Re: [asterisk-users] Verification number / code
that is a bit heavy for me. how about some simple way to announce a random
number. using RAND. and saydigit
exten => s,1,Set(junky=${RAND(1,8)})
On Sat, Nov 21, 2009 at 7:20 PM, Steve Edwards wrote:
> On Sat, 21 Nov 2009, Thomas Perron wrote:
>
> > I want to distribute a random number to each of the first 100 callers to
> > my IVR. This random number will be matched to their telephone number.
> > Where in Asterisk can I do this? And, how?
> >
> > Logic.
> >
> > Call arrives.
> > Context for announcement begins.
> > You will receive a authentication code at the end of the message.
> > Then, if they press a certain digit to confirm then I simply pass a code
> to
> > them.
> > These codes are distributed to the first 100.
> > The 101st call does not get a code.
>
> I'm guessing you really don't want a random number since a random number
> generator can generate duplicates.
>
> "Matching" the number to their ANI also has issues. What if my ANI is
> blocked? What if I spoof my ANI? What if I call from a SIP phone?
>
> I would "pre-compute" the random numbers and store them in a database.
>
> When a call arrives, I would invoke an AGI that would lock the table, read
> the first value with a null ANI, update the row with the caller's ANI, and
> unlock the table.
>
> You could do it in dialplan, but I find database access in dialplan ugly.
>
> Alternatively, you could mung UNIQUEID ( Epoch>.) to
> appear to the caller as random and then store that and their ANI in a
> database.
>
> What happens if Asterisk is restarted in the middle of your campaign?
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
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Re: [asterisk-users] music on hold
On Thu, Nov 19, 2009 at 10:31 PM, wrote: > hello friends i want very simple thing in my dial plan. > > 1.When ever calls come at exten 2000 and if it is not answered with in 60 > secs it should hangup. Set absolute timeout to 60 seconds. > > 2.when ever call comes at exten 2000 and if it is answered within 60 secs > and if person who receives the call, puts the call on hold than music on > hold should begins. Setup music on hold: http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf > > 3.if music on hold is placed for more than 60 secs call should hangup. As far as I know, that is impossible to do with current code, since asterisk sees an answered call the same way a call thats place on hold, therefore asterisk has no way to distinguish between being on hold or actively talking on the phone. > > my extention.conf is like this > > vi /etc/asterisk/extentions.conf > > exten => 2000,1,Answer() > exten => 2000,n,Dial(SIP/2000,60) > exten => 2000,n,Dial(SIP/2000,60,m) > exten => 2000,n,Hangup > > the output of this is that when call is coming at exten 2000 call is > answered and another call comes n first call is on hold after 60 secs music > on hold starts but if i receive call before 60 secs even than MOH starts > even i dont put call on hold. > > thx > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verification number / code
On Sat, 21 Nov 2009, Thomas Perron wrote: > I want to distribute a random number to each of the first 100 callers to > my IVR. This random number will be matched to their telephone number. > Where in Asterisk can I do this? And, how? > > Logic. > > Call arrives. > Context for announcement begins. > You will receive a authentication code at the end of the message. > Then, if they press a certain digit to confirm then I simply pass a code to > them. > These codes are distributed to the first 100. > The 101st call does not get a code. I'm guessing you really don't want a random number since a random number generator can generate duplicates. "Matching" the number to their ANI also has issues. What if my ANI is blocked? What if I spoof my ANI? What if I call from a SIP phone? I would "pre-compute" the random numbers and store them in a database. When a call arrives, I would invoke an AGI that would lock the table, read the first value with a null ANI, update the row with the caller's ANI, and unlock the table. You could do it in dialplan, but I find database access in dialplan ugly. Alternatively, you could mung UNIQUEID (.) to appear to the caller as random and then store that and their ANI in a database. What happens if Asterisk is restarted in the middle of your campaign? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Verification number / code
I want to distribute a random number to each of the first 100 callers to my IVR. This random number will be matched to their telephone number. Where in Asterisk can I do this? And, how? Logic. Call arrives. Context for announcement begins. You will receive a authentication code at the end of the message. Then, if they press a certain digit to confirm then I simply pass a code to them. These codes are distributed to the first 100. The 101st call does not get a code. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDs
thanks
On Sat, Nov 21, 2009 at 12:26 PM, Steve Edwards
wrote:
> > Thomas Perron wrote:
> >
> >> I have two DID numbers. I want to configurate my IVR to initiate a
> >> context 1 if I dial DID 1. If DID2 is dialed then start with context 2.
>
> If the DIDs are from different providers, you can specify different
> contexts in [iax|sip].conf.
>
> On Sat, 21 Nov 2009, Alex Balashov wrote:
>
> > Assuming that the DID originator sends you the number in the Request
> > URI, you can just treat them like "extensions" in Asterisk. Example: if
> > you have DID 6789540670 and 6789540671:
> >
> >exten => 6789540670,1,Goto(context_1,${EXTEN},1)
> >exten => 6789540671,1,Goto(context_2,${EXTEN},1)
>
> I prefer to save EXTEN (after any pattern matching nonsense) in a more
> "meaningful" variable like DNIS and then use the "s" extension from then
> on. I find it "cleaner" and more maintainable. For example (typing off the
> top of my head):
>
> [incoming-from-xyz]
>exten = _678954067x,1, set(DNIS=${EXTEN})
>exten = 6789540670,2, goto(home,s,1)
>exten = 6789540671,2, goto(work,s,1)
>
> [home]
>exten = s,1,dial(sip/home-phone)
>
> [work]
>exten = s,1,dial(sip/work-phone)
>
> If I get another work number, I just add another line to
> incoming-from-xyz. If I change a number, I just change that single line.
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
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Re: [asterisk-users] DIDs
> Thomas Perron wrote:
>
>> I have two DID numbers. I want to configurate my IVR to initiate a
>> context 1 if I dial DID 1. If DID2 is dialed then start with context 2.
If the DIDs are from different providers, you can specify different
contexts in [iax|sip].conf.
On Sat, 21 Nov 2009, Alex Balashov wrote:
> Assuming that the DID originator sends you the number in the Request
> URI, you can just treat them like "extensions" in Asterisk. Example: if
> you have DID 6789540670 and 6789540671:
>
>exten => 6789540670,1,Goto(context_1,${EXTEN},1)
>exten => 6789540671,1,Goto(context_2,${EXTEN},1)
I prefer to save EXTEN (after any pattern matching nonsense) in a more
"meaningful" variable like DNIS and then use the "s" extension from then
on. I find it "cleaner" and more maintainable. For example (typing off the
top of my head):
[incoming-from-xyz]
exten = _678954067x,1, set(DNIS=${EXTEN})
exten = 6789540670,2, goto(home,s,1)
exten = 6789540671,2, goto(work,s,1)
[home]
exten = s,1,dial(sip/home-phone)
[work]
exten = s,1,dial(sip/work-phone)
If I get another work number, I just add another line to
incoming-from-xyz. If I change a number, I just change that single line.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] How to change outgoing DTMF frequencies on zaptel?
Hi, I am generating the tones from Asterisk, using senddtmf and option D in the Dial command. Is there no way to change it? Can I somehow modify it in tonezone.c and recompile Asterisk? -- Zeeshan A Zakaria On Sat, Nov 21, 2009 at 8:56 AM, Tilghman Lesher wrote: > On Friday 20 November 2009 17:03:20 Zeeshan Zakaria wrote: > > Hi, > > > > I am having this issue that with one of the Asterisk servers, on zaptel > > hardware, that DTMF tones are 10-30 Hz too high than the upper limit for > > any DTMF digit frequency. This is causing problem with the equipment on > the > > other end which is trying to recognize the DTMF digits. I tried different > > cards from the same vendor but it didn't help. For example for digit 1 > > where it should generate 697 Hz, it would generate 712 Hz, and where it > > should generate 1209 Hz, it would generate 1230 Hz. > > > > Talking to the hardware support, they said it is not dependent on the > > hardware. I googled and found very little information, but apparently it > > was an issue with ztcfg and zaptel, but it was fixed more than a year > ago. > > > > I found that in dsp.c file we could change the frequencies, but that is > > only for the incoming DTMF. > > > > Is there a way to change DTMF frequencies for the outgoing DTMF? If I > could > > drop all the frequencies by 15 Hz, that might solve the problem which I > am > > having. > > It's hardcoded in tonezone.c. Another possibility is that your phones are > generating tones that are out of whack with the standard, and Asterisk > isn't > detecting them as tones because they're so out of whack and they're simply > getting passed, unchanged, onto the remote side, where they are similarly > not > being detected. > > -- > Tilghman Lesher > Digium, Inc. | Senior Software Developer > twitter: Corydon76 | IRC: Corydon76-dig (Freenode) > Check us out at: www.digium.com & www.asterisk.org > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIDs
Hi Alex,
Thank you
Tom
On Sat, Nov 21, 2009 at 10:24 AM, Alex Balashov
wrote:
> Thomas,
>
> Thomas Perron wrote:
>
> > I have two DID numbers. I want to configurate my IVR to initiate a
> > context 1 if I dial DID 1.
> > If DID2 is dialed then start with context 2.
>
> Assuming that the DID originator sends you the number in the Request
> URI, you can just treat them like "extensions" in Asterisk. Example:
> if you have DID 6789540670 and 6789540671:
>
>exten => 6789540670,1,Goto(context_1,${EXTEN},1)
>exten => 6789540671,1,Goto(context_2,${EXTEN},1)
>
>[context_1]
>
>; IVR
>
>exten => 6789540670,1,Answer
>exten => 6789540670,n,Playback(hello-world)
>exten => 6789540670,n,Hangup
>
>[context_2]
>
>exten => 6789540671,1,Dial(SIP/abalashov,30,r)
>exten => 6789540671,n,Congestion
>
> -- Alex
>
> --
> Alex Balashov - Principal
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Connect two Asterisk Server in IAX ?
Title:
Hi
If you want to exchange call between two server I will do the following
iax svr1
register => svr2:12...@192.168.0.20
Hi
You need to have a user section that authenticate svr1 on svr2 and svr2
on svr1
Phibee Network Operation Center a écrit :
Hi
My first post get no answer :=<, i post new with new elements.
I have two Asterisk server, running on Asterisk 1.6:
SRV1 = 192.168.0.5 on Asterisk 1.6.1.4
SRV2 = 192.168.0.20 on Asterisk 1.6.1.8
I want create a link for exchange call.
on Srv1:
iax.conf:
[general]
bindport=4569
bindaddr=0.0.0.0
language=fr
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
calltokenoptional=192.168.0.20
[Srv2]
type=peer
host=192.168.0.20
qualify=yes
trunk=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
context=Incoming
peercontext=Incoming
extension.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
[Incoming]
exten => _X.,1,Playback(demo-thanks)
exten => _X.,2,Hangup
[Out]
exten => _201X.,1,Dial(IAX2/Srv2/${EXTEN:3},90,r)
exten => _201X.,2,Congestion
==
Srv1*CLI> iax2 show peers
Name/UsernameHost Mask Port Status
Srv2 192.168.0.20 (S) 255.255.255.255 4569 (E) OK (39 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
On Srv2
iax.conf
[general]
bindport=4569
bindaddr=0.0.0.0
language=fr
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
calltokenoptional=192.168.0.5
bandwidth=low
[Srv1]
type=peer
host=192.168.0.5
qualify=yes
trunk=no
encryption=aes128
disallow=all
allow=alaw
allow=g729
context=Incoming
peercontect=Incoming
extensions.conf:
[Incoming]
exten => _X.,1,Playback(demo-thanks)
exten => _X.,2,Hangup
[Out]
exten => _202X.,1,Dial(IAX2/Srv1/${EXTEN:3},90,r)
exten => _202X.,2,Congestion
===
trader-voip*CLI> iax2 show peers
Name/UsernameHost Mask Port Status
Srv1 192.168.0.5 (S) 255.255.255.255 4569 (E) OK (28 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]
===
All SIP Poste are connected and have in context in: Out
Now, when i call from a post connected on Srv1, i have this error on Srv1:
[Nov 21 08:09:44] WARNING[6407]: chan_iax2.c:9018 socket_process: Call
rejected by 192.168.0.20: No authority found
and on Srv2:
[Nov 21 08:09:44] NOTICE[9089]: chan_iax2.c:9785 socket_process:
Rejected connect attempt from 192.168.0.5, who was trying to reach
'1...@incoming'
125 are the number called (201125)
Dialplan on Srv2
Srv2*CLI> dialplan show Incoming
[ Context 'Incoming' created by 'pbx_config' ]
'_X.' => 1. Playback(demo-thanks)
[pbx_config]
2. Hangup()
[pbx_config]
-= 1 extension (2 priorities) in 1 context. =-
Anyone can help me for know where is my error ?
thanks
Jerome
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--
Sylvain
MEYNELLY
NEWTEK
BP 1496 Port-Gentil
GABON
Cell : +241 05 08 20 20
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Re: [asterisk-users] DIDs
Thomas,
Thomas Perron wrote:
> I have two DID numbers. I want to configurate my IVR to initiate a
> context 1 if I dial DID 1.
> If DID2 is dialed then start with context 2.
Assuming that the DID originator sends you the number in the Request
URI, you can just treat them like "extensions" in Asterisk. Example:
if you have DID 6789540670 and 6789540671:
exten => 6789540670,1,Goto(context_1,${EXTEN},1)
exten => 6789540671,1,Goto(context_2,${EXTEN},1)
[context_1]
; IVR
exten => 6789540670,1,Answer
exten => 6789540670,n,Playback(hello-world)
exten => 6789540670,n,Hangup
[context_2]
exten => 6789540671,1,Dial(SIP/abalashov,30,r)
exten => 6789540671,n,Congestion
-- Alex
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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[asterisk-users] DIDs
I have two DID numbers. I want to configurate my IVR to initiate a context 1 if I dial DID 1. If DID2 is dialed then start with context 2. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change outgoing DTMF frequencies on zaptel?
On Friday 20 November 2009 17:03:20 Zeeshan Zakaria wrote: > Hi, > > I am having this issue that with one of the Asterisk servers, on zaptel > hardware, that DTMF tones are 10-30 Hz too high than the upper limit for > any DTMF digit frequency. This is causing problem with the equipment on the > other end which is trying to recognize the DTMF digits. I tried different > cards from the same vendor but it didn't help. For example for digit 1 > where it should generate 697 Hz, it would generate 712 Hz, and where it > should generate 1209 Hz, it would generate 1230 Hz. > > Talking to the hardware support, they said it is not dependent on the > hardware. I googled and found very little information, but apparently it > was an issue with ztcfg and zaptel, but it was fixed more than a year ago. > > I found that in dsp.c file we could change the frequencies, but that is > only for the incoming DTMF. > > Is there a way to change DTMF frequencies for the outgoing DTMF? If I could > drop all the frequencies by 15 Hz, that might solve the problem which I am > having. It's hardcoded in tonezone.c. Another possibility is that your phones are generating tones that are out of whack with the standard, and Asterisk isn't detecting them as tones because they're so out of whack and they're simply getting passed, unchanged, onto the remote side, where they are similarly not being detected. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PCI analog cards on * vs. Quintum
What is the verdict? There was one positive response, but would like to hear a few more. In addition, what I am looking at is FXO ports to be used with GSM gateways, so any recommendations for specific cards are welcomed. From my experience with PRI cards, I am a little biased toward Sangoma. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't call through voip provider
Hello. I have my server running for about 30 days. Every time I did some changes to my sip.conf file I did reload in the cli. I thought this would change the new values. Somehow it wasn't. I decided to do a restart now and that used my new settings. The same settings I've been posting here the past week and weren't working. After restarting asterisk I'm able to use my provider via asterisk to make calls. I would like to thank those who helped me. --- On Fri, 11/20/09, Landy Landy wrote: > From: Landy Landy > Subject: Re: [asterisk-users] can't call through voip provider > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Date: Friday, November 20, 2009, 8:53 AM > Sorry to bother you again with my > problem but, is that I can't figure out what's going on with > my setup. I have no idea of why my asterisk server is not > communicating with my provider's. I've searched, googled, > and can't find my solution. I've followed many tutorials but > can't get anywhere. > > > > --- On Thu, 11/19/09, Landy Landy > wrote: > > > From: Landy Landy > > Subject: Re: [asterisk-users] can't call through voip > provider > > To: "Asterisk Users Mailing List - Non-Commercial > Discussion" > > Date: Thursday, November 19, 2009, 5:53 PM > > Nothing. I don't know what in the > > world is going on with my setup. > > > > Here's my FORWARD rules: > > eth0 = external nic, eth1 = lan > > > > 0 0 ACCEPT > >udp -- > > eth0 eth10.0.0.0/0 > > 0.0.0.0/0 > >udp dpts:5060:5070 > > 0 0 ACCEPT > >udp -- > > eth0 eth10.0.0.0/0 > > 0.0.0.0/0 > >udp dpts:1:10100 > > 162 ACCEPT > >udp -- > > eth1 eth00.0.0.0/0 > > 0.0.0.0/0 > >udp dpts:5060:5070 > >36 2372 ACCEPT > >udp -- > > eth1 eth00.0.0.0/0 > > 0.0.0.0/0 > >udp dpts:1:10100 > > 0 0 ACCEPT > >tcp -- > > eth0 eth10.0.0.0/0 > > 0.0.0.0/0 > >tcp dpts:5060:5070 > > 0 0 ACCEPT > >tcp -- > > eth0 eth10.0.0.0/0 > > 0.0.0.0/0 > >tcp dpts:1:10100 > > 0 0 ACCEPT > >tcp -- > > eth1 eth00.0.0.0/0 > > 0.0.0.0/0 > >tcp dpts:5060:5070 > > 3 144 ACCEPT > >tcp -- > > eth1 eth00.0.0.0/0 > > 0.0.0.0/0 > >tcp dpts:1:10100 > > > > > > and now the debug: > > > > etransmitting #5 (NAT) to 190.80.152.200:5060: > > INVITE sip:18292574...@optimumwireless.myvnc.com > > SIP/2.0 > > Via: SIP/2.0/UDP > > 190.80.152.200:5060;branch=z9hG4bK794de7aa;rport > > Max-Forwards: 70 > > From: "102" > > ;tag=as5084570c > > To: > > Contact: > > Call-ID: > 22569d3b767276276c6c65c84b314...@190.80.152.200 > > CSeq: 102 INVITE > > User-Agent: Asterisk PBX 1.6.1.5 > > Date: Thu, 19 Nov 2009 22:53:06 GMT > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, > SUBSCRIBE, > > NOTIFY, INFO > > Supported: replaces, timer > > Content-Type: application/sdp > > Content-Length: 475 > > > > v=0 > > o=root 135722140 135722140 IN IP4 190.80.152.200 > > s=Asterisk PBX 1.6.1.5 > > c=IN IP4 190.80.152.200 > > t=0 0 > > m=audio 10074 RTP/AVP 0 3 8 112 5 10 7 111 9 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:112 AAL2-G726-32/8000 > > a=rtpmap:5 DVI4/8000 > > a=rtpmap:10 L16/8000 > > a=rtpmap:7 LPC/8000 > > a=rtpmap:111 G726-32/8000 > > a=rtpmap:9 G722/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > a=sendrecv > > > > > > > > I'm already frustrated with this. > > > > > > --- On Thu, 11/19/09, Warren Selby > > wrote: > > > > > From: Warren Selby > > > Subject: Re: [asterisk-users] can't call through > voip > > provider > > > To: "Asterisk Users Mailing List - > Non-Commercial > > Discussion" > > > Date: Thursday, November 19, 2009, 5:11 PM > > > On Thu, Nov 19, > > > 2009 at 3:36 PM, Landy Landy > > > wrote: > > > > > > Can someone please share with me a sip > configuration > > to > > > connect an asterisk server to a voip provider > since > > my > > > configuration isn't working for me. > > > > > > > > > > > > thanks. > > > > > > > > > > > > > > > Who is your voipprovider? Did they give you > the > > settings > > > you're using in your sip.conf? Also, you've > got > > > some typos in your sip config (insucure = > insecure, > > > careinvite = canreinvite). You could try > something > > like > > > this: > > > > > > > > > [voipprovider] > > > > > > type=peer > > > > > > host=208.78.163.3 > > > > > > username=77000 > > > > > > fromuser=77000 > > > > > > secret=77000 > > > > > > port=5060 > > > > > > dtmfmode=rfc2833 > > > > > > nat=yes > > > canreinvite=yes > > > > > > insecure=very > > > disallow=all > > > allow=ulaw > > > allow=alaw > > > > > > > > > > > > > > > > > > -- > > > Thanks, > > > --Warren Selby > > > http://www.s
