[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #3413 - 14 msgs

Jain, Sonal Sonal.Jain at Sterlingbancorp.com
Tue Apr 13 07:35:13 MST 2004


I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn on the mark echo canceller.

 -----Original Message-----
From: 	asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com]  On Behalf Of asterisk-users-request at lists.digium.com
Sent:	Tuesday, April 13, 2004 4:41 AM
To:	asterisk-users at lists.digium.com
Subject:	Asterisk-Users digest, Vol 1 #3413 - 14 msgs

Send Asterisk-Users mailing list submissions to
	asterisk-users at lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
	asterisk-users-request at lists.digium.com

You can reach the person managing the list at
	asterisk-users-admin at lists.digium.com

When replying, please edit your Subject line so it is more specific
than "Re: Contents of Asterisk-Users digest..."


Today's Topics:

   1. VoiceMailBox wav file format in EMAIL. (James Gardiner)
   2. TDM400P Issues (Jeremy Bogan)
   3. Re: TDM400P Issues (Vic Cross)
   4. Re: TDM400P Issues (Jeremy Bogan)
   5. Re: TDM400P Issues (Christian Hoffmeyer)
   6. Re: TDM400P Issues (Jeremy Bogan)
   7. Re: ZAPRTC question(s) (Tony Mountifield)
   8. Re: TDM400P Issues (Jeremy Bogan)
   9. Re: TDM400P Issues (Jeremy Bogan)
  10. Re: X100P and NTL (ex Cable + Wireless) (Stephen Davies)
  11. Re: TDM400P Issues (Vic Cross)
  12. Re: TDM400P Issues (Jeremy Bogan)
  13. Re: Dial Outside SIP address from AGI (Ron McMillin)
  14. Re: X100P and NTL (ex Cable + Wireless) (Vic Cross)

--__--__--

Message: 1
From: "James Gardiner" <asterisk at crafted.com.au>
To: <asterisk-users at lists.digium.com>
Date: Tue, 13 Apr 2004 16:12:15 +1000
Subject: [Asterisk-Users] VoiceMailBox wav file format in EMAIL.
Reply-To: asterisk-users at lists.digium.com


Hi all,
I am not sure if tis is a bug but..
Was learning about VM etc to see how it all worked, and I noticed the
following..

In the default install, the VM system leaves 3 different copies of the Voice
message.
Size	filename
13kb	Msg0000.gsm
13kb	Msg0000.wav
122kb	Msg0000.WAV  <- under UNIX we have case sensitive file names of
course.

I wanted to have a look at these files so loaded them into SOUND FORGE 6.
This first thing I noticed was that the LARGER file is of much HIGHER
volume. Like it had been normalised to 100%
The smaller was file, when loaded into sound forge, did not load properly,
only the first 2 seconds loads.

Can anyone explain these issues and why they exist?

All in all, I was wondering what would be the best format for best quality
but with still great compression.

I want to archive all calls for a period of time with self expire. (For
example dedicate 5 gig disk space to the last number of calls that can fit
in the 5gig.) I want to store the best quality possible but also make best
use of disk space, so I can store for even longer periods.  I was
considering ogg but then is occurred to me that GSM or other codecs designed
for audio with this frequency response may be better. (But the GSM file
above is not as clear as the WAV ones produced.)

I was also wondering if the VM system when emailing the audio can be setup
to use something like ogg or MP3?

Comments appreciated,
James Gardiner


--__--__--

Message: 2
To: asterisk-users at lists.digium.com
From: Jeremy Bogan <jeremy at segpub.com.au>
Date: Tue, 13 Apr 2004 16:14:43 +1000
Subject: [Asterisk-Users] TDM400P Issues
Reply-To: asterisk-users at lists.digium.com

Hi,

I just got my TDM400P card (2 modules) and i installed it no probs. The 
card is detected fine, but for some reason when I add the card to 
zaptel.conf i get the following error:

--snip--
ZT_CHANCONFIG failed on channel 2: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
and that FXO interfaces use FXS signalling?
--snip--

My /etc/zaptel.conf looks like:

--snip--
fxsks=1-2
fxoks=3-4
loadzone=au
defaultzone=au
--snip--

I currently have 2 x X100P cards that work no problem.

Running a ztcfg -vvvv I get:

Zaptel Configuration
======================


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXO Kewlstart (Default) (Slaves: 03)
Channel 04: FXO Kewlstart (Default) (Slaves: 04)

4 channels configured.

ZT_CHANCONFIG failed on channel 2: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
and that FXO interfaces use FXS signalling?


Anyone have any ideas? I tried doing a search but couldn't really find 
anything.

Thanks!

-- 
jeremy bogan    [ jeremy at segpub.com.au ]
segment publishing - design.develop.host


--__--__--

Message: 3
Date: Tue, 13 Apr 2004 16:28:09 +1000 (EST)
From: Vic Cross <vicc at veejoe.com.au>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] TDM400P Issues
Reply-To: asterisk-users at lists.digium.com

G'day Jeremy,

On Tue, 13 Apr 2004, Jeremy Bogan wrote:

> --snip--
> ZT_CHANCONFIG failed on channel 2: Invalid argument (22)
> Did you forget that FXS interfaces are configured with FXO signalling
> and that FXO interfaces use FXS signalling?
> --snip--
> 
> My /etc/zaptel.conf looks like:
> 
> --snip--
> fxsks=1-2
> fxoks=3-4
> loadzone=au
> defaultzone=au
> --snip--
> 
> I currently have 2 x X100P cards that work no problem.
> 
> Running a ztcfg -vvvv I get:
> 
> Zaptel Configuration
> ======================
> 
> 
> Channel map:
> 
> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
> Channel 02: FXS Kewlstart (Default) (Slaves: 02)
> Channel 03: FXO Kewlstart (Default) (Slaves: 03)
> Channel 04: FXO Kewlstart (Default) (Slaves: 04)
> 
> 4 channels configured.
> 
> ZT_CHANCONFIG failed on channel 2: Invalid argument (22)
> Did you forget that FXS interfaces are configured with FXO signalling
> and that FXO interfaces use FXS signalling?

This is your clue...  Do exactly what it says...

At a guess, I'd say you've loaded the wcfxs module before wcfxo.  This 
will push your existing X100P interfaces out to channels 3 and 4.  Either 
change your zaptel.conf to suit, or load wcfxo prior to wcfxs to have your 
FXO cards appear where they used to.

Cheers,
Vic Cross

--__--__--

Message: 4
From: Jeremy Bogan <jeremy at segpub.com.au>
Subject: Re: [Asterisk-Users] TDM400P Issues
Date: Tue, 13 Apr 2004 16:33:02 +1000
To: asterisk-users at lists.digium.com
Reply-To: asterisk-users at lists.digium.com

Hi Vic,

> This is your clue...  Do exactly what it says...
> At a guess, I'd say you've loaded the wcfxs module before wcfxo.  This
> will push your existing X100P interfaces out to channels 3 and 4.  
> Either
> change your zaptel.conf to suit, or load wcfxo prior to wcfxs to have 
> your
> FXO cards appear where they used to.

That's the problem, i've tried that. I swapped them around so that the 
X100P's are channel 3-4 and the TDM400P is channel 1-2, but the same 
thing:

ZT_CHANCONFIG failed on channel 1: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
and that FXO interfaces use FXS signalling?

When I startup zaptel it loads as follows:

Loading zaptel hardware modules: wcfxo wcfxs wcusb

I can't figure it out....

-- 
jeremy bogan    [ jeremy at segpub.com.au ]
segment publishing - design.develop.host


--__--__--

Message: 5
From: "Christian Hoffmeyer" <christian at yottadot.org>
To: <asterisk-users at lists.digium.com>
Subject: Re: [Asterisk-Users] TDM400P Issues
Date: Tue, 13 Apr 2004 01:39:02 -0500
Organization: www.yottadot.org
Reply-To: asterisk-users at lists.digium.com

----- Original Message ----- 
From: "Jeremy Bogan" <jeremy at segpub.com.au>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, April 13, 2004 1:33 AM
Subject: Re: [Asterisk-Users] TDM400P Issues


> That's the problem, i've tried that. I swapped them around so that the
> X100P's are channel 3-4 and the TDM400P is channel 1-2, but the same
> thing:
>
> ZT_CHANCONFIG failed on channel 1: Invalid argument (22)
> Did you forget that FXS interfaces are configured with FXO signalling
> and that FXO interfaces use FXS signalling?

Sounds like an irq problem. If it works with both x100ps in the same slots
without the tdm, but the tdm card makes it fail, try moving the tdm card to
another pci slot.

Christian Hoffmeyer
YottaDot Solutions
Huntsville, AL

(iax)  700.859.4508


--__--__--

Message: 6
From: Jeremy Bogan <jeremy at segpub.com.au>
Subject: Re: [Asterisk-Users] TDM400P Issues
Date: Tue, 13 Apr 2004 16:42:51 +1000
To: asterisk-users at lists.digium.com
Reply-To: asterisk-users at lists.digium.com

> Sounds like an irq problem. If it works with both x100ps in the same 
> slots
> without the tdm, but the tdm card makes it fail, try moving the tdm 
> card to
> another pci slot.

Thanks, i'll give it a try.

-- 
jeremy bogan    [ jeremy at segpub.com.au ]
segment publishing - design.develop.host


--__--__--

Message: 7
To: asterisk-users at lists.digium.com
From: tony at softins.clara.co.uk (Tony Mountifield)
Date:  Tue, 13 Apr 2004 06:51:44 +0000 (UTC)
Organization:  Software Insight Ltd., Winchester, UK
Subject: [Asterisk-Users] Re: ZAPRTC question(s)
Reply-To: asterisk-users at lists.digium.com

In article <1081802051.4782.2.camel at homebrew.bicester.partyvibe.com>,
Fran Boon <flavour at partyvibe.com> wrote:
> On Mon, 2004-04-12 at 17:37, Tony Mountifield wrote:
> > The zaprtc.c code is based on the rtc.c from 2.4.20. I am running 2.4.22,
> > so I isolated the zaprtc changes, and re-applied them to a copy of the
> > rtc.c from 2.4.22. It works a treat.
> > I've also enhanced rtcsetup to be a proper daemon.
> 
> Any chance of sharing these changes somewhere?
> e.g. Wiki

Yes, I'm intending to - I just haven't had the time to do so yet! :-)

Cheers,
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org

--__--__--

Message: 8
From: Jeremy Bogan <jeremy at segpub.com.au>
Subject: Re: [Asterisk-Users] TDM400P Issues
Date: Tue, 13 Apr 2004 16:53:43 +1000
To: asterisk-users at lists.digium.com
Reply-To: asterisk-users at lists.digium.com

>> Sounds like an irq problem. If it works with both x100ps in the same 
>> slots
>> without the tdm, but the tdm card makes it fail, try moving the tdm 
>> card to
>> another pci slot.

Swapped the card to another slot, still no dice :(

-- 
jeremy bogan    [ jeremy at segpub.com.au ]
segment publishing - design.develop.host


--__--__--

Message: 9
From: Jeremy Bogan <jeremy at segpub.com.au>
Subject: Re: [Asterisk-Users] TDM400P Issues
Date: Tue, 13 Apr 2004 16:59:13 +1000
To: asterisk-users at lists.digium.com
Reply-To: asterisk-users at lists.digium.com

Fixed it, had to modify my config, it now reads:

fxsks=1
fxoks=2-3
fxsks=4
loadzone=au
defaultzone=au

-- 
jeremy bogan    [ jeremy at segpub.com.au ]
segment publishing - design.develop.host


--__--__--

Message: 10
Date: Tue, 13 Apr 2004 09:03:25 +0200 (SAST)
From: Stephen Davies <steve at daviesfam.org>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] X100P and NTL (ex Cable + Wireless)
Reply-To: asterisk-users at lists.digium.com



On Tue, 13 Apr 2004, Alex Brett wrote:

> Has anybody got any experience using an X100P on an NTL phone line in 
> the UK (I'm in an ex Cable & Wireless area if that makes any difference).
> 
> The problem I'm having (and judging by the number of references to it 
> I've found searching it is a common one) is getting * to detect when the 
> line has been hung up.  It doesn't matter if it comes through to a 
> person directly as they can just hang that phone up, but when it hits 
> voicemail, and it sits there for two minutes recording an empty message, 
> and then emails it to the person it can be a bit annoying!

Hi Alex,

Indeed the call end termination doesn't work on an NTL line.  I'm not
so sure it works too well on other lines either.

I did some work a while back to add detection of the UK busy/hangup
signal on the line, but I never got it working well enough to depend
on it.  The problem is that it is a single frequency tone.  (The US
one is dual-tone).  Women's voices used to sometimes trigger my
detector - causing hangups.

The main practical issue is with voicemail, as you say.

My final solution was to switch to ISDN.

Steve



--__--__--

Message: 11
Date: Tue, 13 Apr 2004 17:11:57 +1000 (EST)
From: Vic Cross <vicc at veejoe.com.au>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] TDM400P Issues
Reply-To: asterisk-users at lists.digium.com

On Tue, 13 Apr 2004, Jeremy Bogan wrote:

> Fixed it, had to modify my config, it now reads:
> 
> fxsks=1
> fxoks=2-3
> fxsks=4
> loadzone=au
> defaultzone=au

This looks wrong.  What is the full output of ztcfg -vvv?

I'd be surprised if this worked as expected once you got * started...

Out of curiosity, what's the arrangement of the cards in the slots?  Is 
the TDM card between the two X100Ps?

Cheers,
Vic

--__--__--

Message: 12
From: Jeremy Bogan <jeremy at segpub.com.au>
Subject: Re: [Asterisk-Users] TDM400P Issues
Date: Tue, 13 Apr 2004 17:14:16 +1000
To: asterisk-users at lists.digium.com
Reply-To: asterisk-users at lists.digium.com

> This looks wrong.  What is the full output of ztcfg -vvv?

Zaptel Configuration
======================


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXO Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels configured.

> I'd be surprised if this worked as expected once you got * started...

Well now that I just tried * it doesn't work...

--snip--
  [chan_zap.so] => (Zapata Telephony w/PRI)
   == Parsing '/etc/asterisk/zapata.conf': Found
     -- Registered channel 1, FXS Kewlstart signalling
Apr 13 17:13:42 WARNING[16384]: chan_zap.c:665 zt_open: Unable to 
specify channel 4: No such device
Apr 13 17:13:42 ERROR[16384]: chan_zap.c:5319 mkintf: Unable to open 
channel 4: No such device
here = 0, tmp->channel = 4, channel = 4
Apr 13 17:13:42 ERROR[16384]: chan_zap.c:7355 setup_zap: Unable to 
register channel '4'
Apr 13 17:13:42 WARNING[16384]: loader.c:313 ast_load_resource: 
chan_zap.so: load_module failed, returning -1
   == Unregistered channel type 'Tor'
   == Unregistered channel type 'Zap'
     -- Unregistered channel 1
     -- Unregistered channel 2
Apr 13 17:13:42 WARNING[16384]: loader.c:408 load_modules: Loading 
module chan_zap.so failed!
--snip--

> Out of curiosity, what's the arrangement of the cards in the slots?  Is
> the TDM card between the two X100Ps?

The TDM is above the two X100P's, before it was below them.

-- 
jeremy bogan    [ jeremy at segpub.com.au ]
segment publishing - design.develop.host


--__--__--

Message: 13
Date: Tue, 13 Apr 2004 00:16:25 -0700 (PDT)
From: Ron McMillin <sipnow at sbcglobal.net>
Subject: Re: [Asterisk-Users] Dial Outside SIP address from AGI
To: asterisk-users at lists.digium.com
Reply-To: asterisk-users at lists.digium.com

--0-2089280042-1081840585=:81681
Content-Type: text/plain; charset=us-ascii

Thank you. This explains it.

Nathaniel Powning <nat at powning.org> wrote:
On Mon, 12 Apr 2004, Ron McMillin wrote:

> Is it possible to dial an OUTSIDE SIP address while inside AGI application? For example, within extension context, I could use
> [from-sip]
> exten => 7723,1,Dial(SIP/897224 at fwd) and this works
>
> whereas when I'm inside agi app,
> $AGI->exec('Dial',"SIP/897224 at fwd") and this DOESN'T work.

Perl will interpret the @ symbol as referencing an array, put a backslash
before that character in your SIP address.

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

--0-2089280042-1081840585=:81681
Content-Type: text/html; charset=us-ascii

<DIV>Thank you. This explains it.<BR><BR><B><I>Nathaniel Powning &lt;nat at powning.org&gt;</I></B> wrote:
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid"><BR>On Mon, 12 Apr 2004, Ron McMillin wrote:<BR><BR>&gt; Is it possible to dial an OUTSIDE SIP address while inside AGI application? For example, within extension context, I could use<BR>&gt; [from-sip]<BR>&gt; exten =&gt; 7723,1,Dial(SIP/897224 at fwd) and this works<BR>&gt;<BR>&gt; whereas when I'm inside agi app,<BR>&gt; $AGI-&gt;exec('Dial',"SIP/897224 at fwd") and this DOESN'T work.<BR><BR>Perl will interpret the @ symbol as referencing an array, put a backslash<BR>before that character in your SIP address.<BR><BR>_______________________________________________<BR>Asterisk-Users mailing list<BR>Asterisk-Users at lists.digium.com<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users</BLOCKQUOTE></DIV>
--0-2089280042-1081840585=:81681--

--__--__--

Message: 14
Date: Tue, 13 Apr 2004 17:32:10 +1000 (EST)
From: Vic Cross <vicc at veejoe.com.au>
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] X100P and NTL (ex Cable + Wireless)
Reply-To: asterisk-users at lists.digium.com

On Tue, 13 Apr 2004, Stephen Davies wrote:

> I did some work a while back to add detection of the UK busy/hangup
> signal on the line, but I never got it working well enough to depend
> on it.  The problem is that it is a single frequency tone.  (The US
> one is dual-tone).  Women's voices used to sometimes trigger my
> detector - causing hangups.

I'm looking at the same thing now, for AU busy tone.  If there's some 
work-in-progress that you wouldn't mind releasing, I'd be keen to have a 
look.

I think the problem with the current code (for us!) is the short length of
time over which it tests for busy.  Extending this might help prevent
voice-off.  It will be a balancing act though, as down here the ringing
indication is the same frequency tone (and I'd rather not have my outgoing
calls detected as busy when they are actually ringing).


Cheers,
Vic Cross


--__--__--

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users


End of Asterisk-Users Digest





More information about the asterisk-users mailing list