date:20061005

On Thu, Oct 05, 2006 at 09:44:08AM +1000, David Hindmarsh wrote:
 Does anybody have T38passthrough working using the 1.4 Beta?
 
 If so what did you need to do to get it working?
 
 I have 2 SPA-2100 that cannot get a T38 call going via 1.4.

Have a look at issue 7679 on bugs.digium.com (Sipura) and related issue 8078
(Audiocodes)

Regards,

Brian.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Bandwidth requirements

2006-10-05 Thread Erik

Let me paste my old reply to this:

Let's do some calculations on that:

g729a 20ms results in 20 bytes RTP payload in each packet, in order to traverse 
the OSI model there's some headers that need to be added,
as RTP gets an RTP header, the RTP packet gets an UDP header and the UDP 
datagram gets an IP header:
So add a 12 byte RTP header, 8 byte UDP header and a 20 Byte IP header

This results in:
20 byte RTP payload + 12 byte RTP header + 8 byte UDP header + 20 byte IP 
header=60 byte on the ip layer.
Thats a 40 byte overhead (so 2/3 of the packet is just headers :) and were 
still only on the IP layer now)

So to transmit just 1 RTP packet you are actualy transmitting 60 bytes on the 
IP layer, so in order to get the real used bandwidth we
need to knowhow many packets we are sending and on which medium 
(DSL/ethernet/slip/smokesignals):

20 ms results in 50 packets/s so: 50 packet/s *60 bytes/packet=3000 bytes/s
that's 300*8=24000 bit/s total bandwidth on the IP layer so the overhead is 
24000-8000=16000 bit/s.

The fun starts if you are going to send this over DSL, let's continue the 
calculation:

50 packets of 60 byte IP, add the 2 byte PPPoA header for DSL= 62 bytes per 
packet.
However, DSL operates with 53 bytes ATM cells, in which you can fit 48 bytes 
payload (and a 5 byte header) so in order to transmit the 62 bytes of
data you need: 62/48=2 ATM cells.

Why 2 cells you say? Because ATM can't utilize the unused part of cells, so to 
transmit 62 bytes you use the same amount of bandwith (on dsl) as you
would use to transmit 96 (48*2) bytes.

So 1 RTP packet uses 96 bytes on the DSL line, as you already know we have 50 
packets/s so that's 50 packets/s*2 cells=100 Cells/s
100 cells/s * 53 byte = 5300 bytes/s on the DSL line thats 42400 bits/s to 
transmit a 8 kbit/s stream :)

So in order to use 5 simultaneous calls you would need a 1:1 DSL line of at 
least 5*42400=212000 bps so a 256/256 DSL would do, however if
you need 20 calls that would be 20*42400=848000 bps so that would be a 1M/1M 
line (and some bandwidth to spare)

Erik Versaevel


hugolivude wrote:
 Hi,
 
 Age old question it seems but I haven't been able to get a handle on it
 yet.  Let's assume I'm using a g729 codec.  If I wanted to handle 20
 simultaneous calls, how much bandwidth would I need?  Is there a general
 formula for this?
 
 I tried this caluclator:
 http://www.voip-calculator.com/calculator/eipb/
 
 I wasn't sure what Packet Duration to select so I took the default 20ms
 (2 samples) - whatever that means.  I plugged in 5 for the BHT (20
 customers, each getting 3 X 5 minute calls/hour = 5 Erlangs) and the
 default 0.01 for the Blocking.  It worked out to 264 kbps.  Does this
 sound reasonable?  If so great!  A business DSL could support this.
 
 Comments welcome!
 
 Cheers,
 H
 
 
 
 
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



Erik Versaevel
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] snom 360: how to make record button working ?

2006-10-05 Thread Remco Barendse

On Thu, 5 Oct 2006, Joel Hill wrote:

 No worries. Good question, I wasn't sure so I just tested it and it seems that
 the answer is yes it does send the tones to the other side.
 Can I ask why this would matter, I think there could be legal implications of
 recording a call and not notifying the other party. That's why you always get
 the message
 This call may be monitored for training and coaching purposes. Etc..

AFAIK in The Netherlands there is no law to obligatory tell the other 
party that the conversation is / will be recorded, banks do it as a 
standard procedure for example when you are placing forex orders. I think 
even insurance companies do the same when you call in to report a 
claim/damage.  

With audible DTMF tones this function is basically unusable for our 
purposes.

With the recording function already being implemented in * I guess it 
would be trivial to get it working with the SIP info message as well? (Or 
maybe it is intentional behaviour that it will only work out of the box 
with audible DTMF tones)

Cheers!
Remco
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] set verbose 4 in SVN trunk?

In SVN trunk, I see set verbose 4 and set debug 4 no longer work:

asterisk1*CLI set debug 4
No such command 'set debug' (type 'help' for help)
asterisk1*CLI set verbose 4
No such command 'set verbose' (type 'help' for help)

I'm probably being obtuse, but I can't find what these commands are in the
new CLI. Can someone tell me what they are? Or do I just have to start
asterisk with -r ?

More generally, is there a document which lists the CLI changes from 1.2?
(For example, I discovered that sip show peers is now sip list peers)

Thanks,

Brian.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] set verbose 4 in SVN trunk?

2006-10-05 Thread Pryakhin Dimitry

I think its changed to core verbose

Dimitry 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Candler
Sent: Thursday, October 05, 2006 2:55 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] set verbose 4 in SVN trunk?

In SVN trunk, I see set verbose 4 and set debug 4 no longer work:

asterisk1*CLI set debug 4
No such command 'set debug' (type 'help' for help)
asterisk1*CLI set verbose 4
No such command 'set verbose' (type 'help' for help)

I'm probably being obtuse, but I can't find what these commands are in the
new CLI. Can someone tell me what they are? Or do I just have to start
asterisk with -r ?

More generally, is there a document which lists the CLI changes from 1.2?
(For example, I discovered that sip show peers is now sip list peers)

Thanks,

Brian.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] set verbose 4 in SVN trunk?

2006-10-05 Thread Pryakhin Dimitry

I think it changed to core verbose

Dimitry 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Candler
Sent: Thursday, October 05, 2006 2:55 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] set verbose 4 in SVN trunk?

In SVN trunk, I see set verbose 4 and set debug 4 no longer work:

asterisk1*CLI set debug 4
No such command 'set debug' (type 'help' for help)
asterisk1*CLI set verbose 4
No such command 'set verbose' (type 'help' for help)

I'm probably being obtuse, but I can't find what these commands are in the
new CLI. Can someone tell me what they are? Or do I just have to start
asterisk with -r ?

More generally, is there a document which lists the CLI changes from 1.2?
(For example, I discovered that sip show peers is now sip list peers)

Thanks,

Brian.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] answering machine detection

2006-10-05 Thread Raj









i m using asterisk 1.12. and i have followed all the steps define in scratch
installation for answering machine detection, but when i m making a
call i m getting a lot of answering machine is there any thing i have to
change to my extension file . can any one help me out



Raj Ahmed








___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [Asterisk-Java] SipShowPeerAction

2006-10-05 Thread richard Coco


Hi again,

i am still missing something 'cause i am not able to
handle the PeerEntryEvent. The other Events are ok.
Here is what i did.

public void run() throws
IOException,
AuthenticationFailedException,
TimeoutException
{
managerConnection.login();
managerConnection.addEventListener(this);

SipShowPeerAction sipShowPeerAction = new
SipShowPeerAction(2001);
managerConnection.sendAction(sipShowPeerAction);

}
public void onManagerEvent(ManagerEvent event)
{
HashMapString, JButton hmap = new HashMapString,
JButton();
hmap.put(SIP/2000, PresenceGUI.sButton2000);
hmap.put(SIP/2001, PresenceGUI.sButton2001);
hmap.put(SIP/2002, PresenceGUI.sButton2002);
hmap.put(SIP/2003, PresenceGUI.sButton2003);
hmap.put(SIP/2004, PresenceGUI.sButton2004);

if (event instanceof PeerEntryEvent)
{

System.out.println(((PeerEntryEvent)event).getStatus());
}
if (event instanceof PeerStatusEvent)
{
if (((PeerStatusEvent)
event).getPeerStatus().equals(PeerStatusEvent.STATUS_REGISTERED))
{

hmap.get(((PeerStatusEvent)event).getPeer()).setIcon(new
ImageIcon(personal_green.png));
}

if (((PeerStatusEvent)
event).getPeerStatus().equals(PeerStatusEvent.STATUS_UNREGISTERED))
{

hmap.get(((PeerStatusEvent)event).getPeer()).setIcon(new
ImageIcon(personal_gray.png));
}
}
if (event instanceof NewChannelEvent)
{
if (((NewChannelEvent)
event).getState().equals(Ringing))
{

hmap.get(((NewChannelEvent)event).getChannel().substring(0,
8)).setIcon(new ImageIcon(personal_red.png));
}
if (((NewChannelEvent)
event).getState().equals(Ring))
{

hmap.get(((NewChannelEvent)event).getChannel().substring(0,
8)).setIcon(new ImageIcon(personal_red.png));
}
}
if (event instanceof HangupEvent)
{

if(((HangupEvent)event).getChannel().substring(0,
5).equals(SIP/2))
{

hmap.get(((HangupEvent)event).getChannel().substring(0,
8)).setIcon(new ImageIcon(personal_green.png));
}
}
}
}


thx in advance!





--- Tim Panton [EMAIL PROTECTED] wrote:

 
 On 4 Oct 2006, at 16:33, richard Coco wrote:
 
  Hi all,
 
  first of all sorry for the question. I know there
 is
  an asterisk-java mailinglist but i am not
 subscribed
  to this list and i am sure there are asterisk-java
  guru on this list who can help me.
 
  I am trying to get the status of a peer using
  SipShowPeerAction. Unfortunately the getStatus
  method gives me everytime null.
 
  SipShowPeerAction sipShowPeerAction = new
  SipShowPeerAction(2001);
  managerConnection.sendAction(sipShowPeerAction);
  PeerEntryEvent peerEntryEvent = new
  PeerEntryEvent(sipShowPeerAction);
  System.out.println(peerEntryEvent.getStatus());
 
  What wrong with this example? Maybe someone can
 give
  me a working example.
 
 The way Java normally works is that you add register
 yourself as an
 event listener, and the framework then sends you an
 event when  
 something happens.
 
 so your class needs to implement
 ManagerEventListener
 then you say something like :
 
 void doit(){
   managerConnection.addEventListener(this)
   SipShowPeerAction sipShowPeerAction =
 newSipShowPeerAction(2001);
   managerConnection.sendAction(sipShowPeerAction);
 
 }
 
 public void onManagerEvent(ManagerEvent event) {
   if (event instanceof PeerEntryEvent){
   

System.out.println(((PeerEntryEvent)event).getStatus());
   } else {
   System.out.println(Some other event);
   }
 }
 
 
 Tim Panton
 
 www.mexuar.com
 
 
 
 ___
 --Bandwidth and Colocation provided by Easynews.com
 --
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   

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


__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Bandwidth requirements

On Wed, Oct 04, 2006 at 07:51:14PM -0400, hugolivude wrote:
Age old question it seems but I haven't been able to get a handle on
it yet.  Let's assume I'm using a g729 codec.  If I wanted to handle
20 simultaneous calls, how much bandwidth would I need?  Is there a
general formula for this?
I tried this caluclator:
[1]http://www.voip-calculator.com/calculator/eipb/
I wasn't sure what Packet Duration to select so I took the default
20ms (2 samples) - whatever that means.

A 20ms packet duration means that 20ms of audio is stuffed into one IP
packet. Since each packet carries 1/50th of a second of audio, that means
you're generating 50 packets per second for each channel.

With g729 your audio is 8000 bits per second.

The overhead on each packet is 20 bytes (IP) + 8 bytes (UDP) + 12 bytes
(RTP) = 40 bytes or 320 bits.

So your bandwidth requirement per channel is:
- 8000 bits per second for payload
- 320x50 = 16000 bits per second for overhead
making a total of 24000 bits per second.

20 simultaneous calls is therefore 480,000 bits per second.

That is a bit of an underestimate though, because it doesn't include any
layer 2 framing overhead (i.e. for encapsulating the IP frames in the
underlying medium). For example, if it were HDLC serial on a leased line,
that would be just 2 bytes per frame for flags, maybe a couple of bytes for
CRC, plus occasional bit-stuffing.

However on ADSL, you have to add the 15% ATM cell tax. And you would be wise
to add 20% headroom (i.e. so your line is not more than 80% full)

As you can see, the packetisation overhead is twice as large as the useful
data you're transporting. You can reduce this by increasing the packet
duration, but that increases the latency of your audio (and ADSL links
already add 20-30ms of latency themselves). Too much latency is
objectionable to users.

I have read that if you use IAX2 trunking it's able to combine audio from
multiple streams into a single packet, thus sharing the overhead between
them, but I have no experience of this myself.

HTH,

Brian.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] spandsp logs?

2006-10-05 Thread Francesco Francesconi

Hello everybody.
I am trying to build a fax gateway system based on spandsp.
Everything is working fine except that I don't know how to keep track of
the final status of the fax call (if the fax went through ok or failed
somewhere in the transmission).
I've seen in the src directory of spandsp (I'm using the 0.0.2pre26
version) a logging module that (I suppose) gets compiled with spandsp,
but wasn't able to find out how to enable it nor where it saves logs at.
Anybody knows how I can achieve this?
Thanks in advance

Francesco
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: Bandwidth requirements

2006-10-05 Thread Benny Amorsen

 h == hugolivude  [EMAIL PROTECTED] writes:

h I wasn't sure what Packet Duration to select so I took the default
h 20ms (2 samples) - whatever that means. I plugged in 5 for the BHT
h (20 customers, each getting 3 X 5 minute calls/hour = 5 Erlangs)
h and the default 0.01 for the Blocking. It worked out to 264 kbps.
h Does this sound reasonable? If so great! A business DSL could
h support this.

DSL has excessive overhead for small packets, due to ATM. No matter
which codec you pick, you probably won't get below 2 ATM frames per
voice packet. So for G.729 and 20ms frames, that means
50 packets/sec * 2 frames/packet * 48 bytes/frame * 8bits/byte =
38.4kbps per voice call, per direction. A bit worse if you include the
ATM overhead, then it's 50 * 2 * 53 * 8 = 42.4kbps.


/Benny


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Bandwidth requirements

2006-10-05 Thread raphael Jacquot


Brian Candler wrote:


However on ADSL, you have to add the 15% ATM cell tax. And you would be wise
to add 20% headroom (i.e. so your line is not more than 80% full)


ATM cell tax is actually 10% as there's 5 header bytes for each 53 bytes 
cell,

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] set verbose 4 in SVN trunk?

On Thu, Oct 05, 2006 at 03:08:16PM +0700, Pryakhin Dimitry wrote:
 I think its changed to core verbose

asterisk1*CLI core debug 10
Core debug was 1 and is now 10
asterisk1*CLI core verbose 10
Verbosity was 4 and is now 10

Thank you. Yes I was being obtuse :-)

Regards,

Brian.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Silience on random calls

2006-10-05 Thread Pryakhin Dimitry




I faced following 
problem lately. Am running Asterisk 1.2.7.1 [trixbox build] 2 
Linksys SPA2102 7 DLINK DVG2004s Time after time I get an error, 
when an extension recieves a call - it rings, but when you pick up the handset 
you cant hear anything. Just silience. You can hung up and recall - and it might 
be working next time and might be the same problem. Also found out what 
pressing any digit on a phone helps to bring up the sound... Which is weird. 
PC has two eth interfaces, one with pubic ip, one with private. VoIP 
gateways are connected to the * thru local leg with addres 192.168.1.227. So 
they are sitting in one network and in one swithch, what eliminates any relation 
to NAT problem. Does anybody faced the same problem? Or might was "walking" 
just right next to mine?
Dimitry
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Problems with Dial In - Dial Out via SIP - no voice

2006-10-05 Thread Christian Peter

Hi list,

I hope somebody already had this kind of problem:

I want to dial in from a SIP provider and then (in the incoming section
for the provider) do a SIP Dial() out via the same provider. The dialled
out phone number rings and the calls get connected but I can't hear any
voice. If I do a monitor() I don't see the wav file growing, so I guess
there is no RTP stream. Also a rtp debug does not show any data.

Can I do something to test further, or, can anybody point me to the SIP
messages which are important for debugging this? I had a look at them
but with my limited knowledge I can't see where the problem is.

I tested Asterisk 1.2.5 and current SVN 1.2.

Thanks in advance

Regards

Christian Peter

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Problems with Dial In - Dial Out via SIP - no voice

2006-10-05 Thread Christian Peter

Sorry to reply to myself,
if I dial out with ISDN it works. I don't have a different SIP account
to test dial in SIP_PROVIDER_1 and dial out with SIP_PROVIDER_2.


Am Donnerstag, den 05.10.2006, 11:14 +0200 schrieb Christian Peter:
 Hi list,
 
 I hope somebody already had this kind of problem:
 
 I want to dial in from a SIP provider and then (in the incoming section
 for the provider) do a SIP Dial() out via the same provider. The dialled
 out phone number rings and the calls get connected but I can't hear any
 voice. If I do a monitor() I don't see the wav file growing, so I guess
 there is no RTP stream. Also a rtp debug does not show any data.
 
 Can I do something to test further, or, can anybody point me to the SIP
 messages which are important for debugging this? I had a look at them
 but with my limited knowledge I can't see where the problem is.
 
 I tested Asterisk 1.2.5 and current SVN 1.2.
 
 Thanks in advance
 
 Regards
 
 Christian Peter
 
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TNT Max Password reset

2006-10-05 Thread Steve Kennedy

On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote:

 On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
 Anyone have happen know how to reset the password on a TNT Max? Thanks.
 Does your asking here suggest that the the MAX's can do, say, voice
 gateway service?  Protocols?  Codecs?

Ascent TNT's with the right software and hardware can do SIP, E1
termination/origination, and all sorts of codecs.

Similar functionality to Cisco AS5200'ish.


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Message count requested for mailbox [EMAIL PROTECTED] but voicemail not loaded.

2006-10-05 Thread raviprakash sunkara

Hello Can any help this messages .. What it meanMessage count requested for mailbox [EMAIL PROTECTED] but voicemail not loaded. 
  -- Thanks and RegardsRavi Prakash Sunkara		[EMAIL PROTECTED] 	M:+91 9985077535
O:+91 40 23114549F:+91 40 40208727 		[EMAIL PROTECTED]www.hyperion-tech.com
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Message count requested for mailbox [EMAIL PROTECTED] but voicemail not loaded.

2006-10-05 Thread Francesco Francesconi

Hello.
This means that one of your clients (maybe ext 9002 in the context
from-sip) is  requesting VM messages  (probably your hardware  has MWI
enabled)  but you do not  have  voicemail  module  loaded/configured.
Greetings
Francesco

raviprakash sunkara wrote:
 Hello 
 Can  any help this messages .. What it mean

 Message count requested for mailbox [EMAIL PROTECTED] but voicemail not
 loaded. 



 
 -- 
 Thanks and Regards
 Ravi Prakash Sunkara
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 M:+91 9985077535
 O:+91 40 23114549
 F:+91 40 40208727
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 www.hyperion-tech.com http://www.hyperion-tech.com
 

 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
   

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Re: extensions.conf strangeness

On Wed, Oct 04, 2006 at 12:20:40AM -0700, Martin Joseph wrote:
 Are there any debug tools which can show the thought process as a
 dial-plan is processed - for example, what patterns are tried and in what
 order?
 
 You can say show dialplan from the command line...
 
 Don't know if this helps?

Well, it shows each context as a separate list of tests, which at least
gives me the sort order. But that still doesn't explain why context A which
includes W,X,Y,Z behaves differently from context B which also includes
W,X,Y,Z

Is there a debug mode which can say:

dialplan: trying to match 611 against pattern _1X: failed
 dialplan: trying to match 611 against pattern _2X: failed
 dialplan: trying to match 611 against pattern _6X.: matched

?

Cheers,

Brian.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] answering machine detection

2006-10-05 Thread Matt Florell


Are you possibly using GnuDialer or VICIDIAL?

If so, you would probably get a better response by posting to their
forums rather than the general asterisk-users list:

GuDialer Forum:
http://forum.acmcllc.com/

VICIDIAL Forum:
http://www.eflo.net/VICIDIALforum


One more thing, there is no Asterisk 1.12.

MATT---


On 10/5/06, Raj [EMAIL PROTECTED] wrote:





i m using asterisk 1.12. and i have followed all the steps define in scratch
installation for answering machine detection,  but when i  m making a call i
m getting a lot of answering machine  is there any thing i have to change to
my extension file . can any one help me out



Raj Ahmed


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

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




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] snom 360: how to make record button working ?

2006-10-05 Thread sip

That varies from location to location, really. In Georgia, for instance, only
ONE party need know the recording is taking place (calling or receiving)
without a warrant. In some countries, neither party need know, etc, etc. 

N.


On Thu, 05 Oct 2006 15:25:28 +1000, Joel Hill wrote
 No worries. Good question, I wasn't sure so I just tested it and it 
 seems that the answer is yes it does send the tones to the other 
 side. Can I ask why this would matter, I think there could be legal 
 implications of recording a call and not notifying the other party. 
 That's why you always get the message This call may be monitored 
 for training and coaching purposes. Etc..
 
 Cheers,
 Joel.
 
 Remco Barendse wrote:
  Thanks for this, I was looking for this too.
 
  Will the DTMF tone be audible to the other side? (In other words will they 
  know something is happening)
 
  On Thu, 5 Oct 2006, Joel Hill wrote:
 

  Hi Noro,
 
  Depending on what firmware you have this is the way to go.
  Go to the Functions keys page, then look for the Record button, Change the
  type to DTMF and in number put in *1 which is the default Asterisk 
  recording
  function.
 
  Hope this helps
 
  Cheers,
 
  Joel
  Asterisk IT
  www.asteriskit.com.au
 
 
  noro kamen wrote:
  
  Hi,
 
  I'd like to make record button working on snom 320/360 + asterisk.
 
  As I learned from wireshark output,  the phone produces SIP info
  message Record: on, while record button pressed.
 
  Can anybody give me an advice, how to teach asterisk to understand
  that SIP info message and start recording ?
 
  TIA
  noro
  ___
  --Bandwidth and Colocation provided by Easynews.com --
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

  ___
  --Bandwidth and Colocation provided by Easynews.com --
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
  
  ___
  --Bandwidth and Colocation provided by Easynews.com --
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users

 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] no callerid from PSTN using TDM2400P

2006-10-05 Thread Bob Chiodini

On Wed, 2006-10-04 at 14:58 -0700, Naija Man wrote:
 Hello all,
 
 Asterisk 1.2.8
 zaptel 1.2.6
 Hardware: digium TDM2422P
 
 I have a fully configured asterisk system with POTS line for PSTN
 access. I am not receiving the callerid for incoming calls from the
 PSTN. I get the following error message. 
 
 -- Starting simple switch on 'Zap/3-1'
 Oct  3 22:53:14 NOTICE[17948]: chan_zap.c:6061 ss_thread: Got event 18
 (Ring Begin)...
 Oct  3 22:53:16 ERROR[17948]: callerid.c:276 callerid_feed: fsk_serie
 made mylen  0 (-22) 
 Oct  3 22:53:16 WARNING[17948]: chan_zap.c:6091 ss_thread: CallerID
 feed failed: Success
 Oct  3 22:53:16 WARNING[17948]: chan_zap.c:6135 ss_thread: CallerID
 returned with error on channel 'Zap/3-1
 

I've been seeing similar problems, but they are intermittent.  I'm in
the US.  It seems that restarting asterisk clears up the problem for a
while, but it may be only coincidence.

I'm running Trixbox 1.1.0, Asterisk 1.2.12.1, Zaptel 1.2.9.1.  I did not
have a problem with [EMAIL PROTECTED] v 2.8.  H/W TDM11B.

Still looking...

Bob...
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] New Version of Tycho Voicemail Manager released

2006-10-05 Thread Arnd Vehling


Hi,

we are releasing an update of our Tycho Voicemail Manager. The update
to Beta 0.2 contains a bugfix and a couple of improvements over the 0.1 version:

Bug fix:

* missing Channel Type added to extension subscription

Improvements:

* adjustable refresh interval (voicemail)
* manual refresh button (voicemail)
* re-open windows after application restart (extensions, voicemail)
* Support Forum menu item - Web Link to our support Forum
* Reset Perspective menu item - Resets the windows to it's defaut 
locations

Please note that this version will delete your preference settings as well as 
any defined voicemailboxes and extensions. Help us and report all 
problems/bugs to our support forum. Thank you.


Note for Trixbox Users: When subcribing to an extension please use the 
context from-internal in the context field of the subscription menu.


For more Info, documentation and Downloads go here: http://sip-syndication.com

best regards,

  Arnd

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Extremely choppy sound on some of our POTS network calls; goes away with mute

2006-10-05 Thread sdgesa gaeharth

1)Can anyone tell me how to do this on a Polycom 501?2)Can you explain why you think this any why it ony happens on some calls?ThanksAndres [EMAIL PROTECTED] wrote:   For about 20% of the calls to the outside world, the voice on the  other end of an outside line is incredibly choppy.   Enough to where  we have to hang up and call on a cell phone. It is always the same  numbers that are choppy.  The funny thing is, if I press mute while  talking on a choppy call, the choppiness goes away completely.  Maybe you have silence suppression enabled on your phones.  Try to disable it and see if it helps.  --
 AndresTechnical Supporthttp://www.telesip.net___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users 
		Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Extremely choppy sound on some of our POTSnetwork calls; goes away with mute









What codec are you using?



How many phone? What load is the server
under?















From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of sdgesa gaeharth
Sent: 05 October 2006 13:22
To:
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
Extremely choppy sound on some of our POTSnetwork calls; goes away with mute





1)Can anyone tell me how to do this on a Polycom 501?

2)Can you explain why you think this any why it ony happens on some calls?

Thanks

Andres
[EMAIL PROTECTED] wrote:




 For about 20% of the calls to the outside world, the voice on the 
 other end of an outside line is incredibly choppy. Enough to where 
 we have to hang up and call on a cell phone. It is always the same 
 numbers that are choppy. The funny thing is, if I press mute while 
 talking on a choppy call, the choppiness goes away completely.

 

Maybe you have silence suppression enabled on your phones. Try to 
disable it and see if it helps.



 



-- 
Andres
Technical Support
http://www.telesip.net

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



  







Yahoo! Messenger with Voice. Make
PC-to-Phone Calls to the US
(and 30+ countries) for 2¢/min or less.






___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] India:Reliance - E1configuration using TE110P

2006-10-05 Thread Rajkumar S


Hi,

I bought an asterisk TE110P to connect to our Reliance Infocomm E1
line to asterisk, I have loaded the driver, but looking for an
appropriate zaptel.conf and zapata.conf. I googled a lot but there
does not seems to be any india specific configuration. If any one has
successfully configured this on a Reliance E1 line, I would be very
grateful if you can share the appropriate entries in zaptel and zapata
conf.

with warm regards,

raj
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Call Center requirements

2006-10-05 Thread Todd Houle Asterisk

Hi Guys- While I played a little with Asterisk a year or so ago, I'm getting ready to roll out a project now that I think is perfect for it.  My friend with with a commercial solution he has been very unhappy with and is thinking of replacing it with Asterisk.  Below are his requirements.  Anything here jump out as a problem? I'm thinking of purchasing a few Digiium card - not sure which we need yet...   and finding a box to run it on. The only part I'm not sure is how to address is having the client record auto-appear on screen when the call comes in.  I did see plug ins for recording the calls...    Is asterisk the best solution for this? thanks    ToddBegin forwarded message:From: "A. Pathuri" [EMAIL PROTECTED]Date: October 2, 2006 2:51:32 AM EDTTo: Todd Houle [EMAIL PROTECTED]Subject: Call Center requirements Todd,Here is the brief doc you requested.The process that we need is pretty simple...We get a bunch of DID (Direct Inward Dialing) numbers from SBC.As we get a client, we assign them a DID #.They forward their existing phones to their DID number when their linesare busy or after hours.The DID # is programmed into the telephony system so we can program thecaller ID, and enter the appropriate script to pop up when that numbercomes through.When a call comes in, I would like to have all calls automaticallyrecorded without any of the call agents having to press a record buttonfor each call.We also current have conference call functionality where we can connectone caller to another caller (used when the ER needs to speak to adoctor).Ideally also, I would like the recorded calls to sort by client andstore in the appropriate clients folder, which then can beautomatically zipped and sent via email to the clients inbox at anydesired interval.We are also developing a web-based app where the details of each callcan be entered ( a sort of call log) so the clients can also log into aweb interface and see the details of each call (currently, most clientsget their call logs via fax in the am and at midnight).It would be great if somehow, the caller ID on the server/astericks canautomatically pull up the appropriate clients profile from our web app,so the details can be entered into the correct profile.  Otherwise, foreach call that comes in, the call agent has to pull up the clientsprofile while the caller is on the phone, before s/he can take down thedetails of the call.This is really rough, but I hope it gives the basic idea.  We candiscuss in further detail once you take a look at this.Ofcourse, as well it would be great to be able to setup a co-location inIndia utilizing the same infrastructure.Regards,Anand ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] GXP - 2000 BLF









Hello,



I have been trying to get my Grandstream busy line filter to
work for ages..



All the lights flash as they are supposed to.



If one Grandstream 7000 calls another Grandstream 7003 I can
use Grandstream 7002 to pick the call up pressing the BLF button and all works
fine.



However if I call Grandstream 7000 with a mobile phone and
try to pickup the call with Grandstream 7002 all I get is a 603 error on
Grandstream 7002.



I am using firmware 1.1.12 for the Grandstream and 1.2.12.1
version of asterisk





This is the error I get from my log..



if some one could please help

Oct 5 12:12:51 DEBUG[7723] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060Oct 5 12:12:51 VERBOSE[8828] logger.c: -- Executing NoOp(SIP/7003-b721be28, **7002) in new stackOct 5 12:12:51 VERBOSE[8828] logger.c: -- Executing Pickup(SIP/7003-b721be28, 7002) in new stackOct 5 12:12:51 DEBUG[8828] app_directed_pickup.c: No originating channel found.Oct 5 12:12:51 DEBUG[8828] app_directed_pickup.c: No call pickup possible...Oct 5 12:12:51 VERBOSE[8828] logger.c: == Spawn extension (inbound-from-stem, **7002, 2) exited non-zero on 'SIP/7003-b721be28'Oct 5 12:12:51 DEBUG[7716] channel.c: Avoiding initial deadlock for 'SIP/7003-b721be28'



SIP[7000]type=friendcontext=inbound-from-stemSubscribecontext=BLFsecret=*host=dynamiccanreinvite=nocallgroup=2pickupgroup=2[EMAIL PROTECTED]username=7000dtmfmode=rfc2833callerid=STEM 17524543545qualify=yes





EXTENSIONS



[default]

include = stem

include = to-siemens

include = BLF

include = BLF_group_pickup





[stem]

;exten STEM GROUP = 01752 692205

exten = 123454,1,Ringing

exten = 123454,n,Wait(1)

exten = 123454,n,Answer()

exten = 123454,n,NoOp(${CALLERID(all)})

exten = 123454,n,SetCIDName(Outside Caller)

exten = 123454,n,Set(CALLERID(number)=9${CALLERIDNUM})

exten = 123454,n,NoOp(${CALLERID(all)})

exten = 123454,n,Macro(stdexten2,7003,${STEMGROUP},20)



;exten 7000 = 01752 692204

exten = 123455,1,Ringing

exten = 123455,n,Wait(1)

exten = 123455,n,Answer()

exten = 123455,n,NoOp(${CALLERID(all)})

exten = 123455,n,SetCIDName(Outside Caller)

exten = 123455,n,Set(CALLERID(number)=9${CALLERIDNUM})

exten = 123455,n,NoOp(${CALLERID(all)})

exten = 123455,n,Macro(stdexten2,7000,${stem},20)



;exten 7001 = 01752 692283

exten = 123456,1,Ringing

exten = 123456,n,Wait(1)

exten = 123456,n,Answer()

exten = 123456,n,NoOp(${CALLERID(all)})

exten = 123456,n,SetCIDName(Outside Caller)

exten = 123456,n,Set(CALLERID(number)=9${CALLERIDNUM})

exten = 123456,n,NoOp(${CALLERID(all)})

exten = 123456,n,Macro(stdexten2,7001,${stem1},20)





[internal]

;Internal Extensions

exten = _7XXX,1,Ringing

exten = _7XXX,n,Wait(1)

exten = _7XXX,n,Answer()

exten = _7XXX,n,Set(FOO1=${CHANNEL:4})

exten = _7XXX,n,Set(FOO2=${CUT(FOO1,-,1)})

exten = _7XXX,n,Set(CALLERID(number)=${FOO2})

exten = _7XXX,n,Macro(stdexten,${EXTEN},SIP/${EXTEN})





[inbound-from-pstn] ; inbound calls to this context from
outside lines

include = default





[inbound-from-sip]

include = default



[inbound-from-local]

;from sip default context used.. requires hints

include = voicemail

include = provider

include = outbound

;include = stem ;for hints





[inbound-from-stem]

include = BLF

include = internal

include = DefExt

include = voicemail

include = outbound

include = BLF_group_pickup

include = feature-cfu

include = feature-cfna

include = feature-cfb



[inbound-from-logicall]

include = internal

include = DefExt

include = voicemail

include = outbound

include = BLF_group_pickup

include = feature-cfu

include = feature-cfna

include = feature-cfb



;Test section for BLF on Grandstreams for Stem

[BLF_group_pickup]

include =inbound-from-stem

exten = _**.,1,NoOp(${EXTEN})

exten = _**.,2,Pickup(${EXTEN:2})

exten = _**.,3,Hangup



[BLF]

include =inbound-from-stem

exten =7000,hint,SIP/7000

exten =7000,1,Dial(SIP/7000,20,r)

exten =7001,hint,SIP/7001

exten =7001,1,Dial(SIP/7001,20,r)

exten =7002,hint,SIP/7002

exten =7002,1,Dial(SIP/7002,20,r)

exten =7003,hint,SIP/7003

exten =7003,1,Dial(SIP/7003,20,r)

exten =7004,hint,SIP/7004

exten =7004,1,Dial(SIP/7004,20,r)

exten =7005,hint,SIP/7005

exten =7005,1,Dial(SIP/7005,20,r)

exten =7006,hint,SIP/7006

exten =7006,1,Dial(SIP/7006,20,r)






































___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: AEL2 #include madness in Asterisk 1.4 - Murf?

2006-10-05 Thread Steve Murphy

On Thu, 2006-10-05 at 01:08 -0700, [EMAIL PROTECTED] wrote:
 Asterisk 1.4 beta2.
  
 My top level /etc/asterisk/extensions.ael has the following
 two lines:
  
 #include include/syst/extensions.ael
 #include include/btck/extensions.ael
 
 Here is the console output on Asterisk load.
  
 app_system.so = (Generic System() application)
 [Oct  4 15:48:15] NOTICE[1143]: pbx_ael.c:3798
 pbx_load_module: Starting AEL load process.
 [Oct  4 15:48:15] NOTICE[1143]: pbx_ael.c:3805
 pbx_load_module: AEL load process: calculated config file name
 '/etc/asterisk/extensions.ael'.
 [Oct  4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
 --Read in included
 file /etc/asterisk/include/syst/extensions.ael, 4130 chars
 [Oct  4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
 --Read in included file /etc/asterisk/include/syst/macros.ael,
 1463 chars
 [Oct  4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
 --Read in included
 file /etc/asterisk/include/syst/dundiapps.ael, 758 chars
 [Oct  4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
 --Read in included file /etc/asterisk/include/syst/rdapps.ael,
 275 chars
 [Oct  4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
 --Read in included
 file /etc/asterisk/include/btck/extensions.ael, 1385 chars
 [Oct  4 15:48:15] NOTICE[1143]: pbx_ael.c:3808
 pbx_load_module: AEL load process: parsed config file name
 '/etc/asterisk/extensions.ael'.
 [Oct  4 15:48:15] ERROR[1143]: pbx_ael.c:1162 check_goto:
 Error: file /etc/asterisk/include/syst/extensions.ael, line
 157-157: goto:  no label remote exists in the current
 extension!
 [Oct  4 15:48:15] ERROR[1143]: pbx_ael.c:1162 check_goto:
 Error: file /etc/asterisk/include/syst/extensions.ael, line
 159-159: goto:  no label local exists in the current
 extension!
 [Oct  4 15:48:15] ERROR[1143]: pbx_ael.c:3821 pbx_load_module:
 Sorry, but 0 syntax errors and 2 semantic errors were
 detected. It doesn't make sense to compile.
 pbx_ael.so = (Asterisk Extension Language Compiler)
  
 Here's the context
 from /etc/asterisk/include/syst/extensions.ael, that contains
 lines 157 that the parser is complaining about:
  
148  context syst_Route {
149
150  _[*0123456789]. = {
151  NoOp(*** Originated call ${CALLERID} -
 ${EXTEN});
152  Set(TMP=${CALLERID(number)});
153  SysLogger(This is a test message);
154  FastAGIConnectGet(CALLERID);
155  ChanIsAvail(SIP/${EXTEN});
156  if (${AVAILCHAN} = ) {
157  goto remote;
158  } else {
159  goto local;
160  }
161  remote:
162  NoOp(REMOTE);
163  Set(PATH=
 ${DUNDILOOKUP(3254103,DUNDIRegistr)});
164  //Set(PATH=
 ${DUNDILOOKUP(${EXTEN},DUNDIRegistr)});
165  Dial(${PATH});
166  Hangup();
167  local:
168  NoOp(LOCAL);
169  Dial(SIP/${EXTEN});
170  Hangup();
171
172  }
173  }
  
 As you can quite clearly see, labels 'remote' and 'local' DO
 exist in the syst_Route context.
  
 Now, if I switcheroo the two includes around in the top
 level /etc/asterisk/extensions.ael, to:
  
 #include include/btck/extensions.ael
 #include include/syst/extensions.ael
 
 and reload Asterisk, I get:
  
 [Oct  4 15:57:28] NOTICE[1202]: pbx_ael.c:3813
 pbx_load_module: AEL load process: compiled config file name
 '/etc/asterisk/extensions.ael'.
 [Oct  4 15:57:28] NOTICE[1202]: pbx_ael.c:3816
 pbx_load_module: AEL load process: merged config file name
 '/etc/asterisk/extensions.ael'.
 [Oct  4 15:57:28] WARNING[1202]: pbx.c:6194
 ast_context_verify_includes: Context 'syst_PSTNStart' tries
 includes nonexistent context 'syst_AppACDQueue'
 [Oct  4 15:57:28] WARNING[1202]: pbx.c:6194
 ast_context_verify_includes: Context 'btck_CallStart' tries
 includes nonexistent context 'syst_ACD'
 [Oct  4 15:57:28] NOTICE[1202]: pbx_ael.c:3819
 pbx_load_module: AEL load process: verified config file name
 '/etc/asterisk/extensions.ael'.
 pbx_ael.so = (Asterisk Extension Language Compiler)

[asterisk-users] Re: Bandwidth requirements

2006-10-05 Thread Benny Amorsen

 rJ == raphael Jacquot [EMAIL PROTECTED] writes:

rJ ATM cell tax is actually 10% as there's 5 header bytes for each 53
rJ bytes cell,

For VoIP the cell tax is much larger. In the example, each RTP packet
contains 20 useful bytes and 40 bytes IP overhead. 60 bytes doesn't
fit in one cell, so you end up with 106 bytes at the ATM layer to
transport 20 bytes of G.729. The ATM-caused overhead is thus 46 bytes
per voice packet, thereby making the needed bandwidth 77% larger.

All in all VoIP over ADSL adds 430% overhead, when using G.729 and
20ms packets. Lovely, isn't it?


/Benny


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Extremely choppy sound on some of our POTSnetwork calls; goes away with mute

2006-10-05 Thread sdgesa gaeharth

Below is the text of my original post. I am not sure what Codec we are using. The "Codec Preferences" phone setting shows, in order of preference, G.711u, G.711A, G.729ABWe are running asterisk-1.2.4 with zaptel-1.2.7 on Fedora Core 4-2.6.14-1.1656_FC4smp. It is installed on a Dell PE 2500 with 2x900 MHz processors and 1 Gb RAM and 1 SCSI Disk. The server has a Digium TDM400P card which is connected to 4 POTS lines. The server is also connected to a 100MB switched LAN where we have about 20 Polycom 501 phones with the latest firmware updates. Nothing else runs on the server except an ftp daemon which is never used except when a phone reboots.For about 20% of the calls to the outside world, the voice on the other end of an outside line is incredibly choppy. Enough to where we have to hang up and
call on a cell phone. It is always the same numbers that are choppy. The funny thing is, if I press mute while talking on a choppy call, the choppiness goes away completely.I have tried: turning off ACPI, turning off APCI, moving the card to another PCI slot, changing the RX/TX gains. There are no shared IRQs. I have tested the lines by unplugging them from the asterisk server and plugging them directly into an analogue phone. Using "cat /proc/interrupts; sleep 10 ; cat /proc/interrupts" I see that there are about 1,000 interrupts per seconds between the card and the CPU.I do not think it is a network congestion problem as intra-office communications as well as voicemail retrieval are always perfect. The Voip does not go over any routers, just a max of 2 switches with a 1GB trunk. This happens even off-hours when the network isnt being used at
all. There are never more than 2 people on the phone at the same time and it is definitely not an over-utilized processor.I have trying to figure this out for 2 months on and off with no success any help is appreciated. ThanksAndrew Shelton [EMAIL PROTECTED] wrote:

What codec are you using?How many phone? What load is the server under?From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of sdgesa gaeharth Sent: 05 October 2006 13:22 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Extremely choppy sound on some of our POTSnetwork calls; goes away with mute1)Can anyone tell me how to do this on a Polycom 501?2)Can you explain why you think this any why it ony happens on some calls?ThanksAndres [EMAIL PROTECTED] wrote: For about 20% of the calls to the outside world, the voice on theother end of an outside line is incredibly choppy.
Enough to wherewe have to hang up and call on a cell phone. It is always the samenumbers that are choppy. The funny thing is, if I press mute whiletalking on a choppy call, the choppiness goes away completely. Maybe you have silence suppression enabled on your phones. Try to disable it and see if it helps. -- Andres Technical Support http://www.telesip.net___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update
options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Get your email and more, right on the new Yahoo.com
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Re: Bandwidth requirements

2006-10-05 Thread J. Oquendo


Benny Amorsen wrote:

rJ == raphael Jacquot [EMAIL PROTECTED] writes:



rJ ATM cell tax is actually 10% as there's 5 header bytes for each 53
rJ bytes cell,

For VoIP the cell tax is much larger. In the example, each RTP packet
contains 20 useful bytes and 40 bytes IP overhead. 60 bytes doesn't
fit in one cell, so you end up with 106 bytes at the ATM layer to
transport 20 bytes of G.729. The ATM-caused overhead is thus 46 bytes
per voice packet, thereby making the needed bandwidth 77% larger.

  


CRTP solves this issue (40byte waste)

--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 


The happiness of society is the end of government.
John Adams

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] snom 360: how to make record button working ?

2006-10-05 Thread Joe Pukepail

There was a patch to get this working, looks like it has been abandoned, though. Should give you a starting point to get it working, or perhaps a bounty would get someone interested in getting it usable and committed. 


http://bugs.digium.com/view.php?id=4845
On 10/4/06, Joel Hill [EMAIL PROTECTED] wrote:
Hi Noro,Depending on what firmware you have this is the way to go.Go to the Functions keys page, then look for the Record button, Change
the type to DTMF and in number put in *1 which is the default Asteriskrecording function.Hope this helpsCheers,JoelAsterisk ITwww.asteriskit.com.au
noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output,the phone produces SIP info message Record: on, while record button pressed.
 Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___
 --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: 
http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list
To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TNT Max Password reset

2006-10-05 Thread James


I have five MAX TNT's runnig with SIP and g.729.
They will do E1's, T1's, T3's.

James Taylor
1-903-793-1956


- Original Message - 
From: Steve Kennedy [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Thursday, October 05, 2006 4:28 AM
Subject: Re: [asterisk-users] TNT Max Password reset



On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote:


On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
Anyone have happen know how to reset the password on a TNT Max? 
 Thanks.

Does your asking here suggest that the the MAX's can do, say, voice
gateway service?  Protocols?  Codecs?


Ascent TNT's with the right software and hardware can do SIP, E1
termination/origination, and all sorts of codecs.

Similar functionality to Cisco AS5200'ish.


Steve

--
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] two asterisk and one NBX system

2006-10-05 Thread jose diaz


We have three servers: Two asterisk and one NBX 3COM.
The connection between Asterisk1 and Asterisk2 is with IAX2.
The connection between  Asterisk2 and NBX is with a Digium analog 
TDM400P (2FXO and 2 FXS)


The dial plan Asterisk1: 3XXX
The dial plan Asterisk2: 2XXX
The dial plan NBX: 1XXX

The system work well, but the call from Asterisk1 to NBX fail. I'm using 
the IAX2 protocol to call from asterisk1 to asterisk2, i need to 
trasnfer the call to the NBX. How i can to make that?


Regards,

Jose Diaz

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Video Conference

2006-10-05 Thread Noah Miller


Hi Bilal -


We need to apply Video conference, can asterisk
support this?


No.  Asterisk supports video calls between two end points, but not
video conferences with three or more participants.

There is a bounty for someone to add this feature, but nobody has
successfully implemented it yet.



What I need for that?


Something else.  You can get video conferencing software, or if you
have the right hardware you can use it.  There are many hardware video
conferencing units available from Polycom, Tandberg, Sony, etc.


- Noah
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] TNT Max Password reset

2006-10-05 Thread asterisk

Hello james,
I have 1 max with pri, only used for incomming data call.
It is a old box, where to find firmware for this unit ?
If a can use it for voice

Ps: i leave in France..

Many thanks...
 

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de James
Envoyé : jeudi 5 octobre 2006 15:43
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] TNT Max Password reset

I have five MAX TNT's runnig with SIP and g.729.
They will do E1's, T1's, T3's.

James Taylor
1-903-793-1956


- Original Message -
From: Steve Kennedy [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, October 05, 2006 4:28 AM
Subject: Re: [asterisk-users] TNT Max Password reset


 On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote:

 On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
 Anyone have happen know how to reset the password on a TNT Max? 
  Thanks.
 Does your asking here suggest that the the MAX's can do, say, voice
 gateway service?  Protocols?  Codecs?

 Ascent TNT's with the right software and hardware can do SIP, E1
 termination/origination, and all sorts of codecs.

 Similar functionality to Cisco AS5200'ish.


 Steve

 -- 
 NetTek Ltd  UK mob +44-(0)7775 755503
 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
 Euro Tech News Blog http://eurotechnews.blogspot.com
 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TNT Max Password reset

2006-10-05 Thread Don


We used to use em...
I believe you can just use a serial connection to them and reset them...
Could be mistaken been a couple years now...

- Original Message - 
From: James [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Thursday, October 05, 2006 9:43 AM
Subject: Re: [asterisk-users] TNT Max Password reset



I have five MAX TNT's runnig with SIP and g.729.
They will do E1's, T1's, T3's.

James Taylor
1-903-793-1956


- Original Message - 
From: Steve Kennedy [EMAIL PROTECTED]

To: asterisk-users@lists.digium.com
Sent: Thursday, October 05, 2006 4:28 AM
Subject: Re: [asterisk-users] TNT Max Password reset



On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote:


On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
Anyone have happen know how to reset the password on a TNT Max? 
 Thanks.

Does your asking here suggest that the the MAX's can do, say, voice
gateway service?  Protocols?  Codecs?


Ascent TNT's with the right software and hardware can do SIP, E1
termination/origination, and all sorts of codecs.

Similar functionality to Cisco AS5200'ish.


Steve

--
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.407 / Virus Database: 268.12.13/463 - Release Date: 10/4/2006




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Re: extensions.conf strangeness

2006-10-05 Thread Michael Neuhauser

On Thu, 2006-10-05 at 11:12 +0100, Brian Candler wrote:
 Is there a debug mode which can say:
 
 dialplan: trying to match 611 against pattern _1X: failed
  dialplan: trying to match 611 against pattern _2X: failed
  dialplan: trying to match 611 against pattern _6X.: matched

No, there isn't (I assume to keep this central part as fast as possible,
i.e., even if (option_debug) ... costs time and pollutes the cache).

I've created and attached a one line patch (for 1.4 branch, r44464) that
should give you the info you need (sort of). But be aware that I haven't
tested it on 1.4 (only on 1.2, but things are different there). Only use
this patch on a test system as it will generate massive amounts of
output and will considerably slow down call handling.
-- 
Dr. Michael Neuhauser  mailto:[EMAIL PROTECTED]
Firmix Software GmbH  sip:[EMAIL PROTECTED]
Vienna/Austria/Europe   tel:+43-1-7890849-30
Linux Development and Services http://www.firmix.at/
Index: main/pbx.c
===
--- main/pbx.c	(revision 44464)
+++ main/pbx.c	(working copy)
@@ -952,6 +952,7 @@
 	while ( (eroot = ast_walk_context_extensions(tmp, eroot)) ) {
 		int match = extension_match_core(eroot-exten, exten, action);
 		/* 0 on fail, 1 on match, 2 on earlymatch */
+ast_log(LOG_NOTICE, [%s] match(%s, %s, %x) - %d\n, tmp-name, eroot-exten, exten, action, match);
 
 		if (!match || (eroot-matchcid  !matchcid(eroot-cidmatch, callerid)))
 			continue;	/* keep trying */
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute









Well I am using GSM as my main codec which
seems to be very nice

I would also suggest you looking at the
load of you CPU I know that asterisk is very processor hungry



You can also change some settings in the
zapta and zaptel config.. to reduce echo and interference on the line..











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of sdgesa gaeharth
Sent: 05 October 2006 14:38
To:
asterisk-users@lists.digium.com
Subject: RE: [asterisk-users]
Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute





Below is the text of my
original post. I am not sure what Codec we are using. The Codec
Preferences phone setting shows, in order of preference, G.711u, G.711A,
G.729AB



We are running asterisk-1.2.4 with zaptel-1.2.7 on Fedora Core
4-2.6.14-1.1656_FC4smp. It is installed on a Dell PE 2500 with
2x900 MHz processors and 1 Gb RAM and 1 SCSI Disk. The server has a Digium
TDM400P card which is connected to 4 POTS lines. The server is also
connected to a 100MB switched LAN where we have about 20 Polycom 501 phones
with the latest firmware updates. Nothing else runs on the server except an ftp
daemon which is never used except when a phone reboots.

For about 20% of the calls to the outside world, the voice on the other end of
an outside line is incredibly choppy. Enough to where we have to
hang up and call on a cell phone. It is always the same numbers that are
choppy. The funny thing is, if I press mute while talking on a choppy
call, the choppiness goes away completely.





I have tried: turning off ACPI, turning off APCI, moving the card to
another PCI slot, changing the RX/TX gains. There are no shared IRQs. I have
tested the lines by unplugging them from the asterisk server and plugging them
directly into an analogue phone. Using cat /proc/interrupts; sleep 10 ;
cat /proc/interrupts I see that there are about 1,000 interrupts per
seconds between the card and the CPU.





I do not think it is a network congestion problem as intra-office
communications as well as voicemail retrieval are always perfect. The Voip does
not go over any routers, just a max of 2 switches with a 1GB trunk. This
happens even off-hours when the network isnt being used at all.





There are never more than 2 people on the phone at the same time and it
is definitely not an over-utilized processor.





I have trying to figure
this out for 2 months on and off with no success any help is appreciated.





Thanks

Andrew Shelton
[EMAIL PROTECTED] wrote:



What
codec are you using?











How many phone? What load is the server
under?































From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of sdgesa gaeharth
Sent: 05 October 2006 13:22
To:
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
Extremely choppy sound on some of our POTSnetwork calls; goes away with mute













1)Can anyone tell me how to do this on a Polycom 501?

2)Can you explain why you think this any why it ony happens on some calls?

Thanks

Andres
[EMAIL PROTECTED] wrote:








 For about 20% of the calls to the outside world, the voice on the 
 other end of an outside line is incredibly choppy. Enough to where 
 we have to hang up and call on a cell phone. It is always the same 
 numbers that are choppy. The funny thing is, if I press mute while 
 talking on a choppy call, the choppiness goes away completely.

 

Maybe you have silence suppression enabled on your phones. Try to 
disable it and see if it helps.



 



-- 
Andres
Technical Support
http://www.telesip.net

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users











 











Yahoo! Messenger with Voice. Make
PC-to-Phone Calls to the US
(and 30+ countries) for 2¢/min or less.



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



  







Get your email and more, right on the new
Yahoo.com 






___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] two asterisk and one NBX system

I would research the switch statement and DUNDI

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jose diaz
Sent: 05 October 2006 14:51
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] two asterisk and one NBX system

We have three servers: Two asterisk and one NBX 3COM.
The connection between Asterisk1 and Asterisk2 is with IAX2.
The connection between  Asterisk2 and NBX is with a Digium analog 
TDM400P (2FXO and 2 FXS)

The dial plan Asterisk1: 3XXX
The dial plan Asterisk2: 2XXX
The dial plan NBX: 1XXX

The system work well, but the call from Asterisk1 to NBX fail. I'm using

the IAX2 protocol to call from asterisk1 to asterisk2, i need to 
trasnfer the call to the NBX. How i can to make that?

Regards,

Jose Diaz

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] snom 360: how to make record button working ?

2006-10-05 Thread noro kamen


Hi Joel,

thanks for the answer :-).

Yes this is one (the easiest) way how it can be done (on phone side),
but I am still
looking for asterisk side solution ...
i.e. it should understand info message sent by phone and do some
prescribed action.

Haven't u any clue ?

noro



2006/10/5, Joel Hill [EMAIL PROTECTED]:

Hi Noro,

Depending on what firmware you have this is the way to go.
Go to the Functions keys page, then look for the Record button, Change
the type to DTMF and in number put in *1 which is the default Asterisk
recording function.

Hope this helps

Cheers,

Joel
Asterisk IT
www.asteriskit.com.au


noro kamen wrote:
 Hi,

 I'd like to make record button working on snom 320/360 + asterisk.

 As I learned from wireshark output,  the phone produces SIP info
 message Record: on, while record button pressed.

 Can anybody give me an advice, how to teach asterisk to understand
 that SIP info message and start recording ?

 TIA
 noro
 ___

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] for god's sake somebody help me! ANSWEREDTIME=0 in astcc!!

2006-10-05 Thread Ali

hi everybody,


I have been playing with asterisk and astcc for a while and I have got everything up and running but one thing: ANSWEREDTIME !!

I have tried both Zap and SIP but no difference, my calling card system fails because ANSWEREDTIME is always zero and I think the reason is astcc.agi scripts continues execution and exits after executing AGI = exec(Dial $dialsrt)


I am sure there is something that I have missed out because many people say that they are using astcc as a calling card software.

I have read all the documents on google and wiki about astcc.

Please someone give me a hint, this is making me crazy!


Thanks alot



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] AW: asterisk-users Digest, Vol 27, Issue 23

2006-10-05 Thread Daniel Hikel

 = outbound

;include = stem  ;for hints

 

 

[inbound-from-stem]

include = BLF

include = internal

include = DefExt

include = voicemail

include = outbound

include = BLF_group_pickup

include = feature-cfu

include = feature-cfna

include = feature-cfb

 

[inbound-from-logicall]

include = internal

include = DefExt

include = voicemail

include = outbound

include = BLF_group_pickup

include = feature-cfu

include = feature-cfna

include = feature-cfb

 

;Test section for BLF on Grandstreams for Stem

[BLF_group_pickup]

include =inbound-from-stem

exten = _**.,1,NoOp(${EXTEN})

exten = _**.,2,Pickup(${EXTEN:2})

exten = _**.,3,Hangup

 

[BLF]

include =inbound-from-stem

exten =7000,hint,SIP/7000

exten =7000,1,Dial(SIP/7000,20,r)

exten =7001,hint,SIP/7001

exten =7001,1,Dial(SIP/7001,20,r)

exten =7002,hint,SIP/7002

exten =7002,1,Dial(SIP/7002,20,r)

exten =7003,hint,SIP/7003

exten =7003,1,Dial(SIP/7003,20,r)

exten =7004,hint,SIP/7004

exten =7004,1,Dial(SIP/7004,20,r)

exten =7005,hint,SIP/7005

exten =7005,1,Dial(SIP/7005,20,r)

exten =7006,hint,SIP/7006

exten =7006,1,Dial(SIP/7006,20,r)

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

-- next part --
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20061005/f4499d
b5/attachment-0001.htm

--

Message: 2
Date: Thu, 05 Oct 2006 07:10:00 -0600
From: Steve Murphy [EMAIL PROTECTED]
Subject: [asterisk-users] Re: AEL2 #include madness in Asterisk 1.4 -
Murf?
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii

On Thu, 2006-10-05 at 01:08 -0700, [EMAIL PROTECTED] wrote:
 Asterisk 1.4 beta2.
  
 My top level /etc/asterisk/extensions.ael has the following
 two lines:
  
 #include include/syst/extensions.ael
 #include include/btck/extensions.ael
 
 Here is the console output on Asterisk load.
  
 app_system.so = (Generic System() application)
 [Oct  4 15:48:15] NOTICE[1143]: pbx_ael.c:3798
 pbx_load_module: Starting AEL load process.
 [Oct  4 15:48:15] NOTICE[1143]: pbx_ael.c:3805
 pbx_load_module: AEL load process: calculated config file name
 '/etc/asterisk/extensions.ael'.
 [Oct  4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
 --Read in included
 file /etc/asterisk/include/syst/extensions.ael, 4130 chars
 [Oct  4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
 --Read in included file /etc/asterisk/include/syst/macros.ael,
 1463 chars
 [Oct  4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
 --Read in included
 file /etc/asterisk/include/syst/dundiapps.ael, 758 chars
 [Oct  4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
 --Read in included file /etc/asterisk/include/syst/rdapps.ael,
 275 chars
 [Oct  4 15:48:15] NOTICE[1143]: ael.flex:429 ael_yylex:
 --Read in included
 file /etc/asterisk/include/btck/extensions.ael, 1385 chars
 [Oct  4 15:48:15] NOTICE[1143]: pbx_ael.c:3808
 pbx_load_module: AEL load process: parsed config file name
 '/etc/asterisk/extensions.ael'.
 [Oct  4 15:48:15] ERROR[1143]: pbx_ael.c:1162 check_goto:
 Error: file /etc/asterisk/include/syst/extensions.ael, line
 157-157: goto:  no label remote exists in the current
 extension!
 [Oct  4 15:48:15] ERROR[1143]: pbx_ael.c:1162 check_goto:
 Error: file /etc/asterisk/include/syst/extensions.ael, line
 159-159: goto:  no label local exists in the current
 extension!
 [Oct  4 15:48:15] ERROR[1143]: pbx_ael.c:3821 pbx_load_module:
 Sorry, but 0 syntax errors and 2 semantic errors were
 detected. It doesn't make sense to compile.
 pbx_ael.so = (Asterisk Extension Language Compiler)
  
 Here's the context
 from /etc/asterisk/include/syst/extensions.ael, that contains
 lines 157 that the parser is complaining about:
  
148  context syst_Route {
149
150  _[*0123456789]. = {
151  NoOp(*** Originated call ${CALLERID} -
 ${EXTEN});
152  Set(TMP=${CALLERID(number)});
153  SysLogger(This is a test message);
154  FastAGIConnectGet(CALLERID);
155  ChanIsAvail(SIP/${EXTEN});
156  if (${AVAILCHAN} = ) {
157  goto remote;
158  } else {
159  goto local;
160  }
161  remote:
162  NoOp(REMOTE);
163  Set(PATH=
 ${DUNDILOOKUP(3254103,DUNDIRegistr)});
164  //Set(PATH

Re: [asterisk-users] Where is the PlayDTMF command?

2006-10-05 Thread Moises Silva


Where can I get the source to apply the patch, is it difficult  to apply it?

I dont remember the exact place where I get it, was in
bugs.digium.com, but I dont remember the number/name of the bug. I
made a fix to the patch so it can apply to 1.2.12.1
You can get it here
http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch

And you apply it this way:

# cd asterisk-1.2.12.1
# patch -p1  ../play_dtmf-1.2.12.1.patch

compile and install again, and you can test if you have it from the asterisk CLI

CLI show manager command PlayDTMF


Regards
--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] New tutorial - peering two * servers using IAX

2006-10-05 Thread lenz



I was thinking about that, but there does not seem to be so much interest  
in DUNDi at the moment - most people I see are still trying to understand  
what a context is and why they cannot use the transfer button in a  
queue. :)

l.


In data Wed, 04 Oct 2006 20:01:20 +0200, Douglas Garstang  
[EMAIL PROTECTED] ha scritto:


How about preparing a step by step guide to DUNDi? Good luck with that  
though because base DUNDi docs are rarer than periodic element #114 in  
the known universe.


Doug.



--
Home of QueueMetrics - http://queuemetrics.loway.it

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Re: Bandwidth requirements

2006-10-05 Thread Matthew Crocker



True but you need to look at the actual speed of a DSL line.  a 1.5m/ 
384k DSL line is actually 1.7m/467k  so a chunk of that ATM cell tax  
is already factored into the speed you buy from your DSL provider.   
If you get an integrated DSL modem/router like a Zoom X5 you can see  
the actual speed of the link.




On Oct 5, 2006, at 9:31 AM, Benny Amorsen wrote:


rJ == raphael Jacquot [EMAIL PROTECTED] writes:


rJ ATM cell tax is actually 10% as there's 5 header bytes for each 53
rJ bytes cell,

For VoIP the cell tax is much larger. In the example, each RTP packet
contains 20 useful bytes and 40 bytes IP overhead. 60 bytes doesn't
fit in one cell, so you end up with 106 bytes at the ATM layer to
transport 20 bytes of G.729. The ATM-caused overhead is thus 46 bytes
per voice packet, thereby making the needed bandwidth 77% larger.

All in all VoIP over ADSL adds 430% overhead, when using G.729 and
20ms packets. Lovely, isn't it?


/Benny


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


--
Matthew S. Crocker
Vice President
Crocker Communications, Inc.
Internet Division
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] two asterisk and one NBX system

2006-10-05 Thread Moises Silva


Could you describe better the call path?

are you doing this?

Asterisk 2  IAX2 --- Asterisk 1 - FXO/FXS - NBX

Or something else? If so, it would be nice to post relevant parts of
the Asterisk CLI with all the log levels activated in logger.conf

Regards

On 10/5/06, jose diaz [EMAIL PROTECTED] wrote:

We have three servers: Two asterisk and one NBX 3COM.
The connection between Asterisk1 and Asterisk2 is with IAX2.
The connection between  Asterisk2 and NBX is with a Digium analog
TDM400P (2FXO and 2 FXS)

The dial plan Asterisk1: 3XXX
The dial plan Asterisk2: 2XXX
The dial plan NBX: 1XXX

The system work well, but the call from Asterisk1 to NBX fail. I'm using
the IAX2 protocol to call from asterisk1 to asterisk2, i need to
trasnfer the call to the NBX. How i can to make that?

Regards,

Jose Diaz

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users]FIX FOUND: Zaptel problems

2006-10-05 Thread Shea, Matt

Brad's suggestion below fixed my problem.  I'm using a Digium TE410P
card, which is one of the ones he mentions and uses the wct4xxp driver.

Matt
313-667-0970
[EMAIL PROTECTED]
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Watkins,
Bradley
Sent: Wednesday, October 04, 2006 4:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Zaptel problems

You didn't say, but my guess is you are using either a 4-port or 2-port
Digium card, right?

What do the contents of /etc/modprobe.d/zaptel look like?

You will probably find that there isn't an entry like:

install wct4xxp /sbin/modprobe --ignore-install wct4xxp $CMDLINE_OPTS 
/sbin/ztcfg

I put in a bug for this already, though in the report it's for FC5:
http://bugs.digium.com/view.php?id=8071


Of course, tell me if this doesn't apply to your situation.


- Brad

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Shea, Matt
 Sent: Wednesday, October 04, 2006 2:35 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] Zaptel problems
 
 Hmmm,
 
 It appears ztcfg is not being run.  Any ideas why?
 
 Matt
 313-667-0970
 [EMAIL PROTECTED]
  
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Bernardo Vieira
 Sent: Wednesday, October 04, 2006 12:40 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Zaptel problems
 
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 Is ztcfg running at boot after the zaptel modules have been loaded?
 What's the output of ztcfg?
 
 
 Shea, Matt wrote:
  I'm running Asterisk/Zaptel on a Fedora Core 4 machine.  
 The software 
  runs ok with one exception.  Zaptel appears to load OK on 
 bootup, but 
  when you check it on login, zttool still shows red/nop alarms on the
 T1
  lines.  I have to manually start it again for the alarms to 
 disappear 
  and the T1 lines to function properly.  I've updated the drivers to
  1.2.9.1 and double checked my configuration files to no 
 effect.  Any 
  suggestions will be much appreciated.
  
   
  
  Matt
  
  
  
  
 
 --
 --
  
  ___
  --Bandwidth and Colocation provided by Easynews.com --
  
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 - --
 What most profoundly divides two men is a different sense 
 and degree of cleanliness. What help is all honesty and 
 mutual utility, what help is all the good will for each 
 other: in the end the fact remains-they can't stand each 
 other?s smell!
 
 - - Nietzsche
 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.4.1 (MingW32)
 Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
 
 iD8DBQFFI+PS2QVs8jsa1mQRAnRKAKCYVjy0QKxm67RMyc3/kMw+i+sDkgCePu1U
 zeKUkrOK4rPfnl4+HvnpEK8=
 =pxJ+
 -END PGP SIGNATURE-
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 ___
 --Bandwidth and Colocation provided by Easynews.com --
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
The contents of this e-mail are intended for the named addressee only.
It contains information that may be confidential. Unless you are the
named addressee or an authorized designee, you may not copy or use it,
or disclose it to anyone else. If you received it in error please notify
us immediately and then destroy it. 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Detecting Busy AGI Extensions?

2006-10-05 Thread Matthew Rubenstein

I'm initiating a bunch of calls from Perl script that sends
callfiles into the outgoung queue. A callfile dials a phone# thru a SIP
gateway to the PSTN, then connects it to an extension which is mapped to
an AGI script (also Perl, but a separate file) that plays a recording
(and listens for some IVR). The initiating script loops to start the
next call faster than the AGI extension finishes the call/IVR. So I
expect that the initiating script will get lots of busy messages from a
single extension, and therefore fail to call until the extension is free
again, or at least block with a full outgoing spool, though I have lots
of extra SIP outgoing capacity. I want the initiating script to start
several parallel calls.

So I could make the initiating script specify a loop of predefined
extensions, each pointing to a copy of the AGI that manages the actual
call content once connected. But I'm not sure how the initiating script
can get the available status of an extension before specifying the
extension in the callfile or skipping to the next possible  extension.
The Asterisk::AGI module invoked in the initiating script doesn't have
the AGI context of the script to which it's passed, which makes a
catch-22. The manager interface connected to the initiating script
doesn't seem to support the ExtensionState command properly, which just
returns -1 (invalid) unless the extension has a hint, in which case it
always returns 4 (unavailable), regardless of the state. The manager
interface does catch NewExtension events when the AGI on the extension
is actually running, but not when it stops, so the initiating script can
register that the AGI script is busy, but not when it becomes available
again. And I don't see any way for the initiating script to act like a
dialplan to jump to a label on extension busy, or a way to
programmatically refer to the next extension in a context.

I think I'd rather use queues for flexibility, scalability, etc.
However, it looks like creating a queue requires direction indirectly
through Agents, virtual connectors into which each AGI script handling
the actual call content would have to login. I don't know how to make an
AGI script login as an Agent, or how to specify a queue that connects
live outgoing channels to a pool of waiting extensions.

Thanks for your advice.
-- 

(C) Matthew Rubenstein

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] MODEM (data) througt asterisk ?

2006-10-05 Thread Time Bandit


Is it possible to connect a modem to a remote service through asterisk ?
Basicly to ilustrate : Accounting department need to connect with analog
modem to their bank to order some wire transfert.

Modem - Chanel Bank FXS - Asterisk - TDM2400 FXO - Modem in
remote site.


If you get it working you're lucky. Digium's official statement on the
TDM2400 is that card as been designed for voice calls, we don't
support data calls

You would have better luck with a Sangoma A200

hth
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk integration

2006-10-05 Thread Brian Becker


I am sure this is slightly offtopic for this list but it seems there
are a number of smart people here.  My company is experimenting with
Asterisk with great success.  We are looking to integrate it into our
current phone system to run the phones out of a new suite of offices
(same building hard wired).  I  have almost no knowledge of the
current phone system other then what I get told from our phone
contractor.  We currently have a Samsung iDCS 500 digital phone system
(Apparently release 1, whatever that meens) running Telekol Minitel
3.91B (if that even matters?).

In its current setup I setup a zaptel trunk to an analog circuit witch
actually works great.  Running a number of offices off of a single
analog line has its obvious downsides.  Obviously I could hook
multiple analog lines up to the system and have multiple zaptel trunks
but this seems a bit micky mouse (not to mention just how many lines I
would need).  I would rather have a SIP trunk (or some other trunk by
all means) to be handed off from the current system to go into
asterisk.

Now that you have the background I am hoping someone here has the
knowledge to either confirm or deny what the contractor is telling me.
I am being told that I have to upgrade to release 2 in order to
integrate asterisk...simple question, is this true?

My hunch is i could probably buy a module for release 1 that would
give me this capability.

Brian
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Video Conference

2006-10-05 Thread Dean Collins









Noah,

Just for clarification this is no longer a bounty for video
conferencing.

I ended up purchasing an off the shelf system.



I might however restart it with a lower commitment for the
benefit of the community if someone showed an interest.



Regards,



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



-Original
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent: Thursday, 5 October 2006 10:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Video Conference



Hi Bilal -



 We need to apply Video conference, can asterisk

 support this?



No. Asterisk supports video calls between two end points, but not

video conferences with three or more participants.



There is a bounty for someone to add this feature, but nobody has

successfully implemented it yet.





 What I need for that?



Something else. You can get video conferencing software, or if you

have the right hardware you can use it. There are many hardware video

conferencing units available from Polycom, Tandberg, Sony, etc.





- Noah

___

--Bandwidth and Colocation provided by Easynews.com --



asterisk-users mailing list

To UNSUBSCRIBE or update options visit:

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






___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TNT Max Password reset

On Thu, Oct 05, 2006 at 10:28:21AM +0100, Steve Kennedy wrote:
 On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote:
  On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
  Anyone have happen know how to reset the password on a TNT Max? Thanks.
  Does your asking here suggest that the the MAX's can do, say, voice
  gateway service?  Protocols?  Codecs?
 
 Ascent TNT's with the right software and hardware can do SIP, E1
 termination/origination, and all sorts of codecs.

To clarify, since I have an older Max: That's only the TNTs?

I'm pre-speccing a project I've mentioned a couple times before; 390
port hospitality PBX, and I'm always looking for good FXS gateways...

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth  AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] TNT Max Password reset

2006-10-05 Thread Natambu Obleton

Yes, but they require a software upgrade. I am using them for plain ole
dialup, but because of that feature set I figured someone on this list would
know how. I find a lot of people online saying to serial into them and reset
them, but no detail on when to send break or what dip switches to set or
anything. :(


Natambu Obleton
Network Engineer
FastTrack Communications
[EMAIL PROTECTED]
(970) 247-3366 office
(970) 247-2426 fax
 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay R.
Ashworth
Sent: Wednesday, October 04, 2006 4:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TNT Max Password reset

On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
Anyone have happen know how to reset the password on a TNT Max? Thanks.

Does your asking here suggest that the the MAX's can do, say, voice
gateway service?  Protocols?  Codecs?

Cheers,
-- jra
-- 
Jay R. Ashworth
[EMAIL PROTECTED]
Designer  Baylink RFC
2100
Ashworth  AssociatesThe Things I Think'87
e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647
1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re:UPDATE: Zaptel problems

2006-10-05 Thread mavince

I had a similar problem... every timeI rebooted, ztcfgdidn't run. I found that by running "make config" from the Zaptelsource directory this problem was corrected. I had skipped this step in my original setup.

My /etc/rc.local has two entries:
modprobewct4xxpsafe_asterisk
The current Asterisk Business Edition has a script that makes the rc.local entries unneeded... but I found that I still had to manually run the "make config" command to guaranteethat Asterisk would automatically recover from a reboot.

Mark

Mark A. Vince
[EMAIL PROTECTED]

 Message: 15 Date: Wed, 4 Oct 2006 16:26:27 -0400 From: "Shea, Matt" <[EMAIL PROTECTED]> Subject: RE: [asterisk-users] UPDATE: Zaptelproblems To: "Asterisk Users Mailing List - Non-Commercial Discussion"  Message-ID: <[EMAIL PROTECTED]>  Content-Type: text/plain; charset="US-ASCII"  I found a workaround, inspired by Colin'ssuggestion to move the  startuptothe rc.local file. It turned that his exact  suggestion didn't work in my situation. I subsequently discovered, though, that after Zaptel and Asterisk started in the boot sequence in the usual way, all  I had to do for the Zapteldrivers to fully kick in was re-run ztcfg.  So, as of now, the rc scripts are all in place in the usual way with the  followingline added to /etc/rc.local:  runuser -l -c ztcfg -s /bin/bash root  Now it's starting properly, but I'm not really all that happy  that I have to put a jury-rigged workaround in place. If anyone has  the real solution, I'd certainly like to hear it.  Thanks for all the suggestions so far.  Matt   -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of  Shea, Matt Sent: Wednesday, October 04, 2006 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Zaptel problems  Hmmm,  It appears ztcfg is not being run. Any ideas why?  Matt   -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernardo Vieira Sent: Wednesday, October 04, 2006 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zaptel problems  -BEGIN PGP SIGNED MESSAGE- Hash: SHA1  Is ztcfg running at boot after the zaptel modules have been loaded? What's the output of ztcfg?   Shea, Matt wrote:  I'm running Asterisk/Zaptel on a Fedora Core 4 machine. The  software runs ok with one exception. Zaptel appears to load OK  on bootup, but  when you check it on login, zttool still shows red/nop alarms  on the T1  lines. I have to manually start it again for the alarms to  disappear and the T1 lines to function properly. I've updated  the drivers to  1.2.9.1 and double checked my configuration files to no  effect. Any  suggestions will be much appreciated.Matt  - ---___
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TNT Max Password reset

On Thu, Oct 05, 2006 at 08:43:26AM -0500, James wrote:
 I have five MAX TNT's runnig with SIP and g.729.
 They will do E1's, T1's, T3's.

Got it.  I realized after my other reply that no, these wouldn't be FXS
side stuff, they'd be FXO.

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth  AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk manager

2006-10-05 Thread Michel Vaillancourt

Voipers Portugal wrote:
 I know that, that is why I asked if there was any tool that would do
 something like that, but by acessing the Manager API?
 
 Anyone?
 

Our interface uses ARI and MySQL.  There is no reason that you could 
not manage a secure box with the interface app, with MySQL replication of the 
master table out to a slave-only table on the exposed machine.  That way, 
there is really nothing on the exposed machine to compromise.  Go the extra 
step and SSL the replication channel and the only thing you'd have to have in 
the clear would be the SIP connections themselves.

-- 
--Michel Vaillancourt
Senior Telephony Engineer
Neoxo Inc  (www.neoxo.com)
+1 514 395 1106 ext 117
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] no callerid from PSTN using TDM2400P

On Wed, Oct 04, 2006 at 10:04:08PM -0700, Crazy Boy wrote:
This is Chandra from India. You are from which country? I am asking this
because the basic Asterisk setup doesn't recognize callerid in India. I
tried to solve this in many ways. But, no use. I think we have to do some
modifications in source code. Thank you.

You may need to do some mods in the *drivers*; I don't know how
Asterisk and the Zap drivers handle DTMF while still on-hook... which
is what

http://www.ainslie.org.uk/callerid/cli_faq.htm

suggests you may have.

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth  AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] cisco 2600

2006-10-05 Thread Gary Richardson

I'm using a 2811. Just about everything works fine, though we are experiencing a problem with redirecting calls.On 8/5/06, FaberK 
[EMAIL PROTECTED] wrote:Hi,does anybody used cisco 2600 as * gateway with E1?
Thanks-- .:FaberK:.

___--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] snom 360: how to make record button working ?

On Thu, Oct 05, 2006 at 07:17:47AM -0400, sip wrote:
 That varies from location to location, really. In Georgia, for instance, only
 ONE party need know the recording is taking place (calling or receiving)
 without a warrant. In some countries, neither party need know, etc, etc. 

This page: 

http://www.pimall.com/nais/n.recordlaw.html

purports to list the states that require all party consent.  It is from
a private investigation site, and was the number one google hit, so it
may be reliable.  This is not legal advice; IANAL.  If my advice breaks
something, you get to keep both pieces, unless you paid me for it.

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth  AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call Center requirements

2006-10-05 Thread BJ Weschke


On 10/5/06, Todd Houle Asterisk [EMAIL PROTECTED] wrote:

Hi Guys-
While I played a little with Asterisk a year or so ago, I'm getting ready to
roll out a project now that I think is perfect for it.  My friend with with
a commercial solution he has been very unhappy with and is thinking of
replacing it with Asterisk.  Below are his requirements.  Anything here jump
out as a problem? I'm thinking of purchasing a few Digiium card - not sure
which we need yet...   and finding a box to run it on. The only part I'm not
sure is how to address is having the client record auto-appear on screen
when the call comes in.  I did see plug ins for recording the calls...Is
asterisk the best solution for this?
 thanks
Todd

Begin forwarded message:

From: A. Pathuri [EMAIL PROTECTED]
Date: October 2, 2006 2:51:32 AM EDT
To: Todd Houle [EMAIL PROTECTED]
Subject: Call Center requirements


Todd,

Here is the brief doc you requested.

The process that we need is pretty simple...


We get a bunch of DID (Direct Inward Dialing) numbers from SBC.
As we get a client, we assign them a DID #.
They forward their existing phones to their DID number when their lines
are busy or after hours.
The DID # is programmed into the telephony system so we can program the
caller ID, and enter the appropriate script to pop up when that number
comes through.

When a call comes in, I would like to have all calls automatically
recorded without any of the call agents having to press a record button
for each call.

We also current have conference call functionality where we can connect
one caller to another caller (used when the ER needs to speak to a
doctor).

Ideally also, I would like the recorded calls to sort by client and
store in the appropriate clients folder, which then can be
automatically zipped and sent via email to the clients inbox at any
desired interval.


We are also developing a web-based app where the details of each call
can be entered ( a sort of call log) so the clients can also log into a
web interface and see the details of each call (currently, most clients
get their call logs via fax in the am and at midnight).

It would be great if somehow, the caller ID on the server/astericks can
automatically pull up the appropriate clients profile from our web app,
so the details can be entered into the correct profile.  Otherwise, for
each call that comes in, the call agent has to pull up the clients
profile while the caller is on the phone, before s/he can take down the
details of the call.

This is really rough, but I hope it gives the basic idea.  We can
discuss in further detail once you take a look at this.


Ofcourse, as well it would be great to be able to setup a co-location in
India utilizing the same infrastructure.



There are a number of ways to do this, but given the application it
appears to be (medical), and additional requirement not mentioned here
(and quite possibly the most important) is HIPPA compliance with
regard to security of who has access to what information.


--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Dial out trhough a FXS channel on a TDM card

2006-10-05 Thread Robson Ribeiro









Hello.



I have a TDM 2400P and havent figured out how to
attach a phone to one of the FXS channels in the bank and dial out. To dial in
the analog phone is easy, all I had to do was to insert a line in the
extensions.conf saying exten = 430,1,Dial(Zap/17,20,t). But I cant
figure out how to have a line signal on the same phone to dial out. Anybody can
help me please?



Thanks,



Robson








___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute

2006-10-05 Thread Noah Miller


Well I am using GSM as my main codec which seems to be  very nice…


Polycom phones do not support GSM (GSM would not be necessary here
anyway, since all these phones are on a local LAN, so bandwidth does
not need to be conserved).



You can also change some settings in the zapta and zaptel
config.. to reduce
echo and interference on the line..


This is the most important thing here - what does your zapata.conf look like?

Other things:
1. Update asterisk to a newer version.  There have been MANY bugs that
have been fixed since 1.2.4.
2. Update zaptel to a newer version.  Not much has changed for the TDM
cards since 1.2.7, but you should update anyway.

- Noah
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Bandwidth requirements

On Thu, Oct 05, 2006 at 09:44:56AM +0100, Brian Candler wrote:
 A 20ms packet duration means that 20ms of audio is stuffed into one IP
 packet. Since each packet carries 1/50th of a second of audio, that means
 you're generating 50 packets per second for each channel.
 
 With g729 your audio is 8000 bits per second.
 
 The overhead on each packet is 20 bytes (IP) + 8 bytes (UDP) + 12 bytes
 (RTP) = 40 bytes or 320 bits.
 
 So your bandwidth requirement per channel is:
 - 8000 bits per second for payload
 - 320x50 = 16000 bits per second for overhead
 making a total of 24000 bits per second.
 
 20 simultaneous calls is therefore 480,000 bits per second.

A reminder: much equipment, particularly low end/consumer equipment,
chokes *much* faster on high PPS than it does on high BPS.

While short packets are good for latency, they do impose stricter
engineering evaluation requirements on the other links in your chains.

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth  AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] two asterisk and one NBX system

2006-10-05 Thread jose diaz


the call path is:

Asterisk1-IAX2--Asterisk2---FXO/FXSNBX

I need to route call from Asterisk1 to NBX.

Regards,

Jose Diaz


Moises Silva wrote:

Could you describe better the call path?

are you doing this?

Asterisk 2  IAX2 --- Asterisk 1 - FXO/FXS 
- NBX


Or something else? If so, it would be nice to post relevant parts of
the Asterisk CLI with all the log levels activated in logger.conf

Regards

On 10/5/06, jose diaz [EMAIL PROTECTED] wrote:

We have three servers: Two asterisk and one NBX 3COM.
The connection between Asterisk1 and Asterisk2 is with IAX2.
The connection between  Asterisk2 and NBX is with a Digium analog
TDM400P (2FXO and 2 FXS)

The dial plan Asterisk1: 3XXX
The dial plan Asterisk2: 2XXX
The dial plan NBX: 1XXX

The system work well, but the call from Asterisk1 to NBX fail. I'm using
the IAX2 protocol to call from asterisk1 to asterisk2, i need to
trasnfer the call to the NBX. How i can to make that?

Regards,

Jose Diaz

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users






___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] TNT Max Password reset

2006-10-05 Thread Natambu Obleton

maxrouter.com - You are on the page : Voice over IP
http://lnk4.us/VQ02


MAX Routers from Ascend / Lucent for VOIP ?

On these pages we are presenting the use and knowledge of used Ascend/Lucent
and used Compaq server products in the foreground. However, there are more
vendors with excellent equipment for gateways and servers.

As we know, the Ascend MAX 3000 and MAX 6000 and the MAX TNT are able to
handle VOIP connections. These units are able to run multi channel (multi
port) DSP based cards for coding and compression of natural speech.

The MAX 3000 ist rarely sold on the market but the MAX 6000 and MAX TNT are
out in large quantities. So let us focus on these units.

MAX 1800, MAX 2000 and MAX 4000 are not expandable to run VOIP DSP boards.


Natambu Obleton
Network Engineer
FastTrack Communications
[EMAIL PROTECTED]
(970) 247-3366 office
(970) 247-2426 fax
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay R.
Ashworth
Sent: Thursday, October 05, 2006 8:49 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] TNT Max Password reset

On Thu, Oct 05, 2006 at 10:28:21AM +0100, Steve Kennedy wrote:
 On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote:
  On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
  Anyone have happen know how to reset the password on a TNT Max?
Thanks.
  Does your asking here suggest that the the MAX's can do, say, voice
  gateway service?  Protocols?  Codecs?
 
 Ascent TNT's with the right software and hardware can do SIP, E1
 termination/origination, and all sorts of codecs.

To clarify, since I have an older Max: That's only the TNTs?

I'm pre-speccing a project I've mentioned a couple times before; 390
port hospitality PBX, and I'm always looking for good FXS gateways...

Cheers,
-- jra
-- 
Jay R. Ashworth
[EMAIL PROTECTED]
Designer  Baylink RFC
2100
Ashworth  AssociatesThe Things I Think'87
e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647
1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Wouldn't Tri-tone detection in Dial() be cool?

2006-10-05 Thread Richard Lyman


Jay R. Ashworth wrote:

*snipped



The ability to detect precise SIT tones on placed calls would be
*really* good.

  
actually it is damn near impossible. 
in a perfect world, if all the switch providers where adhering to ITU 
spec on SIT's,
then it would be possible. they sad part is (at least out here in 
pacbell/sbc/now att
land) west coast/US), they don't give a *blank* about inband SIT info.  
(note: inband)


i actually went through the whole process of tracking down which 
switches had down
right bad recordings back in 2004, and was able to get them to re-record 
(or fix) the

SIT's.

after the first round of fixes, i provided another batch of like 150+, 
and at that point

they would not fix anymore.

so, goodluck with 'precise SIT detection' (inband)

the below link is data on the tests and such, even a couple screenshots 
showing the

difference in amplitude from a mere whisper, to near eardrum damage levels.

http://www.dynx.net/SBC/

(this is why conversion to PRI (for outofband info) is the path we took)



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] snom 360: how to make record button working ?

2006-10-05 Thread noro kamen


Hi Joe,

this is the link I was looking for - I ggled a lot, but didn't find it.

Thanke you !
noro


2006/10/5, Joe Pukepail [EMAIL PROTECTED]:

There was a patch to get this working, looks like it has been abandoned,
though.  Should give you a starting point to get it working, or perhaps a
bounty would get someone interested in getting it usable and committed.

http://bugs.digium.com/view.php?id=4845



On 10/4/06, Joel Hill [EMAIL PROTECTED] wrote:
 Hi Noro,

 Depending on what firmware you have this is the way to go.
 Go to the Functions keys page, then look for the Record button, Change
 the type to DTMF and in number put in *1 which is the default Asterisk
 recording function.

 Hope this helps

 Cheers,

 Joel
 Asterisk IT
 www.asteriskit.com.au


 noro kamen wrote:
  Hi,
 
  I'd like to make record button working on snom 320/360 + asterisk.
 
  As I learned from wireshark output,  the phone produces SIP info
  message Record: on, while record button pressed.
 
  Can anybody give me an advice, how to teach asterisk to understand
  that SIP info message and start recording ?
 
  TIA
  noro
  ___
  --Bandwidth and Colocation provided by Easynews.com --
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 
http://lists.digium.com/mailman/listinfo/asterisk-users
 ___
 --Bandwidth and Colocation provided by Easynews.com --

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:

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



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

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




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] sdsl

2006-10-05 Thread adebayo omo-dare

I was just thinking of a recent discussion on this list on SDSL and realised that unlike in Europe where SDSL lines are deployed in accordance with the G.shdsl standard, in North America it is/ maynot (be)the same. As such the above difference would tend to obscure an understanding of points raised in relation to the deployement of * within this environment.Would anybody be able to tell me howthe G.shdsl standard is currently distributed in NA, and indeed if there are deep cost differences between G.shdslbased solutionsand SDSL. Also if you can, perhaps,give any information interms of regional coverage it would be greatly appreciated.Best regards  Bayo 
		 
Now you can scan emails quickly with a reading pane. Get the new Yahoo! Mail.___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to make RTP does not go thru asterisk server

2006-10-05 Thread Mojo with Horan Company, LLC


correcting an error cause by my own ambiguity:

Mojo with Horan  Company, LLC wrote:
clarifying that you CANNOT put t or T in there if you want 
canreinvite=no to have no effect.
you cannot put t or T in there if you want canreinvite=no to have ANY 
effect.  If you want the stream to skip asterisk, and first you've told 
it not to allow reinvites with this canreinvite option, then you have to 
make sure asterisk isn't also being TOLD to listen in on the stream for 
transfer requests (t and T)


Moj



Anuj Jain wrote:

Hi All
I am using trixbox asterisk 1.2
I have enabled canreinvite=yes and no  tT in the dialplan as it has 
been described in the various forums.

Still the voice call goes thru the asterisk server.
How can i really make the call between 2 grandstream devices( i am using 
HT 488, HT286 and SIP extensions) after the initial handshake.


Thanks  Regards





___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


!DSPAM:500,4524455a101385315134984!




--
Mojo [EMAIL PROTECTED]
Office Manager, Horan  Company, LLC
(907) 747- x112
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CDR problem with call transfer

2006-10-05 Thread Mojo with Horan Company, LLC


I could be wrong, but I think the ForkCDR Application might help you:

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

Moj

Kamran Ahmad wrote:

Hi

i am using call transfer feature between three
parties.

dial(sip/${EXTEN}||t)

it is working perfectly but the problem is that cdr is
incorrect.

here is the call senrio

A-B (A calls B, A and B connected)
B-C (B transfer call to C)
A-C (C got ringing, B Hangup, A and C connected)

in cdr there is only one record of A to C

any idea why? is it a Bug?




__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
___

--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

!DSPAM:500,45249992125351533111866!



--
Mojo [EMAIL PROTECTED]
Office Manager, Horan  Company, LLC
(907) 747- x112
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] fonality acquires trixbox ([EMAIL PROTECTED]) ?

2006-10-05 Thread shadowym

 
This is kind of strange.  First of all, I didn't know Trixbox was an
official product or company so how can you aquire something that does not
exist?  Secondly, it's all just open source and free as in free beer so what
is the REAL motivation?  

This seems like a very strange fit. I suspect Fonality just want's them to
go away since it would seem to me they fill similar needs except one charges
for the product and support and the other is free.

-Original Message-
From: Zoa [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, October 04, 2006 11:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] fonality acquires trixbox ([EMAIL PROTECTED]) ?


Looks like phonality has bought trixbox. (I suppose they failed to buy
digium :)

http://news.asteriskguru.com/10/773/2006/10/5/Fonality_Aquires_trixbox_(form
[EMAIL PROTECTED])

Earlier on they found venture capitalist:

http://www.fonality.com/press/20060109.htm





___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Getting Asterisk to work with GoogleTalk

2006-10-05 Thread Alvin Austin


Hello all,

We're trying to get the Asterisk to GoogleTalk functionality working, 
using the latest asterisk svn code (we've also tried with 1.4beta2).  
SVN Asterisk's make update displays:

  Updated to revision 59.
  Updated to revision 44477.

We've tried to follow the recipe (without success) in:
http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk

When Asterisk starts up, the WindowsXP GoogleTalk user (xyz456) sees the 
asterisk server (ast123) appearance.  When it tries to call the asterisk 
server, it hears ringing, but Asterisk does not answer (there is no 
indication in the CLI that it has received a call, except for the 
messages below).


Asterisk (run as:  asterisk -cfvv) shows the following messages 
several times:


JABBER: googletalk INCOMING: iq to=[EMAIL PROTECTED]/asterisk709EC6B7 
from=[EMAIL PROTECTED]/gmail.F1D1B5C9 id=c type=result

query xmlns=http://jabber.org/protocol/disco#info;
identity category=client type=pc/
feature var=http://jabber.org/protocol/disco#info/
/query/iq
   -- JABBER: I Dont have an IQ!!!

JABBER: googletalk INCOMING: presence 
from=[EMAIL PROTECTED]/gmail.F1D1B5C9

to=[EMAIL PROTECTED]showaway/showpriority0/priority
caps:c node=http://mail.google.com/xmpp/client/caps; ver=1.1 
xmlns:caps=http://jabber.org/protocol/caps/

status/x xmlns=vcard-temp:x:updatephoto//x/presence
   -- JABBER: I am available ^_* 13
   -- JABBER: type is away
   -- JABBER: I Do know how to handle presence!!

Would anyone shed some light on what we're missing here, please?

Here are the relevant configuration file pieces...

(1) sip.conf

[general]
context=from-gtalk
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
dtmfmode=rfc2833
relaxdtmf=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
maxexpirey=30
defaultexpirey=180
canreinvite=yes
nat=0
UserAgent=Asterisk
echocancel=yes
echocancelwhenbridge=yes


(2) gtalk.conf (this file is not present. Should it be??)


(3) jabber.conf
---
[general]
;debug=yes
;autoprune=yes
;autoregister=yes

[googletalk]
type=client
serverhost=talk.google.com
[EMAIL PROTECTED]
secret=gtpass
port=5222
;port=5223
usetls=yes
usesasl=yes
[EMAIL PROTECTED]
statusmessage=Voice Calls Only
timeout=100

(4) jingle.conf
---
[general]
context=from-gtalk
;context=default
allowguest=yes

[guest]
disallow=all
allow=ulaw
context=from-gtalk
;context=guest

[google]
[EMAIL PROTECTED]
disallow=all
allow=ulaw
context=from-gtalk
connection=asterisk


(5) extensions.conf (partial):
--
;incoming from GoogleTalk
[from-gtalk]
exten = s,1,NoOP(Incoming call from GoogleTalk to [EMAIL PROTECTED])
exten = s,n,Answer()
exten = s,n,Playback(thanks-for-calling)
exten = s,n,Dial(SIP/101,60,t)
exten = s,n,Hangup

;outgoing to GoogleTalk
[to-gtalk]
exten = 190,1,NoOp(Calling GoogleTalk user [EMAIL PROTECTED])
exten = 190,n,Dial(Jingle/googletalk/[EMAIL PROTECTED])


(note that [EMAIL PROTECTED] and [EMAIL PROTECTED] are fictitious names 
for debugging only)



- If you have this working, please share your sanitized configuration files.
- Can you explain the messages JABBER: I Dont have an IQ!!! and 
JABBER: I Do know how to handle presence!! and what's required to 
correct the problems.


Thanks much,
Alvin

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TNT Max Password reset

2006-10-05 Thread Ejay Hire

The procedure you want is called the nimdy/nindy procedure, and the 
authoratative source is the Book of nindy.  If I remember right, you 
can find it and the current software releases on ftp://ftp.ascend.com


I used these units at an ISP for a long time, and they are okay.  If 
they are installed remotely, you should check the fans once or twice a 
year.  they seem to fail more frequently than normal equipment.  If all 
three of the fans fail it will let the smoke out.  Also, they didn't 
take it well when the blaster worm hit, and Lucent did not distribute 
the fix to anyone without an active support contract.  (a $8,000 support 
contract on a box you can get from eBay for  500!)  They do have enough 
community support that we figured it out, an obscure setting that lets 
you turn off the equivalent of route caching.


-Ejay

Natambu Obleton wrote:

Yes, but they require a software upgrade. I am using them for plain ole
dialup, but because of that feature set I figured someone on this list would
know how. I find a lot of people online saying to serial into them and reset
them, but no detail on when to send break or what dip switches to set or
anything. :(


Natambu Obleton
Network Engineer
FastTrack Communications
[EMAIL PROTECTED]
(970) 247-3366 office
(970) 247-2426 fax
 
-Original Message-

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay R.
Ashworth
Sent: Wednesday, October 04, 2006 4:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TNT Max Password reset

On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
  

   Anyone have happen know how to reset the password on a TNT Max? Thanks.



Does your asking here suggest that the the MAX's can do, say, voice
gateway service?  Protocols?  Codecs?

Cheers,
-- jra
  


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] we are having trouble detecting the # for making a transfer from an E1, usually under some load, please help

2006-10-05 Thread MF


Hi,  this is the setting:

a call comes in vi a first span  g0,it get's hairpined to a second 
span g1,   fine,
on that second span we are accesing a legacy IVR,at some stage  we 
need to get the call and transfer it  (hairpin) to another number via 
the same first span g0,  for that we dial a # and a number,  so the 
system tries to dial that number on the first span,
so in the end, the call  comes in via one slot of the g0  and goes out 
via another slot of the same g0, leaving unused the g1 for the IVR.


this works fine with one or two calls,  but when 4, 5 or 6 comes in then 
it seems as though * won't recognize the # tone we are dialing from the IVR.


does anyone have any idea,we're quite urged here and will appreciate 
any kind of advise.


regards

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TNT Max Password reset

2006-10-05 Thread James


MAX TNT's

Simply upgrading the firmware will not allow all features.
You must also have the HASH CODES to enable the features.
They are sold by Lucent.
Innovative hackers with prom reader/writers have been able to work on some 
of this.
However, unless you own a warehouse full, then it's cheaper to buy an 
upgraded, hashed controller.

You can swap out your old controller for a discount.
And, you will need DSP cards instead of the plain old modem cards.
The TNT and APX line will do SIP with several codec options 
729,711,726,723,gsm.
You should also have the ethernet card with a dongle this has the greatest 
processing for VOIP.


A minimum configuration of hardware would be:

TNT HI POWER CHASSIS
TNT - SP - SC CONTROLLER ALL FEATURES
32MB TAOS 11.0
32MB DRAM
TNT SL - E 100 V -C W/DONGLE
TNT SL CT1 8 PORT T1 CARD
ATX 8 - 96 DSP MODEM CARD
2 TNT - H - AC HI POWER
2 TNT - AC - PWR POWER CORDS
This would give you 8 T1's and 96 concurrent calls, add another 96 port DSP 
for more.


about - $3,000

With this you can do: SIP, ISDN, FGD, FGA, FGB, DTMF, MF, dialup internet, 
PPP, H.323 and send your SIP to Asterisk or SER.


If you have existing modem cards you can direct dialup internet calls to 
them and save  your DSP's for VOIP.

The TNT has 12 fans, is rack mount - heavy duty.
110v, 220v, 48v power supplies are available.

Other MAX equipment -
6000 and 4000 series with multivoice turned on (again hash codes) will do 
H.323 and modem calls only.


James Taylor
1-903-793-1956


- Original Message - 
From: Natambu Obleton [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com

Sent: Thursday, October 05, 2006 9:51 AM
Subject: RE: [asterisk-users] TNT Max Password reset



Yes, but they require a software upgrade. I am using them for plain ole
dialup, but because of that feature set I figured someone on this list 
would
know how. I find a lot of people online saying to serial into them and 
reset

them, but no detail on when to send break or what dip switches to set or
anything. :(


Natambu Obleton
Network Engineer
FastTrack Communications
[EMAIL PROTECTED]
(970) 247-3366 office
(970) 247-2426 fax

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay R.
Ashworth
Sent: Wednesday, October 04, 2006 4:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TNT Max Password reset

On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote:
   Anyone have happen know how to reset the password on a TNT Max? 
Thanks.


Does your asking here suggest that the the MAX's can do, say, voice
gateway service?  Protocols?  Codecs?

Cheers,
-- jra
--
Jay R. Ashworth
[EMAIL PROTECTED]
Designer  Baylink RFC
2100
Ashworth  AssociatesThe Things I Think'87
e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647
1274

That's women for you; you divorce them, and 10 years later,
  they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Bandwidth requirements

2006-10-05 Thread omar parihuana


For bandwidth requeriments don't forget Layer 2 overhead. I.e
Frame-relay overhead is lower than Ethernet overhead.

Rgds.

On 10/5/06, Jay R. Ashworth [EMAIL PROTECTED] wrote:

On Thu, Oct 05, 2006 at 09:44:56AM +0100, Brian Candler wrote:
 A 20ms packet duration means that 20ms of audio is stuffed into one IP
 packet. Since each packet carries 1/50th of a second of audio, that means
 you're generating 50 packets per second for each channel.

 With g729 your audio is 8000 bits per second.

 The overhead on each packet is 20 bytes (IP) + 8 bytes (UDP) + 12 bytes
 (RTP) = 40 bytes or 320 bits.

 So your bandwidth requirement per channel is:
 - 8000 bits per second for payload
 - 320x50 = 16000 bits per second for overhead
 making a total of 24000 bits per second.

 20 simultaneous calls is therefore 480,000 bits per second.

A reminder: much equipment, particularly low end/consumer equipment,
chokes *much* faster on high PPS than it does on high BPS.

While short packets are good for latency, they do impose stricter
engineering evaluation requirements on the other links in your chains.

Cheers,
-- jra
--
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth  AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

   That's women for you; you divorce them, and 10 years later,
 they stop having sex with you.  -- Jennifer Crusie; _Fast_Women_
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users




--
Omar E.P.T
-
Certified Networking Professionals make better Connections!

http://omarept.blogspot.com/

 Usysnet Corp
Open Source Solutions
www.usysnet.com.pe
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] pop a web page with DID in url

2006-10-05 Thread Michael Sampson


I'm looking to do this.
When a call comes in to an agent in a queue, pop a web page like this 
http://www.mydomain.com/cgi-bin/script.cgi?did=952900
Where did is the number the caller dialed to reach the system in the 
first place.


I know Hudlite can do this we caller ID, but the DID feature is not 
there yet.


Does anyone have any other software they know of that can do this?

--
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Bandwidth requirements