into transnexus (http://www.transnexus.com/)
OSP (open settlement protocol) server. i am really interested in its
routing flextbility and call clearing capabilities. Has anyone
implemented OSP with Asterisk or Cisco voice devices. I would like to
have production enviroment ideas about this piece of software
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[asterisk-users] Waiting for dial tone in Dial cmd

[Administrator TOOTAI]

Morning,

we have gateways with FXO port registered as SIP endpoint in Asterisk. 
To be able to use this port, the gateway ask for prefix -lets say 9- 
then send dial tone and here the user enter the calling number. We want 
to cancel this step for the users so they can enter the entire number 
and Asterisk will deal with the gateway.


Does Asterisk have a possibility to manage this? We tried with 
Dial(SIP/exten,,D(0w12345678)) but unfortunately this doesn't work. We 
also tried with G option in dial cmd but we receive a busy back from GW 
in the second dial -which seems normal-.


FYI, dialing Dial(SIP/exten,,D(0)) give the dial tone, let the user 
enter the calling number and the call is passing smoothly.


Thanks for any hint


--
Daniel
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[asterisk-users] Re: X100P clone dial problems.

[Tony Mountifield]

In article [EMAIL PROTECTED],
Klaverstyn, David C [EMAIL PROTECTED] wrote:
 
 Thanks for your help.
 
 This is my file.
  
 
 [channels]
 
 language=au
 
 context=from-pstn
 
 signalling=fxo_ks

This should be: signalling=fxs_ks

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] Waiting for dial tone in Dial cmd

[Administrator TOOTAI]

Administrator TOOTAI a écrit :

[...]

FYI, dialing Dial(SIP/exten,,D(0)) give the dial tone, let the user 
enter the calling number and the call is passing smoothly.

Sorry, please read Dial(SIP/exten,,D(9))

--
Daniel
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Re: [asterisk-users] Waiting for dial tone in Dial cmd

[Anselm Martin Hoffmeister]

Am Montag, den 11.12.2006, 11:29 +0100 schrieb Administrator TOOTAI:
 Administrator TOOTAI a écrit :
  [...]
 
  FYI, dialing Dial(SIP/exten,,D(0)) give the dial tone, let the user 
  enter the calling number and the call is passing smoothly.
 Sorry, please read Dial(SIP/exten,,D(9))

Just an idea... Did you try with a M() Macro and SendDTMF() command
instead of D()? This would probably give you a more detailed control
over wait seconds and such.

Hth
Anselm

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[asterisk-users] Re: Xen, Asterisk ISDN: Timing Problems

[Arik Raffael Funke]

Thanks. What kernels do you use for dom0 and the domU's? Custom-built or 
out of the box?


- Arik


jason wrote:
I would vote RAM. I've been using a FXO card in xen for a good year now 
with no issues at all. In fact, my zttest timings are the same between 
xen and native.

Arik Raffael Funke wrote:

Hi,

is anybody running asterisk on a xen domU and can give an opinion on 
the following:


I have delegated a FritzCard and a HFC card to my domU and installed 
an asterisk setup that was running on the same isdn hardware but on a 
dedicated machine flawlessly.


I experienced what I believed to be timing problems: sometimes calls 
on the Fritzcard did not seem to reach asterisk, when calls were 
being made, sometimes they were horribly distorted. I quickly 
abandoned the project at the time for lack of time. I would now make 
another trial.


Can anybody tell me if the problems I was having were more likely to 
result from the fact that the isdn hardware was dedicated to the domU 
(i.e. maybe that produces some sort of bottleneck!?) or from too 
little ram allocated to my domU? (I believe I had 128 MB or so)


Thanks,
Arik

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Re: [asterisk-users] Waiting for dial tone in Dial cmd

[Administrator TOOTAI]

Anselm Martin Hoffmeister a écrit :

Am Montag, den 11.12.2006, 11:29 +0100 schrieb Administrator TOOTAI:
  

Administrator TOOTAI a écrit :


[...]

FYI, dialing Dial(SIP/exten,,D(0)) give the dial tone, let the user 
enter the calling number and the call is passing smoothly.
  

Sorry, please read Dial(SIP/exten,,D(9))



Just an idea... Did you try with a M() Macro and SendDTMF() command
instead of D()? This would probably give you a more detailed control
over wait seconds and such.
  
I love your ideas :-) As ususally, two brains are still thinking better 
then one ;-) Thanks a lot.


For archives, FXOexten being the FXO EndPoint in sip.conf Below 4000ms 
it's not working in our case. Perhaps something to do with early dial or 
others phone features.


exten = 300,1,dial(SIP/FXOexten,,M(WaitDialTone))

[macro-WaitDialTone]
exten = s,1,SendDTMF(9|4000)   ;sending PSTN prefix 
with a 4s timeout

exten = s,n,SendDTMF(CallingNumber) ;send number as DTMF

This does the trick.
--
Daniel
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[asterisk-users] Cannot find ptlib-config, installing 1.4-beta3

[Jan du Toit]

Hi

When trying to install asterisk1.4-beta3 I get the following error when running
./configure:

Cannot find ptlib-config - please install and try again

What is this ptlib-config? Can't seem to find it on google. Where can I find it
and how can I install it? Moreover do I really need it, can I force a bypass?

I have successfully installed zaptel 1.4.0-beta2 and libpri 1.4.0-beta1

Thanks.

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[asterisk-users] promotional info in music on hold

[Richard Soderblom]

Network Configurations
Block D, Surrey Park, Barham Road, Westville, 3610
Helpdesk: (086) 163-8266
Tel: (031) 266-1563
Fax: (031) 266-4206



Hi.

Is it possible to have asterisk insert various audio files into the
playback with the music on hold if they are holding on for an extension
or in a queue? Something like the following:

  |
  V
Welcome to ABC
  |
  V
Music on hold for 30s
  |
  V
Please remember to
  |
  V
More music on hold for 30s or so
  |
  V
More voice overs

And so on.

Could one put these files into a separate folder and the have asterisk
randomly play them back with the moh? It would be easy because then we
could just update the files every month or whenever we need to.

Thanks,
Richard
Best Regards

Richard Soderblom
Network Configurations
Cell: 
E-Mail: [EMAIL PROTECTED]



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[asterisk-users] Asterisk and Fax How To

[Noc Phibee]

Hi

i have a asterisk server with a Digium 4xE1 card connected to my local 
operator.


I am search a How to for :
  - Add a Mail to Fax server
  - Add a Fax to Mail Server

thanks bye


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[asterisk-users] Re: Recommendations for QoS, PoE Switches

[Benny Amorsen]

 ZZ == Zeeshan Zakaria [EMAIL PROTECTED] writes:

ZZ Switches should be Layer 2 or Layer 3, and what's the difference.

You really should hire someone to do the design.

ZZ Another question I have is about 10/100/1000 Mbps. In a standard
ZZ switch, ports don't actually work at 100 Mbps.

They don't? Perhaps you are referring to the bad old days when
switches didn't have enough backplane bandwidth to sustain full speed
on all ports. Those days are fortunately (almost) over.

ZZ But in a Netvanta grade switch, do all the ports work at 100 Mbps
ZZ at all the times, or do they come down to 10 Mbps?

Even if a switch gets overloaded, it doesn't switch port speed. It
just drops packets.


/Benny


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[asterisk-users] Power requirements on the TDM-400 card

[Gustavo Felisberto]

I have a TDM-400 from digium with 2FXO+2FXS ports. Any idea on how much power
will this drain from the 12 and 5 V connector when all ports are in use?

-- 
Gustavo Felisberto
(HumpBack)
Web: http://dev.gentoo.org/~humpback
Blog: http://blog.felisberto.net/

It's most certainly GNU/Linux, not Linux. Read more at
http://www.gnu.org/gnu/why-gnu-linux.html .
-



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Re: [asterisk-users] Asterisk and Fax How To

[Doug Lytle]

Noc Phibee wrote:

Hi

i have a asterisk server with a Digium 4xE1 card connected to my local 
operator.


I am search a How to for :
  - Add a Mail to Fax server
  - Add a Fax to Mail Server


http://iaxmode.sourceforge.net
http://hylafax.sourceforge.net

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 and Fax How To

[Doug Lytle]

Noc Phibee wrote:

I am search a How to for :
  - Add a Mail to Fax server
  - Add a Fax to Mail Server



Oooops, that should have been http://iaxmodem.sourceforge.net

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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[asterisk-users] Asterisk + Zap + CAS Signalling

[Mohammad Shokuie]

Hi folks,

I had a survey online but there i couldnt find a clean sample of CAS 
signalling on E1 interfaces. I defined a span with CAS framing and HDB3 line 
coding but dont know which signalling to use for channels. I'd use 3 bit CAS 
signalling and 20 incoming channels and 10 outgoing ones. Anyone could help 
me define the signalling for these channels.


PS. Im using Sangoma cards.

Any help would be highly appreciated.
---
M. Shokuie Nia.

_
Don't just search. Find. Check out the new MSN Search! 
http://search.msn.click-url.com/go/onm00200636ave/direct/01/


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Re: [asterisk-users] PCI, PCI-X and PCI-e -- Server / Interface Card Selection

[Aaron Daniel]

The Digium TE410P base card does indeed work in PCI-X slots.  We're
using two of the TE412P's in a PCI-X server with no problems :)

On Sun, 2006-12-10 at 00:10 -0500, Time Bandit wrote:
I can't risk spending a few thousand just to reach the
conclusion that Digium's PRI or BRI cards do not work
with a particular system's PCI-X slots/bus... Or, worse,
staying with a dead card / system board in my hands ! :-(
 
Anyone ?
 I don't know about Digium cards, but I just installed a Sangoma A101
 card into an IBM server in a PCI-X slot and it is working perfectly.
 
 You should ask Digium
 
 hth
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-- 
Aaron Daniel
Senior Voice Analyst
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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[asterisk-users] OSP peering VOIP servers

[Dumpolid Exeplish]

Hi,
in addition to my previous post about the OSP support on Asterisk,
does anyone know if there existst OSP peering VOIP hosts who are
willing to connect to simple users like me using OSP protocol
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Re: [asterisk-users] Re: Recommendations for QoS, PoE Switches

[Zeeshan Zakaria]

What's the price for these HP switches?

And also I someone can give me a link to some document where I can read
about Layer 2 and Layer 3, how they help in VoIP traffic, it'll be helpful.
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RE: [asterisk-users] Re: Recommendations for QoS, PoE Switches

[Cory Andrews]

Zeeshan - understanding the Cisco OSI model will help you
conceptualize.http://www.cisco.com/univercd/cc/td/doc/cisintwk/ito_d
oc/introint.htm
 
There is a good graphic depiction here
http://www.certificationzone.com/cisco/images/graphics/VP/IPTT/WP1/VP-IP
TT-WP1-01.gif  Using this image, starting at the bottom you have the
physical layer, or Layer 1, Data Link or Layer 2, Network or Layer
3, etc
 
 
 
Cory Andrews
 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan
Zakaria
Sent: Monday, December 11, 2006 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Recommendations for QoS, PoE Switches


What's the price for these HP switches?

And also I someone can give me a link to some document where I can read
about Layer 2 and Layer 3, how they help in VoIP traffic, it'll be
helpful. 
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Re: [asterisk-users] Re: Recommendations for QoS, PoE Switches

[Patrick May]

On Mon, Dec 11, 2006 at 09:53:53AM -0500, Zeeshan Zakaria wrote:
 What's the price for these HP switches?
 
 And also I someone can give me a link to some document where I can read
 about Layer 2 and Layer 3, how they help in VoIP traffic, it'll be helpful.

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CDW's retail price was about $7,000.



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RE: [asterisk-users] Re: Recommendations for QoS, PoE Switches

[Cory Andrews]

The guy in the UK who bought on Ebay is threatening to buy 2 units 


Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice direct - 716.250.3402
fax - 716.630.1548
e - [EMAIL PROTECTED]
m - 716.907.4059
aim - B2Cory

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
May
Sent: Monday, December 11, 2006 10:22 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Re: Recommendations for QoS, PoE Switches

On Mon, Dec 11, 2006 at 09:53:53AM -0500, Zeeshan Zakaria wrote:
 What's the price for these HP switches?
 
 And also I someone can give me a link to some document where I can 
 read about Layer 2 and Layer 3, how they help in VoIP traffic, it'll
be helpful.

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CDW's retail price was about $7,000.

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RE: [asterisk-users] Re: Recommendations for QoS, PoE Switches

[Cory Andrews]

 Edgewater Networks markets a 24 port switch, with PoE (both Cisco CDP
and 802.3af supported), and Layer 2/3 management features that retails
for less than $1500.  The model is EC-2402POE-01


Cory Andrews

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
May
Sent: Monday, December 11, 2006 10:22 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Re: Recommendations for QoS, PoE Switches

On Mon, Dec 11, 2006 at 09:53:53AM -0500, Zeeshan Zakaria wrote:
 What's the price for these HP switches?
 
 And also I someone can give me a link to some document where I can 
 read about Layer 2 and Layer 3, how they help in VoIP traffic, it'll
be helpful.

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CDW's retail price was about $7,000.

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RE: [asterisk-users] Asterisk and Fax How To

[Michelle Dupuis]

Check out www.generationd.com for a couple of useful scripts (fax2mail and
mail2fax).  If I interpret your question properly, you looking for scripts.
If in fact you are looking for sendmail/libtiff help, have a search through
the archives.

MD 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noc Phibee
Sent: Monday, December 11, 2006 8:01 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk and Fax How To

Hi

i have a asterisk server with a Digium 4xE1 card connected to my local 
operator.

I am search a How to for :
   - Add a Mail to Fax server
   - Add a Fax to Mail Server

thanks bye


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Recall: [asterisk-users] Re: Recommendations for QoS, PoE Switches

[Cory Andrews]

Cory Andrews would like to recall the message, [asterisk-users] Re: 
Recommendations for QoS, PoE Switches.
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Re: [asterisk-users] Power requirements on the TDM-400 card

[Bob Chiodini]

Gustavo,

Take a look at this thread

http://lists.digium.com/pipermail/asterisk-users/2006-October/169627.html

Presumably the supplemental 12v supply is for ringing voltage.

I did not see anything on Digium's support pages about the card itself.
Maybe a call to tech support may help.

Bob...

On Mon, 2006-12-11 at 13:09 +, Gustavo Felisberto wrote:
 I have a TDM-400 from digium with 2FXO+2FXS ports. Any idea on how much power
 will this drain from the 12 and 5 V connector when all ports are in use?
 
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Re: [asterisk-users] Asterisk from Debian Packages

[Carlos Navarro]

On Sun, 10 Dec 2006 20:54:10 -0500
Paul [EMAIL PROTECTED] wrote:
 If you run etch before it is released as stable, you might run into
 problems that are over your head. I have run into a few that weren't
 over my head but they were very inconvenient.

Yes Paul, I'm running 2 etch with asterisk, but it is my own risk.
In Debian I trust.

Charlie
  
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[asterisk-users] How to manipulate FROM header on Asterisk-DIALPLAN

[Ricardo Martins]

Hi all! Do anybody knows any asterisk-dialplan function that can replace 
the username portion of FROM header on an INVITE SIP message that is 
being handled by asterisk?


Thanks in advance for any tiny clue.

Rgds, Ricardo Martins.
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[asterisk-users] asterisk PLAR

[Jeronimo Romero]

Does anyone know if asterisk supports PLAR (Private Line Auto Ringdown).
The Oreilly (Asterisk: Future of Telephony) book mentions it in passing
saying that all you need to enable it is to set immediate=yes in
zapata.conf. Has anyone implemented this in brokerage trading
environments?

 

Thanks in advance. 

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[asterisk-users] CLI History

[Douglas Garstang]

What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, 
the last command in the history always defaults to 'stop now'. This is very 
bad, and it's caused accidental shutdowns more than once.

Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399)
Verbosity is at least 3
hera*CLI A
No such command 'A' (type 'help' for help)
hera*CLI B
No such command 'B' (type 'help' for help)
hera*CLI C
No such command 'C' (type 'help' for help)
hera*CLI D
No such command 'D' (type 'help' for help)
hera*CLI E
No such command 'E' (type 'help' for help)
hera*CLI 
[10:[EMAIL PROTECTED](pbx3):~]# asterisk -trv
Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer [EMAIL PROTECTED]
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=
Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399)
Verbosity is at least 3
hera*CLI stop now -- I pressed the UP arrow upon re-entering the console!

Doug


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[asterisk-users] FW: [asterisk-dev] Kernel crash during modprobe wfxco

[Roman Marchevsky]

1.I do not have access to console because my servers are in collocation
space, but technician from collocation told me that he is seeing E711 PCI
ERR Slot #1 which in the PowerEdge 1950 manual means The system BIOS has
reported PCI system error on a component that resides in specified slot.

2. I am using 2.6.9-42.0.3.ELsmp kernel


Roman Marchevsky

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Sunday, December 10, 2006 3:24 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Kernel crash during modprobe wfxco

[ This question best be asked on asterisk-users, please follow-up there ]

On Sun, Dec 10, 2006 at 01:43:10PM -0600, Roman Marchevsky wrote:
  
 
 I am using Power Edge 1950 with CenyOS4.1. Since I need MeetMe I got
 DigiNetwork X100P clone. I compiled zaptel without any problem, but on
 modprobe of wcfxo module kernel crashes. 

What do you mean by crashed? What exactly do you see?

If this is oops or kernel panic: do you have the claas trace?

 I tried different versions of
 zaptel as well as different servers with the same configuration of
 hardware/software and problem persist.  Does anybody have the same
problem?
 
 Is there any known solution for the problem?


Which kernel version do you have? 

uname -r

-- 
   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] CLI History

[Dave Cotton]

On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote:
 What's wrong with the Asterisk CLI history? When I exit the CLI, and 
 re-enter, the last command in the history always defaults to 'stop now'. This 
 is very bad, and it's caused accidental shutdowns more than once.

Nothing wrong here. 
-- 
Dave Cotton [EMAIL PROTECTED]

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Re: [asterisk-users] New installation CentOS 4 x86 or X86_64

[Carla Schroder]

On Sunday 10 December 2006 11:18 pm, Remco Barendse wrote:
 Hi list!

 I have to do a new bare metal installation of a box running Asterisk with
 bristuff or vzaphfc.

 The box will be used as a really lightly loaded file server and pbx.

 Any advise on which architecture I should use? The cpu is a 64 bit capable
 AMD (the box is running x86_64 now) but is still suffering from echo on
 the BRI lines.

 Should I go with the normal x86 or the 64 bit x86_64 arch.?


x86-32 isn't as fun as x86_64, but it's fewer hassles. With 64-bit systems 
you'll run into the odd app or driver that hasn't been ported to 64-bit 
architectures yet. You can run 32-bit code on 64-bit systems in chroots, 
which I think is a horrid pain, but some folks don't mind. :)

The big advantage of a 64-bit system is being able to handle huge amounts of 
memory (over 4 gigabytes) and gigantic files (up to 4 exabytes, wheee!), 
which doesn't really apply to an Asterisk server.
-- 
~
Carla Schroder
Linux geek and random computer tamer
check out my Linux Cookbook! 
http://www.oreilly.com/catalog/linuxckbk/
best book for sysadmins and power users
~
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RE: [asterisk-users] Re: Recommendations for QoS, PoE Switches

[William McCloskey]

http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l
=enoc=pct3448poe-sapps=bsd

Dell make a nice Poe switch. I've got 20 some odd Cisco 7940G's running
on it at the moment.

-
 William J McCloskey

 Information Technology Manager 
 [EMAIL PROTECTED]
 503-827-8141
 503-228-6747 fax
 www.timbercon.com
-
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cory
Andrews
Sent: Monday, December 11, 2006 7:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: Recommendations for QoS, PoE Switches

The guy in the UK who bought on Ebay is threatening to buy 2 units 


Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice direct - 716.250.3402
fax - 716.630.1548
e - [EMAIL PROTECTED]
m - 716.907.4059
aim - B2Cory

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
May
Sent: Monday, December 11, 2006 10:22 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Re: Recommendations for QoS, PoE Switches

On Mon, Dec 11, 2006 at 09:53:53AM -0500, Zeeshan Zakaria wrote:
 What's the price for these HP switches?
 
 And also I someone can give me a link to some document where I can 
 read about Layer 2 and Layer 3, how they help in VoIP traffic, it'll
be helpful.

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http://lists.digium.com/mailman/listinfo/asterisk-users

CDW's retail price was about $7,000.

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Re: [asterisk-users] Power requirements on the TDM-400 card

[olivier.taylor]

I use a a400p(tdm400p clone) on a soekris, 2 fxs and 2 fxo, some
soldering needed but the Soekris power supply is enough.
Only the fxs need power, Fxo doesn't.

18v 800 ma

hope it could help

Olivier

Bob Chiodini a crit:

  Gustavo,

Take a look at this thread

http://lists.digium.com/pipermail/asterisk-users/2006-October/169627.html

Presumably the supplemental 12v supply is for ringing voltage.

I did not see anything on Digium's support pages about the card itself.
Maybe a call to tech support may help.

Bob...

On Mon, 2006-12-11 at 13:09 +, Gustavo Felisberto wrote:
  
  
I have a TDM-400 from digium with 2FXO+2FXS ports. Any idea on how much power
will this drain from the 12 and 5 V connector when all ports are in use?

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[asterisk-users] Re: good Linux references

[olivier.taylor]

Strange idea to switch from freebsd to another OS, Freebsd is very 
stable with asterisk, I must say, rock solid...

What's the reason?

Olivier


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RE: [asterisk-users] CLI History

[Douglas Garstang]

 -Original Message-
 From: Dave Cotton [mailto:[EMAIL PROTECTED]
 Sent: Monday, December 11, 2006 10:55 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] CLI History
 
 
 On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote:
  What's wrong with the Asterisk CLI history? When I exit the 
 CLI, and re-enter, the last command in the history always 
 defaults to 'stop now'. This is very bad, and it's caused 
 accidental shutdowns more than once.
 
 Nothing wrong here. 

Can you possibly be a little more specific on why it isn't a problem?

Doug.
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Re: [asterisk-users] Repeated Digits

[William Piper]

I've also had these problems. If the call is going between two Asterisk
servers, connect them with dtmf=info. That solved my problems.

bp


On 12/10/06, Forrest Beck [EMAIL PROTECTED] wrote:


I too have seen this.  I have to press the digits just right.  I have
tried RFC2833, and Inband to send the digits.

If I find the problem, I will let you know.

On 12/8/06, Gustavo Flores [EMAIL PROTECTED] wrote:
 Hi,

 Have anyone experience repeated digits when connecting a call from SIP
and
 terminating it to a PRI Channel? On the other side of the PRI Channel is
an
 IVR that expect a pin but the digits come repeated. For example, you
dial
 12345 but it is received as 12224445

 --
 Gustavo Flores
 IT Manager
 IAS FILM Corp.


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RE: [asterisk-users] CLI History

[Jonathan k. Creasy]

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Douglas Garstang
 Sent: Monday, December 11, 2006 1:37 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] CLI History
 
  -Original Message-
  From: Dave Cotton [mailto:[EMAIL PROTECTED]
  Sent: Monday, December 11, 2006 10:55 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] CLI History
 
 
  On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote:
   What's wrong with the Asterisk CLI history? When I exit the
  CLI, and re-enter, the last command in the history always
  defaults to 'stop now'. This is very bad, and it's caused
  accidental shutdowns more than once.
 
  Nothing wrong here.
 
 Can you possibly be a little more specific on why it isn't a problem?
 
 Doug.
 ___

Sounds like it is working as intended if that is the last command you
executed. I'd say be more careful when executing commands. 
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Re: [asterisk-users] Re: How to communicated Both SIP and IAX2 each other?

[Pavel Jezek]

nobody knows, how jitterbuffer actually working when asterisk doing 
protocol translation? i.e. sip-iax, skinny-iax...

how current two jb implementations (generic rtp  iax jb) working together?
PJ




Pavel Jezek wrote:
so that, jitterbuffer should be enabled  forced on sip and iax 
channel on asterisk (because UAs have no knowledge about jitter on 
opposite link), from first example?


UA(sip)---OpenSER-- Asterisk-- UA(IAX2)




Steven wrote:

Nothing is end to end in this case.

It is two separate sessions, one SIP and one iax.



  

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Re: [asterisk-users] asterisk PLAR

[Eric \ManxPower\ Wieling]

Jeronimo Romero wrote:

Does anyone know if asterisk supports PLAR (Private Line Auto Ringdown).
The Oreilly (Asterisk: Future of Telephony) book mentions it in passing
saying that all you need to enable it is to set immediate=yes in
zapata.conf. Has anyone implemented this in brokerage trading
environments?


Set immediate=yes for that FXS channel.  When the phone goes off hook a 
call will be generated to exten = s in the context that the channel is in.


If you are using SIP then you must configure this in the SIP device.

I can't imagine that a brokerage enviroment is any different from any 
other enviroment that needs PLAR.

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Re: [asterisk-users] CLI History

[Carla Schroder]

On Monday 11 December 2006 9:31 am, Douglas Garstang wrote:
 What's wrong with the Asterisk CLI history? When I exit the CLI, and
 re-enter, the last command in the history always defaults to 'stop now'.
 This is very bad, and it's caused accidental shutdowns more than once.

 Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399)
 Verbosity is at least 3
 hera*CLI A
 No such command 'A' (type 'help' for help)
 hera*CLI B
 No such command 'B' (type 'help' for help)
 hera*CLI C
 No such command 'C' (type 'help' for help)
 hera*CLI D
 No such command 'D' (type 'help' for help)
 hera*CLI E
 No such command 'E' (type 'help' for help)
 hera*CLI
 [10:[EMAIL PROTECTED](pbx3):~]# asterisk -trv
 Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
 Created by Mark Spencer [EMAIL PROTECTED]
 Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for
 details. This is free software, with components licensed under the GNU
 General Public License version 2 and other licenses; you are welcome to
 redistribute it under certain conditions. Type 'show license' for details.
 =
 Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399)
 Verbosity is at least 3
 hera*CLI stop now -- I pressed the UP arrow upon re-entering the console!


 I'm a bit confused by your example. What are A,B,C, etc? To exit the Asterisk 
console, I type 'exit'. Asterisk continues to run, as it should. To re-enter 
the console I use asterisk -rvvv.

-- 
~
Carla Schroder
Linux geek and random computer tamer
check out my Linux Cookbook! 
http://www.oreilly.com/catalog/linuxckbk/
best book for sysadmins and power users
~
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[asterisk-users] Re: Xen, Asterisk ISDN: Timing Problems

[Arik Raffael Funke]

That has been fixed in the current Xen, and as far as I can tell works 
without problems. (At least for some NICs I had dedicated to another domU.)


Regards,
Arik


Howard Lowndes wrote:
I have to run Asterisk on the dom0 host as earlier versions of Xen had 
problems handing PCI control over to a domU kernel.  Does anyone know if 
this has been fixed yet?



Arik Raffael Funke wrote:
Thanks. What kernels do you use for dom0 and the domU's? Custom-built 
or out of the box?


- Arik


jason wrote:
I would vote RAM. I've been using a FXO card in xen for a good year 
now with no issues at all. In fact, my zttest timings are the same 
between xen and native.

Arik Raffael Funke wrote:

Hi,

is anybody running asterisk on a xen domU and can give an opinion on 
the following:


I have delegated a FritzCard and a HFC card to my domU and installed 
an asterisk setup that was running on the same isdn hardware but on 
a dedicated machine flawlessly.


I experienced what I believed to be timing problems: sometimes calls 
on the Fritzcard did not seem to reach asterisk, when calls were 
being made, sometimes they were horribly distorted. I quickly 
abandoned the project at the time for lack of time. I would now make 
another trial.


Can anybody tell me if the problems I was having were more likely to 
result from the fact that the isdn hardware was dedicated to the 
domU (i.e. maybe that produces some sort of bottleneck!?) or from 
too little ram allocated to my domU? (I believe I had 128 MB or so)


Thanks,
Arik

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RE: [asterisk-users] asterisk PLAR

[Alexander Lopez]

It can be configured and DOES work with ZAP channels.

 

 If you are looking to use IP based devices your Mileage may vary from
Hybrid to Sherman Tank.

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo
Romero
Sent: Monday, December 11, 2006 12:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk PLAR

 

Does anyone know if asterisk supports PLAR (Private Line Auto Ringdown).
The Oreilly (Asterisk: Future of Telephony) book mentions it in passing
saying that all you need to enable it is to set immediate=yes in
zapata.conf. Has anyone implemented this in brokerage trading
environments?

 

Thanks in advance. 

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[asterisk-users] IAX2 to SIP protocol translation overhead?

[David Thomas]

Just wondering if there is much CPU overhead in the translation from
IAX2 to SIP, and how taxing this function is as compared to
transcoding.

We're trying to build an efficient system and would like to avoid
taxing the CPU as much as possible. Our upstream service provider is
100% SIP, however we'd like to use IAX2 in our network as well, if it
does not cause too much overhead.

Not sure if it matters, but we will be running aprox 100 simultaneous calls.

Thanks,
David
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RE: [asterisk-users] CLI History

[Jonathan k. Creasy]

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Carla Schroder
 Sent: Monday, December 11, 2006 2:17 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] CLI History
 
 On Monday 11 December 2006 9:31 am, Douglas Garstang wrote:
  What's wrong with the Asterisk CLI history? When I exit the CLI, and
  re-enter, the last command in the history always defaults to 'stop
now'.
  This is very bad, and it's caused accidental shutdowns more than
once.
 
  Connected to Asterisk 1.2.9.1 currently running on hera (pid =
17399)
  Verbosity is at least 3
  hera*CLI A
  No such command 'A' (type 'help' for help)
  hera*CLI B
  No such command 'B' (type 'help' for help)
  hera*CLI C
  No such command 'C' (type 'help' for help)
  hera*CLI D
  No such command 'D' (type 'help' for help)
  hera*CLI E
  No such command 'E' (type 'help' for help)
  hera*CLI
  [10:[EMAIL PROTECTED](pbx3):~]# asterisk -trv
  Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
  Created by Mark Spencer [EMAIL PROTECTED]
  Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for
  details. This is free software, with components licensed under the
GNU
  General Public License version 2 and other licenses; you are welcome
to
  redistribute it under certain conditions. Type 'show license' for
 details.
 


=
  Connected to Asterisk 1.2.9.1 currently running on hera (pid =
17399)
  Verbosity is at least 3
  hera*CLI stop now -- I pressed the UP arrow upon re-entering the
 console!
 
 

Mine appears to work: 

##Connected to Asterisk and execute stop now: 

dragon*CLI stop now
dragon*CLI
Disconnected from Asterisk server

## Restarted Asterisk: 

[EMAIL PROTECTED] ~]# asterisk -p

## Connected to Asterisk then ran exit: 

[EMAIL PROTECTED] ~]# asterisk -r
Asterisk 1.4.0-beta3, Copyright (C) 1999 - 2006 Digium, Inc. and
others.
Created by Mark Spencer [EMAIL PROTECTED]
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty'
for details.
This is free software, with components licensed under the GNU
General Public
License version 2 and other licenses; you are welcome to
redistributeit under
certain conditions. Type 'show license' for details.

==
===
Connected to Asterisk 1.4.0-beta3 currently running on dragon
(pid =  32521)
dragon*CLI exit

## Connected to Asterisk Again and hit the up arrow:

[EMAIL PROTECTED] ~]# asterisk -r
Asterisk 1.4.0-beta3, Copyright (C) 1999 - 2006 Digium, Inc. and
others.
Created by Mark Spencer [EMAIL PROTECTED]
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty'
for details.
This is free software, with components licensed under the GNU
General Public
License version 2 and other licenses; you are welcome to
redistributeit under
certain conditions. Type 'show license' for details.

==
===
Connected to Asterisk 1.4.0-beta3 currently running on dragon
(pid =  32521)
dragon*CLI exit

Exit is displayed not stop now. If you hit A and it's an invalid
command...maybe that is your problem...
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Re: [asterisk-users] CLI History

[Aaron Daniel]

On Mon, 2006-12-11 at 11:17 -0800, Carla Schroder wrote:
 On Monday 11 December 2006 9:31 am, Douglas Garstang wrote:
  What's wrong with the Asterisk CLI history? When I exit the CLI, and
  re-enter, the last command in the history always defaults to 'stop now'.
  This is very bad, and it's caused accidental shutdowns more than once.
 
  Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399)
  Verbosity is at least 3
  hera*CLI A
  No such command 'A' (type 'help' for help)
  hera*CLI B
  No such command 'B' (type 'help' for help)
  hera*CLI C
  No such command 'C' (type 'help' for help)
  hera*CLI D
  No such command 'D' (type 'help' for help)
  hera*CLI E
  No such command 'E' (type 'help' for help)
  hera*CLI
  [10:[EMAIL PROTECTED](pbx3):~]# asterisk -trv
  Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
  Created by Mark Spencer [EMAIL PROTECTED]
  Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for
  details. This is free software, with components licensed under the GNU
  General Public License version 2 and other licenses; you are welcome to
  redistribute it under certain conditions. Type 'show license' for details.
  =
  Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399)
  Verbosity is at least 3
  hera*CLI stop now -- I pressed the UP arrow upon re-entering the console!
 
 
  I'm a bit confused by your example. What are A,B,C, etc? To exit the 
 Asterisk 
 console, I type 'exit'. Asterisk continues to run, as it should. To re-enter 
 the console I use asterisk -rvvv.
 
He was demonstrating how the CLI history shows stop now as the last
command (which um... it's a history?  you're last command is gonna be
the um... last command you ran... i.e. stop now).

Douglas,  why're you even running stop now on a live production
server.  If you're not, quit complaining and watch what you type before
hitting enter.
-- 
Aaron Daniel
Senior Voice Analyst
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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Re: [asterisk-users] Re: How to communicated Both SIP and IAX2 each other?

[Andrew Joakimsen]

The protocol does not matter. If jitterbuffer is off then asterisk gets the
packets and sends them to the IAX clients without jitterbuffer just as if it
was another SIP client w/o jb.

On 12/8/06, Pavel Jezek [EMAIL PROTECTED] wrote:


so that, jitterbuffer should be enabled  forced on sip and iax channel
on asterisk (because UAs have no knowledge about jitter on opposite
link), from first example?

UA(sip)---OpenSER-- Asterisk-- UA(IAX2)




Steven wrote:
 Nothing is end to end in this case.

 It is two separate sessions, one SIP and one iax.




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[asterisk-users] re: L option in dial command

[Yair Hakak]

Hello all,
I'm having a bit for a problem with the dial command limit option. I have
the following dial command (executed from inside the a2billing agi)

AGI Script Executing Application: (Dial) Options: (
IAX2/[EMAIL PROTECTED]/18005551212|30|HL(6:2:0)0)

Now, from what i read in the wiki, this is supposed to limit me to one
minute (6 ms), and warn me when there are 20 seconds left.

Instead, it hangs up after 40 seconds.

i understand there was an open bug about this...is it fixed, is there a
patch, what can i do about this?

for reference version is:
Asterisk 1.2.13 built by root @ phone2 on a i686 running Linux on 2006-10-28
10:51:43 UTC

any help is appreciated
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RE: [asterisk-users] CLI History

[Douglas Garstang]

 -Original Message-
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]
 Sent: Monday, December 11, 2006 12:57 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] CLI History
 
 
 On Mon, 2006-12-11 at 11:17 -0800, Carla Schroder wrote:
  On Monday 11 December 2006 9:31 am, Douglas Garstang wrote:
   What's wrong with the Asterisk CLI history? When I exit 
 the CLI, and
   re-enter, the last command in the history always defaults 
 to 'stop now'.
   This is very bad, and it's caused accidental shutdowns 
 more than once.
  
   Connected to Asterisk 1.2.9.1 currently running on hera 
 (pid = 17399)
   Verbosity is at least 3
   hera*CLI A
   No such command 'A' (type 'help' for help)
   hera*CLI B
   No such command 'B' (type 'help' for help)
   hera*CLI C
   No such command 'C' (type 'help' for help)
   hera*CLI D
   No such command 'D' (type 'help' for help)
   hera*CLI E
   No such command 'E' (type 'help' for help)
   hera*CLI
   [10:[EMAIL PROTECTED](pbx3):~]# asterisk -trv
   Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. 
 and others.
   Created by Mark Spencer [EMAIL PROTECTED]
   Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show 
 warranty' for
   details. This is free software, with components licensed 
 under the GNU
   General Public License version 2 and other licenses; you 
 are welcome to
   redistribute it under certain conditions. Type 'show 
 license' for details.
   
 ==
 ===
   Connected to Asterisk 1.2.9.1 currently running on hera 
 (pid = 17399)
   Verbosity is at least 3
   hera*CLI stop now -- I pressed the UP arrow upon 
 re-entering the console!
  
  
   I'm a bit confused by your example. What are A,B,C, etc? 
 To exit the Asterisk 
  console, I type 'exit'. Asterisk continues to run, as it 
 should. To re-enter 
  the console I use asterisk -rvvv.
  
 He was demonstrating how the CLI history shows stop now as the last
 command (which um... it's a history?  you're last command is gonna be
 the um... last command you ran... i.e. stop now).

For crying out loud, why is this so hard to understand? It isn't rocket 
science. I said that when I exit the CLI and re-enter, no matter what my 
previous set of commands was, when I hit the UP arrow key, it was always 'stop 
now'. 'Stop now' WAS NOT MY PREVIOUS COMMAND.

For the person that suggested maybe unknown commands are not added to the 
history...

hera*CLI show channels
Channel  Location State   Application(Data) 
0 active channels
0 active calls
hera*CLI sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last 
Message   
0 active SIP channels
hera*CLI 
(I Pressed Ctrl-c here)

[13:[EMAIL PROTECTED](pbx3):~]# asterisk -trv
Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer [EMAIL PROTECTED]
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=
Connected to Asterisk 1.2.9.1 currently running on hera (pid = 18149)
Verbosity is at least 3
hera*CLI stop now (I pressed the up arrow key here)

As you can see, my previous commands where 'show channels' and 'sip show 
channels'. When I exited the CLI and re-entered and pressed ctrl-c, the 
commands in the history where not 'show channels and 'sip show channels' but 
'stop now' instead.

Doug.


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Re: [asterisk-users] CLI History

[Mailing List]

- Original Message - 


On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote:
 What's wrong with the Asterisk CLI history? When I exit the 
CLI, and re-enter, the last command in the history always 
defaults to 'stop now'. This is very bad, and it's caused 
accidental shutdowns more than once.


Nothing wrong here. 


Can you possibly be a little more specific on why it isn't a problem?


It's most likely how he is quitting the client.
If you exit properly (exit or quit) it retains it but if you can cancel out 
(ctrl-c) it just drops.
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[asterisk-users] Asterisk 1.4 realtime with mysql 5.0 and unixODBC.

[Maps]

Dear Friends and Supporters!

I am trying to upgrade my testing asterisk realtime 1.2.13 with MySQL 5.0 and 
unixODBC to the beta asterisk 1.4.
I run the make and make install for the asterisk-addon just fine, It created 
the modules res_config_mysql.so and  cdr_addon_mysql.so without any problem or 
error.  However, when I run the asterisk, it comes up with the error :

  == Parsing '/etc/asterisk/res_mysql.conf': Found
asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: 
undefined symbol: mysql_init

and the asterisk would not be run.

However, if I do the noload those modules

noload = res_config_mysql.so
noload = cdr_addon_mysql.so

Then the asterisk running just fine, but there is no database connection for 
asterisk realtime.

Would anyone help me, I would very appreciated.

Thanks in advance!

Lan.
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Re: [asterisk-users] re: L option in dial command

[Anthony LaMantia]

this problem is being actively worked on right now in mantis(bugs.digium.com), 
your best bet is to monitor the issue while it's being worked on. and test the 
any patches as they are uploaded

-anthony

- Original Message -
From: Yair Hakak [EMAIL PROTECTED]
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Monday, December 11, 2006 2:19:26 PM GMT-0600 US/Central
Subject: [asterisk-users] re: L option in dial command


Hello all, 
I'm having a bit for a problem with the dial command limit option. I have the 
following dial command (executed from inside the a2billing agi) 

AGI Script Executing Application: (Dial) Options: ( IAX2/[EMAIL 
PROTECTED]/18005551212|30|HL(6:2:0)0 ) 

Now, from what i read in the wiki, this is supposed to limit me to one minute 
(6 ms), and warn me when there are 20 seconds left. 

Instead, it hangs up after 40 seconds. 

i understand there was an open bug about this...is it fixed, is there a patch, 
what can i do about this? 

for reference version is: 
Asterisk 1.2.13 built by root @ phone2 on a i686 running Linux on 2006-10-28 
10:51:43 UTC 

any help is appreciated 

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Re: [asterisk-users] CLI History

[Todd- Asterisk]

short version:  me too

long version:  The same thing happens on my asterisk boxes - both  
built with the latest trixbox image...  perhaps that's a factor?  My  
history is always restart now, although I typically connect and run  
sip show peers.  I haven't typed restart now in a long time, but  
that is the first thing when I hit up-arrrow upon connecting


I have had history written to when I type 'exit' at the console  
instead of ctrl-c.   I haven't tested though as the school bus just  
arrived  ;)

   Todd


On Dec 11, 2006, at 3:35 PM, Douglas Garstang wrote:


-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Monday, December 11, 2006 12:57 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CLI History


On Mon, 2006-12-11 at 11:17 -0800, Carla Schroder wrote:

On Monday 11 December 2006 9:31 am, Douglas Garstang wrote:

What's wrong with the Asterisk CLI history? When I exit

the CLI, and

re-enter, the last command in the history always defaults

to 'stop now'.

This is very bad, and it's caused accidental shutdowns

more than once.


Connected to Asterisk 1.2.9.1 currently running on hera

(pid = 17399)

Verbosity is at least 3
hera*CLI A
No such command 'A' (type 'help' for help)
hera*CLI B
No such command 'B' (type 'help' for help)
hera*CLI C
No such command 'C' (type 'help' for help)
hera*CLI D
No such command 'D' (type 'help' for help)
hera*CLI E
No such command 'E' (type 'help' for help)
hera*CLI
[10:[EMAIL PROTECTED](pbx3):~]# asterisk -trv
Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc.

and others.

Created by Mark Spencer [EMAIL PROTECTED]
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show

warranty' for

details. This is free software, with components licensed

under the GNU

General Public License version 2 and other licenses; you

are welcome to

redistribute it under certain conditions. Type 'show

license' for details.



==
===

Connected to Asterisk 1.2.9.1 currently running on hera

(pid = 17399)

Verbosity is at least 3
hera*CLI stop now -- I pressed the UP arrow upon

re-entering the console!




 I'm a bit confused by your example. What are A,B,C, etc?

To exit the Asterisk

console, I type 'exit'. Asterisk continues to run, as it

should. To re-enter

the console I use asterisk -rvvv.


He was demonstrating how the CLI history shows stop now as the last
command (which um... it's a history?  you're last command is gonna be
the um... last command you ran... i.e. stop now).


For crying out loud, why is this so hard to understand? It isn't  
rocket science. I said that when I exit the CLI and re-enter, no  
matter what my previous set of commands was, when I hit the UP  
arrow key, it was always 'stop now'. 'Stop now' WAS NOT MY PREVIOUS  
COMMAND.


For the person that suggested maybe unknown commands are not added  
to the history...


hera*CLI show channels
Channel  Location State   Application(Data)
0 active channels
0 active calls
hera*CLI sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form   
Hold Last Message

0 active SIP channels
hera*CLI
(I Pressed Ctrl-c here)

[13:[EMAIL PROTECTED](pbx3):~]# asterisk -trv
Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer [EMAIL PROTECTED]
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty'  
for details.
This is free software, with components licensed under the GNU  
General Public
License version 2 and other licenses; you are welcome to  
redistribute it under

certain conditions. Type 'show license' for details.
== 
===

Connected to Asterisk 1.2.9.1 currently running on hera (pid = 18149)
Verbosity is at least 3
hera*CLI stop now (I pressed the up arrow key here)

As you can see, my previous commands where 'show channels' and 'sip  
show channels'. When I exited the CLI and re-entered and pressed  
ctrl-c, the commands in the history where not 'show channels and  
'sip show channels' but 'stop now' instead.


Doug.


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RE: [asterisk-users] CLI History

[Douglas Garstang]

 -Original Message-
 From: Mailing List [mailto:[EMAIL PROTECTED]
 Sent: Monday, December 11, 2006 1:38 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] CLI History
 
 
 
 - Original Message - 
  
  On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote:
   What's wrong with the Asterisk CLI history? When I exit the 
  CLI, and re-enter, the last command in the history always 
  defaults to 'stop now'. This is very bad, and it's caused 
  accidental shutdowns more than once.
  
  Nothing wrong here. 
 
 Can you possibly be a little more specific on why it isn't a problem?
 
 
 It's most likely how he is quitting the client.
 If you exit properly (exit or quit) it retains it but if you 
 can cancel out (ctrl-c) it just drops.

Yes, I think that's it. It seems that hitting ctrl-c breaks the history.
I'd file a bug (the cli process should be able to catch SIGINT), but I'm not 
running the latest 1.2.

Doug.
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RE: [Asterisk-Users] zaptel and zapata configuration

[Joe Tahan]

hello there, 
 
 I wonder if you were able to over come your problem in configuring your aculab 
card?
 
Ammar Ali



 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Thu, 20 
 Apr 2006 16:44:38 +0100 Subject: [Asterisk-Users] zaptel and zapata 
 configuration  Hi  I am trying to use asterisk with an Aculab card using 
 ss7 protocol. i have a  problem when configuring zaptel and zapata files. 
 could you give me the  right configuration of this files to get asterisk 
 functionning with ss7  protocol? I hope that you could help me!  thanks 
 and best regards  
 _ MSN 
 Hotmail sur i-mode™ : envoyez et recevez des e-mails depuis votre  téléphone 
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[asterisk-users] Extending Avaya IP Office ISDN30e with Asterisk

[Gavin Henry]

Hi All,

Has anyone hooked up * as an extension/trunk of an Avaya system that has
around 2 ISDN30e's.

Trying to add 100 extensions to one of our systems, but not sure where to
start reading.

Thanks.

-- 
Kind Regards,

Gavin Henry.

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RE: [Asterisk-Users] Aculab

[Joe Tahan]

Hello Trevor,
 
 I wonder how I can find out for sure what is the H/W version for a PROSODY 
ACULAB SS7 Card? I dought that I have a ver 1.1 which have may issues with 
recently made computers.
 
 I have a case opened for my problem with aculab but  sysdiag shows that I have 
ver 1.1 and aculab says I have ver1.5! can sysdiag be inaccurate? I tried 
config summary and I had the same result!
 
wish you can read this!
 
Ammar Ali


From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: [Asterisk-Users] Howto cut the 
first digitDate: Fri, 31 Mar 2006 13:41:38 +0100

Christian Reelfs wrote: 
 example:  044612345  should be after cut operation:  44612345   My try 
 in the extension.conf: 
 exten = _0[0-9].,2,Cut(mynum=EXTEN,/ ,1)  exten = 
 _0[0-9].,3,Dial(Zap/g1/${mynum},90,T)   but it didn't work, my problem is 
 the delemiter, I have no delemiter,  the default is - but how to use the 
 function cut() without an delemiter?  Just snip the first digit of a 
 phonenumber. 
Use the substring notation as in: ${mynum:1} 
which snips the first character from the string. See the docs for more info 
http://www.voip-info.org/wiki/view/Asterisk+variables 
Trevor Raynsford Software Engineer Aculab 
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[asterisk-users] Re: CLI History

[Steven]

Don't hit Ctrl-C!

If I type ? in the CLI, Ctrl-C is not listed as a command.
*CLI
!   abort   add ael agent   agi cdr
databasedebug   dnsmgr  dontdumpdundi   
extensions
feature group   helpiax2include indication  init
loadlocal   logger  meetme  mgcpmixmonitor  moh
no  pri realtimereload  remove  restart rtp
set showsip skinny  softstopunload
zap

The funny thing is that neither is exit.

Type exit when exiting asterisk CLI, and it will close out properly.







-- 
-- 
Steven

http://www.glimasoutheast.org



Todd- Asterisk [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
 short version:  me too

 long version:  The same thing happens on my asterisk boxes - both  built with 
 the latest trixbox image...  perhaps that's a 
 factor?  My  history is always restart now, although I typically connect 
 and run  sip show peers.  I haven't typed restart 
 now in a long time, but  that is the first thing when I hit up-arrrow upon 
 connecting

 I have had history written to when I type 'exit' at the console  instead of 
 ctrl-c.   I haven't tested though as the school bus 
 just  arrived  ;)
Todd


 On Dec 11, 2006, at 3:35 PM, Douglas Garstang wrote:

 -Original Message-
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]
 Sent: Monday, December 11, 2006 12:57 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] CLI History


 On Mon, 2006-12-11 at 11:17 -0800, Carla Schroder wrote:
 On Monday 11 December 2006 9:31 am, Douglas Garstang wrote:
 What's wrong with the Asterisk CLI history? When I exit
 the CLI, and
 re-enter, the last command in the history always defaults
 to 'stop now'.
 This is very bad, and it's caused accidental shutdowns
 more than once.

 Connected to Asterisk 1.2.9.1 currently running on hera
 (pid = 17399)
 Verbosity is at least 3
 hera*CLI A
 No such command 'A' (type 'help' for help)
 hera*CLI B
 No such command 'B' (type 'help' for help)
 hera*CLI C
 No such command 'C' (type 'help' for help)
 hera*CLI D
 No such command 'D' (type 'help' for help)
 hera*CLI E
 No such command 'E' (type 'help' for help)
 hera*CLI
 [10:[EMAIL PROTECTED](pbx3):~]# asterisk -trv
 Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc.
 and others.
 Created by Mark Spencer [EMAIL PROTECTED]
 Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show
 warranty' for
 details. This is free software, with components licensed
 under the GNU
 General Public License version 2 and other licenses; you
 are welcome to
 redistribute it under certain conditions. Type 'show
 license' for details.

 ==
 ===
 Connected to Asterisk 1.2.9.1 currently running on hera
 (pid = 17399)
 Verbosity is at least 3
 hera*CLI stop now -- I pressed the UP arrow upon
 re-entering the console!


  I'm a bit confused by your example. What are A,B,C, etc?
 To exit the Asterisk
 console, I type 'exit'. Asterisk continues to run, as it
 should. To re-enter
 the console I use asterisk -rvvv.

 He was demonstrating how the CLI history shows stop now as the last
 command (which um... it's a history?  you're last command is gonna be
 the um... last command you ran... i.e. stop now).

 For crying out loud, why is this so hard to understand? It isn't  rocket 
 science. I said that when I exit the CLI and re-enter, 
 no  matter what my previous set of commands was, when I hit the UP  arrow 
 key, it was always 'stop now'. 'Stop now' WAS NOT MY 
 PREVIOUS  COMMAND.

 For the person that suggested maybe unknown commands are not added  to the 
 history...

 hera*CLI show channels
 Channel  Location State   Application(Data)
 0 active channels
 0 active calls
 hera*CLI sip show channels
 Peer User/ANRCall ID  Seq (Tx/Rx)  Form   Hold Last 
 Message
 0 active SIP channels
 hera*CLI
 (I Pressed Ctrl-c here)

 [13:[EMAIL PROTECTED](pbx3):~]# asterisk -trv
 Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
 Created by Mark Spencer [EMAIL PROTECTED]
 Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty'  for 
 details.
 This is free software, with components licensed under the GNU  General Public
 License version 2 and other licenses; you are welcome to  redistribute it 
 under
 certain conditions. Type 'show license' for details.
 == ===
 Connected to Asterisk 1.2.9.1 currently running on hera (pid = 18149)
 Verbosity is at least 3
 hera*CLI stop now (I pressed the up arrow key here)

 As you can see, my previous commands where 'show channels' and 'sip  show 
 channels'. When I exited the CLI and re-entered and 
 pressed  ctrl-c, the commands in the history where not 'show channels and  
 'sip show 

[asterisk-users] Unable to open pseudo channel for timing... Sound may be choppy.

[Phil Finkler]

Any idea what causes the warning Unable to open pseudo channel for
timing...  Sound may be choppy.?  Any ideas what I need to resolve
this?  I do have the zaptel module installed but don't have a zaptel
card.  I'm guessing this has to do with ztdummy?  I'm running Debian and
installed asterisk, zaptel, and zaptel-source from the backports.  Any
information appreciated!

 

Cheer,

Phil

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Re: [asterisk-users] Re: Xen, Asterisk ISDN: Timing Problems

[jason]

Im passing a PVR-500, a PVR-250, a dual Intel Pro100 NIC (2 interfaces) 
one of the onboard IDE controllers, all of my USB ports and my FXO card 
without any hiccup. I stay pretty bleeding edge, so I can't say if this 
would work out of the box.  I did have to tweak a few PCI latency timers 
but nothing major.


Arik Raffael Funke wrote:
That has been fixed in the current Xen, and as far as I can tell works 
without problems. (At least for some NICs I had dedicated to another 
domU.)


Regards,
Arik


Howard Lowndes wrote:
I have to run Asterisk on the dom0 host as earlier versions of Xen 
had problems handing PCI control over to a domU kernel.  Does anyone 
know if this has been fixed yet?



Arik Raffael Funke wrote:
Thanks. What kernels do you use for dom0 and the domU's? 
Custom-built or out of the box?


- Arik


jason wrote:
I would vote RAM. I've been using a FXO card in xen for a good year 
now with no issues at all. In fact, my zttest timings are the same 
between xen and native.

Arik Raffael Funke wrote:

Hi,

is anybody running asterisk on a xen domU and can give an opinion 
on the following:


I have delegated a FritzCard and a HFC card to my domU and 
installed an asterisk setup that was running on the same isdn 
hardware but on a dedicated machine flawlessly.


I experienced what I believed to be timing problems: sometimes 
calls on the Fritzcard did not seem to reach asterisk, when 
calls were being made, sometimes they were horribly distorted. I 
quickly abandoned the project at the time for lack of time. I 
would now make another trial.


Can anybody tell me if the problems I was having were more likely 
to result from the fact that the isdn hardware was dedicated to 
the domU (i.e. maybe that produces some sort of bottleneck!?) or 
from too little ram allocated to my domU? (I believe I had 128 MB 
or so)


Thanks,
Arik

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--
Jason
The place where you made your stand never mattered,
only that you were there... and still on your feet


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RE: [asterisk-users] Re: CLI History

[Douglas Garstang]

But ctrl-c is 3 less keystrokes than exit\n !

 -Original Message-
 From: Steven [mailto:[EMAIL PROTECTED]
 Sent: Monday, December 11, 2006 2:16 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Re: CLI History
 
 
 Don't hit Ctrl-C!
 
 If I type ? in the CLI, Ctrl-C is not listed as a command.
 *CLI
 !   abort   add ael agent   
 agi cdr
 databasedebug   dnsmgr  dontdump
 dundi   extensions
 feature group   helpiax2include 
 indication  init
 loadlocal   logger  meetme  mgcp
 mixmonitor  moh
 no  pri realtimereload  remove  
 restart rtp
 set showsip skinny  soft
 stopunload
 zap
 
 The funny thing is that neither is exit.
 
 Type exit when exiting asterisk CLI, and it will close out properly.
 
 
 
 
 
 
 
 -- 
 -- 
 Steven
 
 http://www.glimasoutheast.org
 
 
 
 Todd- Asterisk [EMAIL PROTECTED] wrote in 
 message news:[EMAIL PROTECTED]
  short version:  me too
 
  long version:  The same thing happens on my asterisk boxes 
 - both  built with the latest trixbox image...  perhaps that's a 
  factor?  My  history is always restart now, although I 
 typically connect and run  sip show peers.  I haven't typed 
 restart 
  now in a long time, but  that is the first thing when I 
 hit up-arrrow upon connecting
 
  I have had history written to when I type 'exit' at the 
 console  instead of ctrl-c.   I haven't tested though as the 
 school bus 
  just  arrived  ;)
 Todd
 
 
  On Dec 11, 2006, at 3:35 PM, Douglas Garstang wrote:
 
  -Original Message-
  From: Aaron Daniel [mailto:[EMAIL PROTECTED]
  Sent: Monday, December 11, 2006 12:57 PM
  To: asterisk-users@lists.digium.com
  Subject: Re: [asterisk-users] CLI History
 
 
  On Mon, 2006-12-11 at 11:17 -0800, Carla Schroder wrote:
  On Monday 11 December 2006 9:31 am, Douglas Garstang wrote:
  What's wrong with the Asterisk CLI history? When I exit
  the CLI, and
  re-enter, the last command in the history always defaults
  to 'stop now'.
  This is very bad, and it's caused accidental shutdowns
  more than once.
 
  Connected to Asterisk 1.2.9.1 currently running on hera
  (pid = 17399)
  Verbosity is at least 3
  hera*CLI A
  No such command 'A' (type 'help' for help)
  hera*CLI B
  No such command 'B' (type 'help' for help)
  hera*CLI C
  No such command 'C' (type 'help' for help)
  hera*CLI D
  No such command 'D' (type 'help' for help)
  hera*CLI E
  No such command 'E' (type 'help' for help)
  hera*CLI
  [10:[EMAIL PROTECTED](pbx3):~]# asterisk -trv
  Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc.
  and others.
  Created by Mark Spencer [EMAIL PROTECTED]
  Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show
  warranty' for
  details. This is free software, with components licensed
  under the GNU
  General Public License version 2 and other licenses; you
  are welcome to
  redistribute it under certain conditions. Type 'show
  license' for details.
 
  ==
  ===
  Connected to Asterisk 1.2.9.1 currently running on hera
  (pid = 17399)
  Verbosity is at least 3
  hera*CLI stop now -- I pressed the UP arrow upon
  re-entering the console!
 
 
   I'm a bit confused by your example. What are A,B,C, etc?
  To exit the Asterisk
  console, I type 'exit'. Asterisk continues to run, as it
  should. To re-enter
  the console I use asterisk -rvvv.
 
  He was demonstrating how the CLI history shows stop now 
 as the last
  command (which um... it's a history?  you're last command 
 is gonna be
  the um... last command you ran... i.e. stop now).
 
  For crying out loud, why is this so hard to understand? It 
 isn't  rocket science. I said that when I exit the CLI and re-enter, 
  no  matter what my previous set of commands was, when I 
 hit the UP  arrow key, it was always 'stop now'. 'Stop now' 
 WAS NOT MY 
  PREVIOUS  COMMAND.
 
  For the person that suggested maybe unknown commands are 
 not added  to the history...
 
  hera*CLI show channels
  Channel  Location State   Application(Data)
  0 active channels
  0 active calls
  hera*CLI sip show channels
  Peer User/ANRCall ID  Seq (Tx/Rx)  
 Form   Hold Last Message
  0 active SIP channels
  hera*CLI
  (I Pressed Ctrl-c here)
 
  [13:[EMAIL PROTECTED](pbx3):~]# asterisk -trv
  Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. 
 and others.
  Created by Mark Spencer [EMAIL PROTECTED]
  Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show 
 warranty'  for details.
  This is free software, with components licensed under the 
 GNU  General Public
  License version 2 and other licenses; you are welcome to  
 redistribute it under
  certain conditions. Type 'show license' for details.
  
 

Re: [asterisk-users] TDM2400

[O . Kamal]

here is the latest update:
in zaptel.conf i used
fxsks=1-4
fxsks=5-8
fxsks=9-12
fxsks=13-16
zttool shows hardware OK
ztcfg worked normally
in zapata.conf when i define the channels channel=1-16 and restaring
asterisk it gives the below errors:
Dec 12 00:48:28 WARNING[3141]: chan_zap.c:921 zt_open: Unable to specify
channel 1: No such device
Dec 12 00:48:28 ERROR[3141]: chan_zap.c:6879 mkintf: Unable to open channel
1: No such device
here = 0, tmp-channel = 1, channel = 1
Dec 12 00:48:28 ERROR[3141]: chan_zap.c:10307 setup_zap: Unable to register
channel '1-16'
Dec 12 00:48:28 WARNING[3141]: loader.c:414 __load_resource: chan_zap.so:
load_module failed, returning -1
Dec 12 00:48:28 WARNING[3141]: loader.c:554 load_modules: Loading module
chan_zap.so failed!

When I remove the channel=1-16, it loads normally. zapata.conf is below:

[channels]
context=default
signalling=fxs_ls
;channel=1-16
usecallerid=yes
hidecallerid=no
callwaiting=yes
restrictcid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
;accountcode=lss0101
answeronpolarityswitch=yes
hanguponpolarityswitch=yes

any clue?

On 12/11/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:


On Mon, Dec 11, 2006 at 09:40:18AM +1100, Howard Lowndes wrote:


 O.Kamal wrote:
 I have 16 channels FXO (4 FXO Modules), I did follow the below link,
but
 maybe I understand it wrong (what is a module and slot?), I need an
 example.
 http://kb.digium.com/entry/1/90/ http://kb.digium.com/entry/1/90/
 
 For each FXO module, you should have a coresponding line that reads:
 fxs followed by the type of signalling (gs, ls, or ks) and the equals
 sign (=) followed by the position of the module times 4 minus 3 a dash,
 and then the number of the slot times 4.  For example, if you had a FXO
 module on slot 2 of the board using loopstart signalling, the line
would
 read: fxols=5-8, or if the module was on slot 5, the line would read:
 fxols=17-20

 OK, try either:

 fxsks=1-16

 or:

 fxsks=1-4
 fxsks=5-8
 fxsks=9-12
 fxsks=13-16


 probably the latter will be correct

Both are. Alternatively, genzaptelconf (xpp/utils/genzaptelconf) will
generate a working (though a bit verbose) configuration.

--
   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] TDM2400

[O . Kamal]

I figured out the problem, it is the location of FXO boards on cards,
channels are from 9-24 not 1-16.

Thanks all for your help, specially Tzafrir, genzaptelconf shows it clearly.

On 12/11/06, O. Kamal [EMAIL PROTECTED] wrote:


here is the latest update:
in zaptel.conf i used
fxsks=1-4
fxsks=5-8
fxsks=9-12
fxsks=13-16
zttool shows hardware OK
ztcfg worked normally
in zapata.conf when i define the channels channel=1-16 and restaring
asterisk it gives the below errors:
Dec 12 00:48:28 WARNING[3141]: chan_zap.c:921 zt_open: Unable to specify
channel 1: No such device
Dec 12 00:48:28 ERROR[3141]: chan_zap.c:6879 mkintf: Unable to open
channel 1: No such device
here = 0, tmp-channel = 1, channel = 1
Dec 12 00:48:28 ERROR[3141]: chan_zap.c:10307 setup_zap: Unable to
register channel '1-16'
Dec 12 00:48:28 WARNING[3141]: loader.c:414 __load_resource: chan_zap.so:
load_module failed, returning -1
Dec 12 00:48:28 WARNING[3141]: loader.c:554 load_modules: Loading module
chan_zap.so failed!

When I remove the channel=1-16, it loads normally. zapata.conf is
below:

[channels]
context=default
signalling=fxs_ls
;channel=1-16
usecallerid=yes
hidecallerid=no
callwaiting=yes
restrictcid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
;accountcode=lss0101
answeronpolarityswitch=yes
hanguponpolarityswitch=yes

any clue?

On 12/11/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:

 On Mon, Dec 11, 2006 at 09:40:18AM +1100, Howard Lowndes wrote:
 
 
  O.Kamal wrote:
  I have 16 channels FXO (4 FXO Modules), I did follow the below link,
 but
  maybe I understand it wrong (what is a module and slot?), I need an
  example.
  http://kb.digium.com/entry/1/90/ http://kb.digium.com/entry/1/90/
  
  For each FXO module, you should have a coresponding line that reads:

  fxs followed by the type of signalling (gs, ls, or ks) and the
 equals
  sign (=) followed by the position of the module times 4 minus 3 a
 dash,
  and then the number of the slot times 4.  For example, if you had a
 FXO
  module on slot 2 of the board using loopstart signalling, the line
 would
  read: fxols=5-8, or if the module was on slot 5, the line would
 read:
  fxols=17-20
 
  OK, try either:
 
  fxsks=1-16
 
  or:
 
  fxsks=1-4
  fxsks=5-8
  fxsks=9-12
  fxsks=13-16
 
 
  probably the latter will be correct

 Both are. Alternatively, genzaptelconf (xpp/utils/genzaptelconf) will
 generate a working (though a bit verbose) configuration.

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

[Benny Amorsen]

 DG == Douglas Garstang [EMAIL PROTECTED] writes:

DG When I exited the CLI and re-entered and pressed ctrl-c,

That's where your problem is. Use exit and not ctrl-c to leave
asterisk -r.


/Benny


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[asterisk-users] Using SIP with NAT (technical code question)

[jezzzz .]

In chan_sip.c, line 5876 (Asterisk-1.2.13), the
function parse_ok_contact returns whether the host
that requested an invite is a valid or invalid host.

In line 5925 the following clause is tested:

if (!(ast_test_flag(pvt, SIP_NAT)  SIP_NAT_ROUTE))
hp = ast_gethostbyname(n, ahp);

If this clause is true then Asterisk will attempt to
retrieve the IP address by using the hostname provided
in the invite.

My question is, is this test always going to be true
if a user (who receives the invite) uses NAT? (this is
set up in sip.conf as nat=yes) Is there a reason why
this was set up only for NAT? 

Thanks,

Jez

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[asterisk-users] VPN As SIP Tunneling?

[Barry Fawthrop]

Hi All

Could a VPN be used to help with SIP Tunneling and QoS issues.

State 1:
Two IP Networks Connected via the Public Internet transmitting VoIP Traffic
Say a VoIP User and VoIP Termination Provider.
Each side can put QoS onto their part, but if QoS does NOT exist between 
them

then call quality will be bad anyhow.

State 2:
Same as above except a VPN tunnel is setup between each side.
Thus making them appear on the same network and possibly same subnet.

(1) Would this now traceroute a one hop ?
(2) Would this have a lower or higher ping time, thus latency ?
(3) With the additional Encryption etc.. if using a 1 Mbps Internet 
connection

   What would the actual amount available now be 700kbps, in other words
   How much overhead is there with a VPN tunnel that would reduce the 
available bandwidth ?



Thanks to all

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Re: [asterisk-users] Recommendations for QoS, PoE Switches

[Barry Fawthrop]

Hi David
Care to share how you approached using Diffserv and VLANs with the FSM7326P
We are considering the same switch. But I'm unsure about the configurations
required.

Thanks in advance
Barry


David Coulson wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Typically we deploy the FSM7326P from Netgear. 24 10/100 ports w/ PoE 
2 GigE ports (copper or SFP GBIC). We've not had any problems filling
all the ports up with SPIP501s. It has L3 switching/routing features, so
it's not as cheap as some other basic PoE switches that exist - I
believe Netgear have a L2 only 24 port PoE switch on the way in Q1 of
'07. It has all of the goodies like SNMP, Diffserv/CoS, VLANs, dot1q and
so forth, so it's a pretty nice switch for most installations.

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Re: [asterisk-users] Using SIP with NAT (technical code question)

[Bob Chiodini]

It looks to me that if the test clause is false then 
ast_gethostbyname is called.  Presumably not needed when NAT is enabled.


Bob...

je . wrote:

In chan_sip.c, line 5876 (Asterisk-1.2.13), the
function parse_ok_contact returns whether the host
that requested an invite is a valid or invalid host.

In line 5925 the following clause is tested:

if (!(ast_test_flag(pvt, SIP_NAT)  SIP_NAT_ROUTE))
hp = ast_gethostbyname(n, ahp);

If this clause is true then Asterisk will attempt to
retrieve the IP address by using the hostname provided
in the invite.

My question is, is this test always going to be true
if a user (who receives the invite) uses NAT? (this is
set up in sip.conf as nat=yes) Is there a reason why
this was set up only for NAT? 


Thanks,

Jez

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[asterisk-users] Asterisk Sends 180-RINGING to UA even with progressinband=yes

[Douglas Garstang]

I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my 
polycom phones and then it also sends 183-Session Progress. That doesn't seem 
to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ?

Doug.
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Re: [asterisk-users] VPN As SIP Tunneling?

[Anselm Martin Hoffmeister]

Hi Barry,

I used SIP over OpenVPN when travelling, especially from hotel rooms or
showfloors. Of course I did not expect the performance of a local SIP
connection, but generally it worked OK. The latency would not suffer
much in comparison to direct connection, but a WLAN was involved which
would screw quality anyway. Using a bluetooth headset would not help
much, either. Having my geographic number from Europe ringing on my
twinkle softphone over in California was nice-to-have

Everything I say will be relevant to OpenVPN, which might be a bit
different from IPsec, PPTP or other solutions.

Am Montag, den 11.12.2006, 17:26 -0500 schrieb Barry Fawthrop:
 Hi All
 
 Could a VPN be used to help with SIP Tunneling and QoS issues.
 
 State 1:
 Two IP Networks Connected via the Public Internet transmitting VoIP Traffic
 Say a VoIP User and VoIP Termination Provider.
 Each side can put QoS onto their part, but if QoS does NOT exist between 
 them
 then call quality will be bad anyhow.
 
 State 2:
 Same as above except a VPN tunnel is setup between each side.
 Thus making them appear on the same network and possibly same subnet.
 
 (1) Would this now traceroute a one hop ?

Yep, but the packet containing the ICMP (ping) packet to traceroute the
connection will itself be bumped around, so it will be the same number
of hops really, just those between the VPN endpoints will be hidden.

You win the bonus of getting around complicated NAT, possibly.

 (2) Would this have a lower or higher ping time, thus latency ?

Higher, it can impossibly be faster than the packets carrying the VPN. I
think you will not notice the difference though, because OpenVPN seems
to do a good job.

 (3) With the additional Encryption etc.. if using a 1 Mbps Internet 
 connection
 What would the actual amount available now be 700kbps, in other words
 How much overhead is there with a VPN tunnel that would reduce the 
 available bandwidth ?

Sorry, no numbers from me. For connection oriented protocols like HTTP,
FTP, Mail, the additional problem of encapsulating TCP in TCP will kill
the TCP windowing, as such not allowing for full line saturation (do not
ask me for details, I slept to much during the networking lecture last
year to tell you without a look into transcripts). No problem for UDP
(like SIP/RTP). General data/overhead ratio will be probably better with
larger data packets - I seem to remember you can configure the
packetation size for some audio codecs in Asterisk. Alas I did not care,
telephony was good enough.

I would expect some throughput value wildly between 70 and 95%, but that
is a guess, and not even an educated one. It probably depends on the VPN
technology you use, and the maker(s) will be your authoritative data
source there.

Let us not forget that of course the data stream can be encrypted. Just
that you think you are talking boring stuff does not mean there would be
noone interested in wiretapping and listening in.
(Even if I'm not paranoid they may be after me... ;)

Hth
Anselm

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Re: [asterisk-users] Asterisk Sends 180-RINGING to UA even with progressinband=yes

[Andrew Joakimsen]

When we send 183, that means 'inband progress' is available. That does _not_
necessarily mean that it is ringing, it could be any sort of progress tone,
or even audio from an IVR. If your ATA does not stop its own ringing
generator and start forwarding the audio, it is broken.

It is my understanding that Polycom's SIP implemenation does not currectly
handle these responses. See: http://bugs.digium.com/view.php?id=3129

In the future it would help that instead of nitpicking some little low level
technical detail you describe what your actual problem is, you would get
more input that way. progessinband=yes means that the call progress WILL BE
SEND INBAND, which in 99% of cases is not needed, and does not make sense.
You are also wasting additinal resources because asterisk must generate
progress tones too.

On 12/11/06, Douglas Garstang [EMAIL PROTECTED] wrote:


I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to
my polycom phones and then it also sends 183-Session Progress. That doesn't
seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if
progressinband=yes ?

Doug.
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Re: [asterisk-users] VPN As SIP Tunneling?

[Andrew Joakimsen]

So in your example you can manage QoS within the VPN but have no control
whatsoever over the VPN tunnel as a hole, it would be the same result as if
you just passed straigth TCP over your connection with QoS, however  you
will waste more resourses for the VPN and probably introduce a bit of
latency, small but latency none-the-less.

On 12/11/06, Barry Fawthrop [EMAIL PROTECTED] wrote:


Hi All

Could a VPN be used to help with SIP Tunneling and QoS issues.

State 1:
Two IP Networks Connected via the Public Internet transmitting VoIP
Traffic
Say a VoIP User and VoIP Termination Provider.
Each side can put QoS onto their part, but if QoS does NOT exist between
them
then call quality will be bad anyhow.

State 2:
Same as above except a VPN tunnel is setup between each side.
Thus making them appear on the same network and possibly same subnet.

(1) Would this now traceroute a one hop ?
(2) Would this have a lower or higher ping time, thus latency ?
(3) With the additional Encryption etc.. if using a 1 Mbps Internet
connection
What would the actual amount available now be 700kbps, in other
words
How much overhead is there with a VPN tunnel that would reduce the
available bandwidth ?


Thanks to all

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Re: [asterisk-users] NAT and Dial to two channels at once

[Andrew Joakimsen]

You need to understand how NAT works, if you can chan2 and chan2 is behind a
NAT and suddenly someone else is invited to chan2's IP address port 5060
chan2's router willl say WTF I dont have an estabished connection on port
5060 (to the client being reinvited to chan2) and it wont work. You need to
have the media path go through asterisk in that case.

On 12/10/06, Brad Templeton [EMAIL PROTECTED] wrote:


On Sun, Dec 10, 2006 at 01:39:10PM -0500, Eric Jacksch wrote:
 Someone may have a more elegant solution, but I have found that allowing
 reinvite on a phone connected via NAT gateway causes too many problems,
 especially with the difference in the various NAT implementations.  I
 set canreinvite=no host=dynamic nat=yes qualify=yes for all phones that
 connect from the Internet.

 A note to anyone doing Internet telephony -- the latest Linksys routers
 seem to ignore the small UDP keepalive that some phones like the
 grandstreams send, so the NAT hole closes.  I've had to start using
 qualify=yes to get asterisk to keep the NAT hole open.


Actually, that part is working, and in any event I can hand-open holes
in the NAT in cases where it doesn't work.

As long as the incoming call is redirected to a single channel, it
works fine.   Setting canreinvite=no and nat=yes and the rest don't
help the problem we get when the incoming call is sent to a
Dial(chan1chan2).

I don't want to turn reinvite off, it's a very important feature.
I'm giving IP phones to all members of the family this christmas,
and most of them live 2500 miles away.   Nobody will want to have
their audio hairpin through my server.   That would just add lots of
latency -- indeed enough to make the calls much less pleasant -- as
well as eat my own bandwidth and add risk of packet loss for no
reason.

As noted, it all works with just one channel, but for many of them,
to get them introduced to voip, it would be nice if I could have their
DID ring both their new SIP phone and their old PSTN phone (and
possibly their cell phone at the same time, though the voice-mail
problem remains a curse if you do that until I can convince Mark to
put in my fix for that.)


Ringing both the pstn phone and the IP phone is not just for
newcomers, however.  If you only have one IP phone in the house,
and you are not near it, you may not hear it and you don't want to
have to run elsewhere to get it.  (Could do call pickup over
PSTN.)

I understand if you Dial(chan1chan2) you need to do independent
invites with their own SDPs, because the two channels can be in
different places, with different codecs.  You don't know until
one answers what you will finally do, though if the one that answers
can be native bridged, it should be native bridged.

However, the current situation -- no audio at all until a reinvite
is triggered by putting the call on hold -- is obviously not very
exciting.
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Re: [asterisk-users] VPN As SIP Tunneling?

[Barry Fawthrop]

Hi Anselm
Thanks for your input
Yes I was thinking of using OpenVPN so it was good to hear your experiences
I'm not so much concerned with the encryption of traffic etc..
But the Level of QoS.
If my IP Phone set QoS and the VoIP Termination provider's * PBX sets QoS

And we now connected via a VPN tunnel. We should be able to guarantee 
Quality due to the Tunnel.

The main issue is would I expect a higher latency ?
and (2) If I were using a 1 Mbps connect would I have less bandwidth due 
to overheads. That where I could do
8 concurrent calls x 115 bps 920 kbps  I could now only do 6  or will I 
still be able to do 8 ?


Thanks as always
Barry

Anselm Martin Hoffmeister wrote:

Hi Barry,

I used SIP over OpenVPN when travelling, especially from hotel rooms or
showfloors. Of course I did not expect the performance of a local SIP
connection, but generally it worked OK. The latency would not suffer
much in comparison to direct connection, but a WLAN was involved which
would screw quality anyway. Using a bluetooth headset would not help
much, either. Having my geographic number from Europe ringing on my
twinkle softphone over in California was nice-to-have

Everything I say will be relevant to OpenVPN, which might be a bit
different from IPsec, PPTP or other solutions.

Am Montag, den 11.12.2006, 17:26 -0500 schrieb Barry Fawthrop:
  

Hi All

Could a VPN be used to help with SIP Tunneling and QoS issues.

State 1:
Two IP Networks Connected via the Public Internet transmitting VoIP Traffic
Say a VoIP User and VoIP Termination Provider.
Each side can put QoS onto their part, but if QoS does NOT exist between 
them

then call quality will be bad anyhow.

State 2:
Same as above except a VPN tunnel is setup between each side.
Thus making them appear on the same network and possibly same subnet.

(1) Would this now traceroute a one hop ?



Yep, but the packet containing the ICMP (ping) packet to traceroute the
connection will itself be bumped around, so it will be the same number
of hops really, just those between the VPN endpoints will be hidden.

You win the bonus of getting around complicated NAT, possibly.

  

(2) Would this have a lower or higher ping time, thus latency ?



Higher, it can impossibly be faster than the packets carrying the VPN. I
think you will not notice the difference though, because OpenVPN seems
to do a good job.

  
(3) With the additional Encryption etc.. if using a 1 Mbps Internet 
connection

What would the actual amount available now be 700kbps, in other words
How much overhead is there with a VPN tunnel that would reduce the 
available bandwidth ?



Sorry, no numbers from me. For connection oriented protocols like HTTP,
FTP, Mail, the additional problem of encapsulating TCP in TCP will kill
the TCP windowing, as such not allowing for full line saturation (do not
ask me for details, I slept to much during the networking lecture last
year to tell you without a look into transcripts). No problem for UDP
(like SIP/RTP). General data/overhead ratio will be probably better with
larger data packets - I seem to remember you can configure the
packetation size for some audio codecs in Asterisk. Alas I did not care,
telephony was good enough.

I would expect some throughput value wildly between 70 and 95%, but that
is a guess, and not even an educated one. It probably depends on the VPN
technology you use, and the maker(s) will be your authoritative data
source there.

Let us not forget that of course the data stream can be encrypted. Just
that you think you are talking boring stuff does not mean there would be
noone interested in wiretapping and listening in.
(Even if I'm not paranoid they may be after me... ;)

Hth
Anselm

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Re: [asterisk-users] VPN As SIP Tunneling?

[Eric \ManxPower\ Wieling]

Some VPN implementations allow you to copy the ToS of the encapsulated 
packets to the ToS of the wrapper packet.


Andrew Joakimsen wrote:
So in your example you can manage QoS within the VPN but have no control 
whatsoever over the VPN tunnel as a hole, it would be the same result as 
if you just passed straigth TCP over your connection with QoS, however  
you will waste more resourses for the VPN and probably introduce a bit 
of latency, small but latency none-the-less.

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[asterisk-users] How to add include statement into Realtime static

[Tielin Xu]

Hi List:

I can not find out an example how to store include = context name 
statement into Realtime static.
Please help me on this one.

Thanks,

Tielin
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RE: [asterisk-users] Asterisk Sends 180-RINGING to UA even withprogressinband=yes

[Douglas Garstang]

Andrew,
 
I don't think it's a Polycom issue. We took Asterisk out of the picture and had 
our Polycom phones communicate directly with an Audiocodes PSTN gateway. Unlike 
Asterisk, the audiocodes do not send 180 Ringing before sending 183 Session 
Progress, and the polycom's play the correct tones in this case.
 
We WANT Asterisk to send progress tones in band. In our case it IS needed. 
What's the SIP response for a reorder then if we don't need in band progress 
tones? There is none. In a situation where the PSTN end sends back a reorder, 
or some other unusual tone, all the UA ends up hearing is the closest SIP 
approximation, which is ringing, which is not correct.
 
I have tried to explain my issues in detail in this list in the past, and I 
have invariably met with responses like 'I don't understand' or 'why would you 
want to do that?'. I get much better understanding of my issues, and therefore 
better replies, when I break the problem down and only explain the relevant 
portions.
 
I really don't appreciate your tone.
 
Douglas.
 
 
 -Original Message-
From: Andrew Joakimsen [mailto:[EMAIL PROTECTED]
Sent: Monday, December 11, 2006 4:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Sends 180-RINGING to UA even 
withprogressinband=yes



When we send 183, that means 'inband progress' is available. That does _not_ 
necessarily mean that it is ringing, it could be any sort of progress tone, or 
even audio from an IVR. If your ATA does not stop its own ringing generator and 
start forwarding the audio, it is broken.

It is my understanding that Polycom's SIP implemenation does not currectly 
handle these responses. See: http://bugs.digium.com/view.php?id=3129

In the future it would help that instead of nitpicking some little low level 
technical detail you describe what your actual problem is, you would get more 
input that way. progessinband=yes means that the call progress WILL BE SEND 
INBAND, which in 99% of cases is not needed, and does not make sense. You are 
also wasting additinal resources because asterisk must generate progress tones 
too. 


On 12/11/06, Douglas Garstang  [EMAIL PROTECTED] wrote: 

I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my 
polycom phones and then it also sends 183-Session Progress. That doesn't seem 
to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ? 

Doug.
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Re: [asterisk-users] Asterisk Sends 180-RINGING to UA even withprogressinband=yes

[Eric \ManxPower\ Wieling]

Douglas Garstang wrote:



Andrew,

 

I don't think it's a Polycom issue. We took Asterisk out of the picture 
and had our Polycom phones communicate directly with an Audiocodes PSTN 
gateway. Unlike Asterisk, the audiocodes do not send 180 Ringing before 
sending 183 Session Progress, and the polycom's play the correct tones 
in this case.


Have you tried an Answer() before your Dial?  That should FORCE inband 
progress tones.  You'll have to have a /etc/asterisk/indications.conf of 
course.

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Re: [asterisk-users] Asterisk with IM

[Anton Raharja]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Mochamad Susantok wrote:
 Hi all,
 Howto configure asterisk 1.2.13 (debian-base) with support Instant
 Messaging, especially using client Xlite v.3.
 
 Thanks
 

Hello,

Im using my patched chan_sip.c for that.
http://www.voiprakyat.or.id/download/server/asterisk/sip-messaging/1.2.13/

anton
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (MingW32)

iD8DBQFFffXU5ByPs8h3tvwRAtvrAJ4+otMwOEdohO6acrLgdPPuBPuZRwCgv3Up
IPheq/tk8dV5eCmK7hVbJro=
=vrNg
-END PGP SIGNATURE-
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Re: [asterisk-users] Using SIP with NAT (technical code question)

[jezzzz .]

My mistake, I misread it. So if a hostname is provided
(e.g. [EMAIL PROTECTED]) instead of an IP (e.g.
123.123.123.123) and the recipient of the INVITE is
not using NAT then ast_gethostbyname will be run - is
that correct? In this case, why the distinction
between a NATted and non_NATted implementation?

--- Bob Chiodini [EMAIL PROTECTED] wrote:

 It looks to me that if the test clause is false then
 
 ast_gethostbyname is called.  Presumably not needed
 when NAT is enabled.
 
 Bob...
 
 je . wrote:
  In chan_sip.c, line 5876 (Asterisk-1.2.13), the
  function parse_ok_contact returns whether the host
  that requested an invite is a valid or invalid
 host.
 
  In line 5925 the following clause is tested:
 
  if (!(ast_test_flag(pvt, SIP_NAT) 
 SIP_NAT_ROUTE))
  hp = ast_gethostbyname(n, ahp);
 
  If this clause is true then Asterisk will attempt
 to
  retrieve the IP address by using the hostname
 provided
  in the invite.
 
  My question is, is this test always going to be
 true
  if a user (who receives the invite) uses NAT?
 (this is
  set up in sip.conf as nat=yes) Is there a reason
 why
  this was set up only for NAT? 
 
  Thanks,
 
  Jez
 
  __
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Do you Yahoo!?
Everyone is raving about the all-new Yahoo! Mail beta.
http://new.mail.yahoo.com
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Re: [asterisk-users] TDM2400

[Time Bandit]

 [channels]
 context=default
 signalling=fxs_ls
 ;channel=1-16
 usecallerid=yes
 hidecallerid=no
 callwaiting=yes
 restrictcid=no
 echocancel=yes
 echocancelwhenbridged=yes
 rxgain=0.0
 txgain=0.0
 group=1
 ;accountcode=lss0101
 answeronpolarityswitch=yes
 hanguponpolarityswitch=yes

To the best of my knowledge, all the settings you put after defining
the channles (channel= line) are useless. You have to set all the
settings BEFORE you define the channels.

hth
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RE: [asterisk-users] Asterisk Sends 180-RINGING to UAeven withprogressinband=yes

[Douglas Garstang]

 -Original Message-
 From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
 Sent: Monday, December 11, 2006 5:11 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Asterisk Sends 180-RINGING to UAeven
 withprogressinband=yes
 
 
 Douglas Garstang wrote:
  
  
  Andrew,
  
   
  
  I don't think it's a Polycom issue. We took Asterisk out of 
 the picture 
  and had our Polycom phones communicate directly with an 
 Audiocodes PSTN 
  gateway. Unlike Asterisk, the audiocodes do not send 180 
 Ringing before 
  sending 183 Session Progress, and the polycom's play the 
 correct tones 
  in this case.
 
 Have you tried an Answer() before your Dial?  That should 
 FORCE inband 
 progress tones.  You'll have to have a 
 /etc/asterisk/indications.conf of 
 course.

No... but if we answer the call before dialling, isn't that going to cause a 
whole world of billing hurt?

Doug
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Re: [asterisk-users] Asterisk Sends 180-RINGING to UAeven withprogressinband=yes

[Eric \ManxPower\ Wieling]

Douglas Garstang wrote:


No... but if we answer the call before dialling, isn't that going to cause a 
whole world of billing hurt?


You are only answering the call leg from the Polycom to Asterisk.  You 
are not answering the Asterisk - PSTN leg (I assume that is the only 
leg you bill for)

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Re: [asterisk-users] Mediatrix 1124 setup

[Tim Panton]

On 11 Dec 2006, at 04:25, cb wrote:

I recently purchased a Mediatrix 1124 from an auction of a company  
that went out of business. It came with nothing other than the unit  
itself.


In digging thru the Mediatrix web site, and various google  
searches, it looks like it only supports SNMP setup, and only with  
their software (or the correct MIB). However, Mediatrix doesn't  
appear to let you download said software or MIB from their web site.


Does anyone know where I can get the setup software or MIB needed  
to program this thing? I *think* I need the correct one for its  
firmware version, but I can't find out how to tell what version  
firmware it has. There is what appears to be the remains of a  
sticker marked Rev 4 on the bottom if that is any help.



It looks like there might be enough info on these pages to get you  
going:


http://www.sonoracomm.com/index.php? 
option=com_contenttask=viewid=68Itemid=32
Tells you that you can use netsnmp to talk to it with the public  
community name


http://www.abptech.com/mainpages/support/qa/index.php?target=mdtx- 
register

Gives you the OIDs of a number of 'useful' variables to set.

http://sipx-wiki.calivia.com/index.php/ 
HowTo_configure_Mediatrix_SIP_Gateway_with_sipX

Gives you some valid values

http://web.abptech.com/firmware/mdtx/ 
Configuration_Notes_0217_Remote_Line_Extension_(SIP).pdf
Gives you some more OIDs and the overall structure of the MIB if you  
squint at the screen grabs.


Good luck,

If you need a hand with the SNMP side, drop me a mail

Tim.

www.mexuar.net
www.westhawk.co.uk/



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Re: [asterisk-users] Recommendations for QoS, PoE Switches

[Zeeshan Zakaria]

Thanks for everybody's help.

Cory, thanks for the links. I once studied OSI model, many years ago, when I
was doing MCSE for Win NT. I'll go through these Cisco documents to
improve/update my knowledge about OSI layers and see how it can help me in
VoIP networking.
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Re: [asterisk-users] Asterisk Sends 180-RINGING to UA even withprogressinband=yes

[Andrew Joakimsen]

Reorder tone can be used for many things, is there anything I've missed?


  7.4.2  401 Unauthorized    78
  7.4.4  403 Forbidden ...   78
  7.4.5  404 Not Found ...   78
  7.4.6  405 Method Not Allowed ..   78
  7.4.7  406 Not Acceptable ..   79
  7.4.11 410 Gone    79
  7.4.16 420 Bad Extension ...   80
  7.4.17 480 Temporarily Unavailable .   80
  7.4.18 481 Call Leg/Transaction Does Not Exist .   81
  7.4.19 482 Loop Detected ...   81
  7.4.20 483 Too Many Hops ...   81
  7.4.21 484 Address Incomplete ..   81
  7.4.22 485 Ambiguous ...   81
  7.4.23 486 Busy Here ...   82
  7.5Server Failure 5xx ..   82
  7.5.1  500 Server Internal Error ...   82
  7.5.2  501 Not Implemented .   82
  7.5.3  502 Bad Gateway .   82
  7.5.4  503 Service Unavailable .   83
  7.5.5  504 Gateway Time-out    83
  7.5.6  505 Version Not Supported ...   83
  7.6Global Failures 6xx .   83
  7.6.1  600 Busy Everywhere .   83
  7.6.2  603 Decline .   84
  7.6.3  604 Does Not Exist Anywhere .   84
  7.6.4  606 Not Acceptable ..   84

All of these are defined by RFC2543. 183 is not defined until 2 years later.

Do you have any examples where ringing is indicated and it should not be? I
would really like to know, I am not trying to say you are wrong, I've must
have never encountered such a situation, if a recorded message is played
from the far switch, the audio should be passed, if it tone is played that
is legacy pstn if its over the network or the near end such as a PBX
generating the tone, anything that is digitally interconnected to a proper
ss7 network, be it an ISDN line, PRI or SIP provider, should pass proper
progress out of band. If you are using analog lines then get rid of
progressinband configurations and do as Mr. Wieling suggests.

On 12/11/06, Douglas Garstang [EMAIL PROTECTED] wrote:


 Andrew,

I don't think it's a Polycom issue. We took Asterisk out of the picture
and had our Polycom phones communicate directly with an Audiocodes PSTN
gateway. Unlike Asterisk, the audiocodes do not send 180 Ringing before
sending 183 Session Progress, and the polycom's play the correct tones in
this case.

We WANT Asterisk to send progress tones in band. In our case it IS needed.
What's the SIP response for a reorder then if we don't need in band progress
tones? There is none. In a situation where the PSTN end sends back a
reorder, or some other unusual tone, all the UA ends up hearing is the
closest SIP approximation, which is ringing, which is not correct.

I have tried to explain my issues in detail in this list in the past, and
I have invariably met with responses like 'I don't understand' or 'why would
you want to do that?'. I get much better understanding of my issues, and
therefore better replies, when I break the problem down and only explain the
relevant portions.

I really don't appreciate your tone.

Douglas.


 -Original Message-
*From:* Andrew Joakimsen [mailto:[EMAIL PROTECTED]
*Sent:* Monday, December 11, 2006 4:40 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Asterisk Sends 180-RINGING to UA even
withprogressinband=yes

When we send 183, that means 'inband progress' is available. That does
_not_ necessarily mean that it is ringing, it could be any sort of progress
tone, or even audio from an IVR. If your ATA does not stop its own ringing
generator and start forwarding the audio, it is broken.

It is my understanding that Polycom's SIP implemenation does not currectly
handle these responses. See: http://bugs.digium.com/view.php?id=3129

In the future it would help that instead of nitpicking some little low
level technical detail you describe what your actual problem is, you would
get more input that way. progessinband=yes means that the call progress WILL
BE SEND INBAND, which in 99% of cases is not needed, and does not make
sense. You are also wasting additinal resources because asterisk must
generate progress tones too.

On 12/11/06, Douglas Garstang [EMAIL PROTECTED] wrote:

 I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing
 

Re: [asterisk-users] Mediatrix 1124 setup

[cb]

On Dec 11, 2006, at 8:58 PM, Tim Panton wrote:

It looks like there might be enough info on these pages to get you  
going:


Thanks for the links! Hopefully I can get somewhere with the info.


If you need a hand with the SNMP side, drop me a mail


I'm pretty new to SNMP, so I may take you up on that once I have some  
intelligent questions to ask. I'll play around with it for a while  
and see what I can learn first.



Thanks again!

-chris
www.mythtech.net


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Re: [asterisk-users] Asterisk with IM

[Mochamad Susantok]

How do i patch file chan_sip.so ?
I use asterisk with Debian distro not asterisk-XXX.tar.gz


 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Mochamad Susantok wrote:
 Hi all,
 Howto configure asterisk 1.2.13 (debian-base) with support Instant
 Messaging, especially using client Xlite v.3.

 Thanks


 Hello,

 Im using my patched chan_sip.c for that.
 http://www.voiprakyat.or.id/download/server/asterisk/sip-messaging/1.2.13/

 anton
 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.4.5 (MingW32)

 iD8DBQFFffXU5ByPs8h3tvwRAtvrAJ4+otMwOEdohO6acrLgdPPuBPuZRwCgv3Up
 IPheq/tk8dV5eCmK7hVbJro=
 =vrNg
 -END PGP SIGNATURE-
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-
This email was sent using Student EEPIS-Webmail.
http://student.eepis-its.edu/

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Re: [asterisk-users] VPN As SIP Tunneling?

[Luki]

If my IP Phone set QoS and the VoIP Termination provider's
* PBX sets QoS. And we now connected via a VPN tunnel.
We should be able to guarantee Quality due to the Tunnel.


Nope. You only control the QOS within your tunnel (i.e. among other
traffic flowing through the tunnel). But what QOS guarantee does your
tunnel traffic have? None, if it goes through the public Internet. You
don't gain anything QOS-wise by going through a tunnel, except hiding
your traffic in case your ISP purposefully assigns lower priority to
VoIP traffic and doesn't do it to OpenVPN/GRE/insert your favorite
tunnel protocol traffic.

--Luki
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Re: [asterisk-users] Low beep on voicemail

[Anthony Rodgers]

Just 'sox -v 1.5 beep.gsm loudbeep.gsm' ?

CP

On 2-Dec-06, at 11:29 AM, Peder @ NetworkOblivion wrote:


We've had a few people complain that the beep before leaving a
voicemail is not loud enough and too short.  Does anybody have a
recorded beep that they can share, that is a little louder and a  
little

longer?  We've had this box in production for 2+ years, so I hate to
mess with the gain on the PRI or anything like that because everything
else works fine.

I know nothing about recording sounds, and I am sure I could spend  
a few
hours and come up with a suitable version, but I thought I'd ask  
around

before I waste my time trying to figure it out.

Thanks in advance.

Peder

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RE: [asterisk-users] Asterisk Sends 180-RINGINGto UAeven withprogressinband=yes

[Douglas Garstang]

Hmmm. Ok, that's true. At the very least it will create confusing CDR's I 
think... maybe. We're not billing our OnNet traffic at all. Only the traffic 
that goes OffNet, to our switch is billed (if it leaves our switch that is...).

I was thinking earlier too that we only need progressinband on traffic that 
goes to the PSTN, via our switch. OnNet traffic will never generate reorder 
tones and such. The docs say that progressinband can either go into the general 
section of sip.conf, or the extension. I couldn't get it to have any effect at 
the extension level. 

Doug.



 -Original Message-
 From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
 Sent: Monday, December 11, 2006 6:47 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Asterisk Sends 180-RINGINGto UAeven
 withprogressinband=yes
 
 
 Douglas Garstang wrote:
 
  No... but if we answer the call before dialling, isn't that 
 going to cause a whole world of billing hurt?
 
 You are only answering the call leg from the Polycom to 
 Asterisk.  You 
 are not answering the Asterisk - PSTN leg (I assume that is the only 
 leg you bill for)
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Re: [asterisk-users] Asterisk from Debian Packages

[Alex]

You can run Asterisk 1.2 in sarge using the packages in backports.

Just add:

deb http://www.backports.org/debian/ sarge-backports main contrib non-free

to /etc/apt/sources.list

then apt-get update

and then apt-get -t sarge-backports install asterisk

(you can also pin-priority asterisk's packages, look at APT documentation).

-Alex

On 12/10/06, Phil Finkler [EMAIL PROTECTED] wrote:


 Hi all,



I've gotten asterisk installed on Debian only to realize that the packaged
version is 1.0.7.  Is there a reason why they're not up to a 1.2.xrelease?  I'm 
building a system for production and I'm wondering if I should
remain at this old version or if there are any serious issues with 1.2.13on 
Debian?  Should I be able to do an apt-get from unstable and get
1.2.13 and be on my happy way?



Thanks for the help on a stupid question,

Phil



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Re: [asterisk-users] CLI History

[Benjamin Jacob]

On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote:
 


What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, 
the last command in the history always defaults to 'stop now'. This is very 
bad, and it's caused accidental shutdowns more than once.
   

thats prety smart...   think hard.. wot was the command u gave to exit 
the CLI??

history is a last-in-first-out kinda setup, anywhere, not just in * CLI.

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[asterisk-users] Problem in making outbound calls in PRI

[Danny]

Hey everyone !


I have a problem in making outbound calls in PRI connection.
I have E1 PRI airtel connection [ India ]
[ asterisk-1.2.12.1 on CentOS 4.4 ]


zaptel.conf
--
[channels]
language=en

usecallerid = yes
hidecallerid = no
callwaiting=yes
threewaycalling = yes
usecallingpres=yes
transfer = yes
echocancel = yes
echotraining = yes
immediate = no


;group=0
;context = from-pstn
;signalling = fxs_ks
;channel = 1

callwaitingcallerid=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
answeronpolarityswitch=yes
rxgain=0.0
txgain=0.0

; faxdetect=incoming


;-
immediate=no
overlapdial=yes
pridialplan=national
prilocaldialplan=national

group=0
context = from-pstn
callerid=asreceived
switchtype = euroisdn
signalling = pri_cpe
channel = 1-15,17-31
; 
--- 




; zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31


#fxsks=1

# Global data

loadzone= us
defaultzone = us

; 
--- 





When i try to make an outbound call, I get this in the error message : -



 -- Executing Dial(SIP/1001-0879af90, ZAP/g0/908239793) in new stack
 -- Requested transfer capability: 0x00 - SPEECH
 -- Called g0/9821795097
 -- Zap/1-1 is proceeding passing it to SIP/1001-0879af90
 -- Channel 0/1, span 1 got hangup request
 -- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
CLI 






I am not sure of what is wrong with my zaptel config. Any suggestions ?

- Danny




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[asterisk-users] Sip communicator issue

[Thirumal Saminathan]

Hi all,
Thanks for your reply,
I'm using sip communicator(in java that is intergrated with.. ) and asterisk
is interfaced with this.
i'm able to make calls between pingtel and Voip user,
and also i can able to make call from Sip communicator to pingtel or Voip
phone.
but now i'm can't make calls between 2 sip communicator.. it mean i can
able to make a call and receive.. but i can't hear the  other person voice.
but my voice he can able to hear...
some times i can't able to make (Between 2 sip comm.)call also...

I'm using asterisk 1.4 versoin...

could u tell me any suggestions..

Regards,
nsthi
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