[Asterisk-Users] ISDN Dialout

2003-10-06 Thread Jay Tyndall 
Hi,

I am having some trouble with ISDN Dialout. Using a Netjet-s PCI Card.

When in Minicom, the only way I can dialout is if i issue ATS18=1 First.
Otherwise I get a BUSY message.  So thats fine.
But when I dialout from asterisk, I get an immediate hangup, so my guess is 
that asterisk is not issuing ATS18=1 to the ttyI device.

Here are my configs, any input would be greatly appriciated.



extensions.conf
exten => 4000,1,Dial,Modem/g3:0422xx|60|r


modems.conf
; net jet suff
context=default
msn=0269xx&L*
icomingmsn=0269xx&L*
driver => i4l
group=2
stripmsd=1
mode=immediate
device => /dev/ttyI0
msn=0269xx
group=3
device => /dev/ttyI1
mode => immediate
type => autodetect
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Re: [Asterisk-Users] ISDN

2003-09-03 Thread Jay Tyndall 
Stripmsd is commented out, problem still occurs.
Does this simply use ATDT to dialout ?
When I attempt to dialout using minicom it comes back with "NO MSN/EAZ"
Looks like I may need to issue another AT Command to the netjet to set the 
MSN...

Has anyone encountered this before?

Jay

Message: 10
Date: Wed, 3 Sep 2003 07:35:02 +0200 (CEST)
From: Jac Kersing <[EMAIL PROTECTED]>
To: Jay Tyndall <[EMAIL PROTECTED]>
Cc: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] ISDN
Reply-To: [EMAIL PROTECTED]
On Wed, 3 Sep 2003, Jay Tyndall wrote:
I am using a Netjet-s ISDN Card, and am having some trouble dialling out 
(Incoming Works Fine).
...
I get the following when diallingout:
-- Starting simple switch on 'Zap/2-1'
-- Executing Dial("Zap/2-1", "Modem/ttyI0/04||Ttm") in new
Check the line 'stripmsd=1'. If the number to be dialed does not need to
have the most significant digit stripped this line needs to be
commented/removed in order to dial a valid number. (with stripmsd active
the number dialed in your case would be 4)
Regards,
Jac
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Re: [Asterisk-Users] DTMF Tones During Call

2003-09-03 Thread Jay Tyndall 
Can someone please point me in the right direction to find some info on 
this?
I have searched the mailling list archive on asterisk.org but cannot find 
the answer...

Thanks again.
Jay


Message: 11
Date: Wed, 3 Sep 2003 07:37:06 +0200 (CEST)
From: Jac Kersing <[EMAIL PROTECTED]>
To: Jay Tyndall <[EMAIL PROTECTED]>
Cc: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] DTMF Tones During Call
Reply-To: [EMAIL PROTECTED]
On Wed, 3 Sep 2003, Jay Tyndall wrote:
I am receiving calls via a Netjet-S card on asterisk, and I notice that 
whenever I am talkimng to someone, if their voice is loud enough, 
sometimes asterisk generates a DTMF Tone as they speak. that is played to 
me. (Caller doesn't hear it).

Any ideas how to stop this?
Check the list archive. This has been discussed a number of times. You'll
need patches (should be archived as well.)
Regards,
Jac
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[Asterisk-Users] DTMF Tones During Call

2003-09-02 Thread Jay Tyndall 
Hi,

I am receiving calls via a Netjet-S card on asterisk, and I notice that 
whenever I am talkimng to someone, if their voice is loud enough, sometimes 
asterisk generates a DTMF Tone as they speak. that is played to me. (Caller 
doesn't hear it).

Any ideas how to stop this?

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[Asterisk-Users] ISDN

2003-09-02 Thread Jay Tyndall 
Hi,

I am using a Netjet-s ISDN Card, and am having some trouble dialling out 
(Incoming Works Fine).

TRUNK=Modem/ttyI0
exten => _90X,1,Dial(${TRUNK}/${EXTEN:1}||Ttm)
exten => _90X,2,Congestion
I get the following when diallingout:
  -- Starting simple switch on 'Zap/2-1'
   -- Executing Dial("Zap/2-1", "Modem/ttyI0/04||Ttm") in new 
stack
 == Everyone is busy at this time
   -- Executing Congestion("Zap/2-1", "") in new stack
 == Spawn extension (local, 90422456118, 2) exited non-zero on 'Zap/2-1'
   -- Hungup 'Zap/2-1'
	
I have tried inserting a "v" infront of the number, but to no avail.

Any Ideas??
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[Asterisk-Users] Faking Ring tone

2003-08-14 Thread Jay Tyndall 
When doing a call transfer
exten => 201,1,BackGround(transfer)
exten => 201,2,Dial,Zap/2|40|tr
How Can I fake the ring tone (during transfer) to one that is defined in 
indications.conf ??

I have tried the following: But it still rings in the users phone after the 
zap phone has picked up.
exten => 201,1,BackGround(transfer)
exten => 201,2,Playtones,ring
exten => 201,3,Dial,Zap/2|40|tr

THanks
Jay.
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[Asterisk-Users] Direct Indial with ISDN and Netjet-S

2003-07-26 Thread Jay Tyndall 


Hi,

I am looking at using a Netjet-S ISDN card with Asterisk, and would like to 
know if it is possible
for asterisk to determine the dialled number (From the Indial Number range) 
and route the call accordingly.

Would I just set up an extension number based on the Indial number?

Thanks
Jay
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[Asterisk-Users] Call Parking

2003-07-18 Thread Jay Tyndall 
On the topic of call parking,  once a call is parked,  how do I stop 
asterisk ringing back the station if the call isn't picked up ?

Sometimes a caller would be put on hold, and left for a minute or so, when 
we come back we have found its gone to voicemail.

Would be nice to turn this "ringback" off. (or at least set the ringback 
time to a longer number)

Thanks
Jay.
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[Asterisk-Users] Call Pickup

2003-07-16 Thread Jay Tyndall 
	Hi,

I have been trying to workout how to use the call pickup.

So Far, I have the following in zapata.conf
[channels]
signalling => fxo_ks
context => local
pickupgroup=1
callgroup=1
channel => 1-3
When I dial *8# all I hear is busy tone.

What have I missed?

thanks
Jay.
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[Asterisk-Users] Analog commands

2003-07-15 Thread Jay Tyndall 
Hi,

When I use the analog phone connected to Zap/1 how do I transfer & hold the 
caller ?

When I hit the flash key, all that happens is the caller hears a beep 
(sounds like DTMF).
But no stutter dial tone on the Zap/1 Port, just continuing conversation 
with the caller.

What could be wrong here?

Cheers
Jay
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[Asterisk-Users] Making Analog Phones Work

2003-07-14 Thread Jay Tyndall 
Hi,

I have got my TDM400P working.(3 modules), asterisk dials Zap/1 and says 
"Ringing" but the analogue phone plugged in, does not ring, or does not 
have any tone when I pickup the handpiece.

Here are by configs:
zapata.conf:
[channels]
signalling => fxo_ks
context=internal
channel => 1-3
zaptel.conf:
fxoks=1-3
Any ideas would be greatly appriciated Thanks
Jay
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Re: [Asterisk-Users] Setting up A TDM400P

2003-07-14 Thread Jay Tyndall 
Thanks for the reply,

I have made those changes and still get the following error:

WARNING[16384]: File chan_zap.c, Line 576 (zt_open): Unable to specify 
channel 1: No such device or address
ERROR[16384]: File chan_zap.c, Line 4746 (mkintf): Unable to open channel 
1: No such device or address
here = 0, tmp->channel = 0, channel = 1
ERROR[16384]: File chan_zap.c, Line 6404 (load_module): Unable to register 
channel '1'
WARNING[16384]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: 
load_module failed, returning -1
WARNING[16384]: File loader.c, Line 394 (load_modules): Loading module 
chan_zap.so failed!

I find it quite odd, expecially since the card is detected and showing when 
I do a dmesg.

Jay

On Sun, 13 Jul 2003 23:54:13 -0500, John Bigelow <[EMAIL PROTECTED]> 
wrote:

The channel has to come after the signalling and other configuration 
lines.
It should look something like this:

signalling=fxo_ks
context=internal
channel => 1-3
Don't forget to configure zaptel.conf either. Add this line to it.

fxoks=1-3

-John

- Original Message - From: "Jay Tyndall" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, July 13, 2003 11:38 PM
Subject: [Asterisk-Users] Setting up A TDM400P



Hi,

I am having some trouble getting a TDM400P working, and would be very
appriciative of  some ideas.
I have installed a TDM400P, and downloaded the appropriate files from 
CVS,
compiled, installed and modprobe'd the devices.

From dmesg:
Module 0: Initialized
Module 1: Initialized
Module 2: Initialized
Module 3: Not installed
Found a Wildcard FXS: Wildcard S400P Prototype (4 modules)
So, it has found the card OK,  but I am  little confused as to what I 
need
to put in zapata.conf to make these 3 ports work with the analog phones 
I
have plugged into these ports.

I have tried looking at the docs on the digium site, but cannot seem to
get
it worked out.

I tried putting the following in zapata.conf
channel => 1
signalling => fxo_ks
And this in extensions.conf
exten => 200,1,Dial(Zap/1)
When I start asterisk, I get the following:
[chan_zap.so] => (Zapata Telephony)
WARNING[16384]: File chan_zap.c, Line 6654 (load_module): Ignoring
switchtype
WARNING[16384]: File chan_zap.c, Line 6654 (load_module): Ignoring 
rxwink
WARNING[16384]: File chan_zap.c, Line 576 (zt_open): Unable to specify
channel 1: No such device or address
ERROR[16384]: File chan_zap.c, Line 4746 (mkintf): Unable to open 
channel
1: No such device or address
here = 0, tmp->channel = 0, channel = 1
ERROR[16384]: File chan_zap.c, Line 6404 (load_module): Unable to 
register
channel '1'
WARNING[16384]: File loader.c, Line 299 (ast_load_resource): 
chan_zap.so:
load_module failed, returning -1
WARNING[16384]: File loader.c, Line 394 (load_modules): Loading module
chan_zap.so failed!

Thanks for your help
Jay
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[Asterisk-Users] Setting up A TDM400P

2003-07-13 Thread Jay Tyndall 


Hi,

I am having some trouble getting a TDM400P working, and would be very 
appriciative of  some ideas.

I have installed a TDM400P, and downloaded the appropriate files from CVS, 
compiled, installed and modprobe'd the devices.

From dmesg:
Module 0: Initialized
Module 1: Initialized
Module 2: Initialized
Module 3: Not installed
Found a Wildcard FXS: Wildcard S400P Prototype (4 modules)
So, it has found the card OK,  but I am  little confused as to what I need 
to put in zapata.conf to make these 3 ports work with the analog phones I 
have plugged into these ports.

I have tried looking at the docs on the digium site, but cannot seem to get 
it worked out.

I tried putting the following in zapata.conf
channel => 1
signalling => fxo_ks
And this in extensions.conf
exten => 200,1,Dial(Zap/1)
When I start asterisk, I get the following:
[chan_zap.so] => (Zapata Telephony)
WARNING[16384]: File chan_zap.c, Line 6654 (load_module): Ignoring 
switchtype
WARNING[16384]: File chan_zap.c, Line 6654 (load_module): Ignoring rxwink
WARNING[16384]: File chan_zap.c, Line 576 (zt_open): Unable to specify 
channel 1: No such device or address
ERROR[16384]: File chan_zap.c, Line 4746 (mkintf): Unable to open channel 
1: No such device or address
here = 0, tmp->channel = 0, channel = 1
ERROR[16384]: File chan_zap.c, Line 6404 (load_module): Unable to register 
channel '1'
WARNING[16384]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: 
load_module failed, returning -1
WARNING[16384]: File loader.c, Line 394 (load_modules): Loading module 
chan_zap.so failed!

Thanks for your help
Jay
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Re: [Asterisk-Users] IAX G729 Codec

2003-07-09 Thread Jay Tyndall 
This is going across a 256k/64 to a 512k/128.
They are about 2 hops away from each other and ping times are sub 70ms.
(Even when the * audio is playing)


- Original Message - 
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, July 10, 2003 12:02 PM
Subject: Re: [Asterisk-Users] IAX G729 Codec


> On Wed, 2003-07-09 at 18:06, Jay Tyndall wrote:
> > Hi,
> >
> > I have recently purchased some Asterisk G729 Codecs and installed them
> > to overcome by bandwidth problem I was having with GSM.
> >
> > The G729 keeps the pings nice and low, but the audio stutters or
> > jitters a fair bit.  (Starts and stops)
> >
> > Any Idea what would be causing this ?  I am just testing it using the
> > OSS/Console at the moment, as I am waiting for my Digium cards to
> > arrive.
>
> No zapata device for timing would probably be one thing.
>
> Also, what kind of network are you crossing? I have found some problems
> when the network is not instantaneous, but not internet level lagging.
> -- 
> Steven Critchfield <[EMAIL PROTECTED]>
>
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[Asterisk-Users] IAX G729 Codec

2003-07-09 Thread Jay Tyndall 



Hi,
 
I have recently purchased some Asterisk G729 Codecs 
and installed them to overcome by bandwidth problem I was having with 
GSM.
 
The G729 keeps the pings nice and low, but the 
audio stutters or jitters a fair bit.  (Starts and stops)
 
Any Idea what would be causing this ?  I am 
just testing it using the OSS/Console at the moment, as I am waiting for my 
Digium cards to arrive.
 
Thanks
Jay

Fw: [Asterisk-Users] IAX Bandwidth Question

2003-07-07 Thread Jay Tyndall 
Subject: Re: [Asterisk-Users] IAX Bandwidth Question


 Hi,

 I have changed the codec to lpc10 (I see what they mean by Mr. Roboto!!)
and
 the ping times generally dont go over 300ms.
 It seems very odd that GSM saturates the link.

 I would love to give g.723.1 a try but have no idea where to get the
proprietary codec... can anyone help?

Thanks
 Jay

> - Original Message - 
> From: "WipeOut ." <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, July 07, 2003 10:14 PM
> Subject: Re: [Asterisk-Users] IAX Bandwidth Question
>
>
> > I have 2 asterisk systems connected by a 56kbps internet dialup (so at
> best 33.6k in both directions) using IAX and GSM...
> >
> > The one * box is at my home and the other is in the office (before
anyone
> freaks this is a test environment)... Provided nothing is using the line
at
> the same time I am able to carry 1 voice call over the dialup link with no
> real latency.. Ping times when no call is in session are about 165-175ms
and
> when a call is active 190-240ms.. I just ran a call for 5 mins to see if
the
> ping time climbed as you are experiencing and it didn't increase at all..
> >
> > So you may have some other issue causing the increase in ping times..
> >
> > Sorry I probably wasn't much help..
> >
> > > Hi,
> > >
> > > I am using IAX to communicate between 2 sites, each site is using a
> 256k/64k ADSL Connection.
> > >
> > >
> > > I have noticed that when I connect my ping time to my 1st hop jumps
from
> 30~ms to over 12,000ms in over a period of about 10 minutes, it just keeps
> climbing until the link is saturated.
> > >
> > > Naturally, there is a very long delay when speaking.
> > >
> > > What bandwidth would be adequate for IAX? or how can I tune my config
to
> work better with my current bandwidth situation.
> > > I am using GSM codec and bandwidth=low in iax.conf
> > >
> > > Thanks in advance.
> > >
> > Jay.
> > -- 
> > __
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> >
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>

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[Asterisk-Users] IAX Bandwidth Question

2003-07-07 Thread Jay Tyndall 



Hi,
 
I am using IAX to communicate between 2 sites, each 
site is using a 256k/64k ADSL Connection.
 
 
I have noticed that when I connect my ping time to 
my 1st hop jumps from 30~ms to over 12,000ms in over a period of about 10 
minutes, it just keeps climbing until the link is saturated.
 
Naturally, there is a very long delay when 
speaking.
 
What bandwidth would be adequate for IAX? or how 
can I tune my config to work better with my current bandwidth 
situation.
I am using GSM codec and bandwidth=low in 
iax.conf
 
Thanks in advance.
 
Jay.

[Asterisk-Users] Fw: SIP Client X-Lite

2003-07-05 Thread Jay Tyndall 



Hi,
 
I have setup X-Lite to dial into our Asterisk box 
using SIP.
It connects, and i can dial the extension no. 
and hear ring tone.
But cannot hear any of the GSM audio.
There is a message that says:
 
WARNING[278542]: File dsp.c, Line 1106 
(ast_dsp_process): Unable to detect process 2 frames.
 
Is this a simple fix? any pointers in the right 
direction would be great!
 
Thanks
Jay