[Asterisk-Users] RE: Asterisk-Users Digest, Vol 21, Issue 130
Jordan Novak
jnovak at logisticshealth.com
Sun Apr 23 09:12:02 MST 2006
Have you thought about making them agents, they would both be reachable by dialing there agent number then, and I know only one agent can be logged in at once. Just a thought.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com
Sent: Saturday, April 22, 2006 8:26 PM
To: asterisk-users at lists.digium.com
Subject: Asterisk-Users Digest, Vol 21, Issue 130
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Today's Topics:
1. Re: Sipura SP3000 question (Roshan Sembacuttiaratchy)
2. Re: Sipura SP3000 question (Gonzalo Servat)
3. Re: PANASONIC KX-TS208W - Speakerphone Incompatible With
Asterisk 1.2.3 (broadbandvoice at comcast.net)
4. Re: Sipura SP3000 question (Rich Adamson)
5. How can I get a recording from a CD to my asterisk digital
assistant (Davi-Ann)
6. Asterisk on FreeBSD + Passive ISDN BRI (Cian Hughes)
7. Re: How can I get a recording from a CD to my asterisk
digital assistant (Alberto Sagredo)
8. Re: PANASONIC KX-TS208W - Speakerphone Incompatible With
Asterisk 1.2.3 (John Novack)
9. Re: How can I get a recording from a CD to myasterisk digital
assistant (Davi-Ann)
10. Re: RE: SPA 3000 - UK Replacement (Wayne)
11. Re: Sipura SP3000 question (Wayne)
12. RE: Pinouts for T1/E1 crossover cable WAS "RE:
[Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ?"
(Steven Totaro)
13. RE: Pinouts for T1/E1 crossover cable WAS "RE:
[Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?"
(Steven Totaro)
14. RE: Don't see my post (broadbandvoice at comcast.net)
15. RE: How to restrict simultaneous phone registrations
(broadbandvoice at comcast.net)
16. RE: No DTMF (broadbandvoice at comcast.net)
17. RE: Don't see my post (Steven Totaro)
----------------------------------------------------------------------
Message: 1
Date: Sat, 22 Apr 2006 19:17:52 +0000
From: Roshan Sembacuttiaratchy <rns.asterisklist.n.semba at xoxy.net>
Subject: Re: [Asterisk-Users] Sipura SP3000 question
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <20060422191752.GR24768 at roshan.info>
Content-Type: text/plain; charset=us-ascii
On Sat, Apr 22, 2006 at 11:19:35PM +1000, RumaTech scribbled:
> As this part is still in testing, I want all the outgoing calls got to
> PSTN by default and dial, say 0, to get an "outside VoIP line".
> I would like to do it as part of SP3000 configuration, not as part of
> * dialplan. Can someone help me?
I use the following dialplan within the Sipura:
([2-79]11<:@gw0>|999<:@gw0>|112<:@gw0>|0[12]x.|[*x]xx.<:@gw0>|<#9,:>[*x]x.|**)
Using this, all emergency numbers go directly to PSTN, all numbers
starting with 01 and 02 go via VoIP, and all other numbers go through
PSTN. Any number prefixed with #9 is then forced to go through VoIP,
with the initial #9 not being passed to Asterisk.
Adapt and use. :-)
Hope this helps,
Roshan
--
http://roshan.info
Be different, act normal.
------------------------------
Message: 2
Date: Sun, 23 Apr 2006 05:34:15 +1000
From: "Gonzalo Servat" <gservat at gmail.com>
Subject: Re: [Asterisk-Users] Sipura SP3000 question
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<dcc007e10604221234o415d5009od0ab1720bac4ff6c at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
On 4/23/06, Roshan Sembacuttiaratchy <rns.asterisklist.n.semba at xoxy.net> wrote:
> I use the following dialplan within the Sipura:
>
> ([2-79]11<:@gw0>|999<:@gw0>|112<:@gw0>|0[12]x.|[*x]xx.<:@gw0>|<#9,:>[*x]x.|**)
[..snip..]
Is this @stuff something new in the SPA3000 dialplan syntax? I have
SPA-200x ATAs and I never saw any mention of this in the manual, which
makes sense if it's a SPA3k new dialplan feature.
Cheers,
Gonzalo.
------------------------------
Message: 3
Date: Sat, 22 Apr 2006 19:42:28 +0000
From: broadbandvoice at comcast.net
Subject: Re: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone
Incompatible With Asterisk 1.2.3
To: jnovack at stromberg-carlson.org, Asterisk Users Mailing List -
Non-Commercial Discussion <asterisk-users at lists.digium.com>
Message-ID:
<042220061942.1215.444A87240001E385000004BF220073407608010B020E9B02 at comcast.net>
Content-Type: text/plain; charset="us-ascii"
Thanks for the response, I'll ask the client to change batteries, though it is a new phone less than two weeks. is there any reason why the Lanline(Verizon) work and not the Asterisk? The only differences is the Asterisk, Linksys router and the DSL modem. One of these 3 should be interfering.
-------------- Original message --------------
From: John Novack <jnovack at stromberg-carlson.org>
>
>
>
>
> broadbandvoice at comcast.net wrote:
>
> > I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does
> > not work with it. It works fine when you pick up the handset. Anyone
> > experinced this problem before, the speaker works fine with Verizon
> > line. The phone is behind a Linsys router RT31P2.
> > Replace the batteries! Alkaline only, replace every 6 months
> > 1.2.3 is also defective for other reasons. Upgrade
> > Using a TDM400, an ATA or ??
> > The phone works best with a 48V 20mA or better loop, so the FXS source
> > voltage may have an effect, and this cheap phone has no previsions
> > for external power.
> >
> > John Novack
>
>
> _______________________________________________
> --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
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Message: 4
Date: Sat, 22 Apr 2006 14:52:05 -0500
From: Rich Adamson <radamson at routers.com>
Subject: Re: [Asterisk-Users] Sipura SP3000 question
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <444A8965.60503 at routers.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Gonzalo Servat wrote:
> On 4/23/06, Roshan Sembacuttiaratchy <rns.asterisklist.n.semba at xoxy.net> wrote:
>> I use the following dialplan within the Sipura:
>>
>> ([2-79]11<:@gw0>|999<:@gw0>|112<:@gw0>|0[12]x.|[*x]xx.<:@gw0>|<#9,:>[*x]x.|**)
> [..snip..]
>
> Is this @stuff something new in the SPA3000 dialplan syntax? I have
> SPA-200x ATAs and I never saw any mention of this in the manual, which
> makes sense if it's a SPA3k new dialplan feature.
That dialplan function has been around since v2 code for the spa3k, but
using the gw0 and gw1 part of it only applies to the spa3k. The gw0
implies the physical fxo pstn port.
------------------------------
Message: 5
Date: Sat, 22 Apr 2006 15:59:54 -0400
From: "Davi-Ann" <dthurn at tstt.net.tt>
Subject: [Asterisk-Users] How can I get a recording from a CD to my
asterisk digital assistant
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <002c01c66647$52e2e330$b4dbdbdb at infosys200>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original
I got someone to record the messages we want for our auto-attendant menu on
a CD.
All I have to do not is to upload the files into the asterisk box, however
the format is not recognized by the Asterisk box.
Question 1) What formats should the sound file be, so I can upload it to my
asterisk box?
Thanks
--Davi-Ann
------------------------------
Message: 6
Date: Tue, 24 May 2005 18:04:20 +0100
From: Cian Hughes <cianlists at cian.ws>
Subject: [Asterisk-Users] Asterisk on FreeBSD + Passive ISDN BRI
To: Asterisk on BSD discussion <asterisk-bsd at lists.digium.com>,
asterisk-users at lists.digium.com, freebsd-isdn at freebsd.org
Message-ID: <95321E77-C79C-4BB7-9E5B-739849F6C365 at cian.ws>
Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
Ok, from what I can see _NO_ passive ISDN cards will work with
Asterisk on freebsd, is this correct & is it likely to change soon?
Secondly, if this is likely to be the way for a while, what is the
lease expensive card that will work with FreeBSD?
Also, can I use DID (Direct Inward Dialling) on FreeBSD?
Thanks for all your help to date.
Regards,
Cian Hughes
_______________________________________________
freebsd-isdn at freebsd.org mailing list
http://lists.freebsd.org/mailman/listinfo/freebsd-isdn
To unsubscribe, send any mail to "freebsd-isdn-unsubscribe at freebsd.org"
------------------------------
Message: 7
Date: Sat, 22 Apr 2006 22:15:32 +0200
From: Alberto Sagredo <asagredo at peoplecall.com>
Subject: Re: [Asterisk-Users] How can I get a recording from a CD to
my asterisk digital assistant
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <444A8EE4.8080809 at peoplecall.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
You will need them in one of asterisk supported formats.
wav, slin,gsm, g729, g723...
Davi-Ann escribió:
> I got someone to record the messages we want for our auto-attendant
> menu on a CD.
>
> All I have to do not is to upload the files into the asterisk box,
> however the format is not recognized by the Asterisk box.
>
> Question 1) What formats should the sound file be, so I can upload it
> to my asterisk box?
>
> Thanks
> --Davi-Ann
>
> _______________________________________________
> --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
>
------------------------------
Message: 8
Date: Sat, 22 Apr 2006 16:33:04 -0400
From: John Novack <jnovack at stromberg-carlson.org>
Subject: Re: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone
Incompatible With Asterisk 1.2.3
To: broadbandvoice at comcast.net
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <444A9300.90202 at stromberg-carlson.org>
Content-Type: text/plain; charset="us-ascii"
The book states "batteries not supplied" so perhaps they were never
installed?
And what FXS circuit are you using to interface to Asterisk?
The difference in loop current between VeriZon and the local interface
could be an answer
John Novack
broadbandvoice at comcast.net wrote:
> Thanks for the response, I'll ask the client to change batteries,
> though it is a new phone less than two weeks. is there any reason why
> the Lanline(Verizon) work and not the Asterisk? The only differences
> is the Asterisk, Linksys router and the DSL modem. One of these 3
> should be interfering.
>
>
> -------------- Original message --------------
> From: John Novack <jnovack at stromberg-carlson.org>
>
> >
> >
> >
> >
> > broadbandvoice at comcast.net wrote:
> >
> > > I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W -
> Speakerphone does
> > > not work with it. It works fine when you pick up the handset.
> Anyone
> > > experinced this problem before, the speaker works fine with
> Verizon
> > > line. The phone is behind a Linsys router RT31P2.
> > > Replace the batteries! Alkaline only, replace every 6 months
> > > 1.2.3 is also defective for other reasons. Upgrade
> > > Using a TDM400, an ATA or ??
> > > The phone works best with a 48V 20mA or better loop, so the
> FXS source
> > > voltage may have an effect, and this cheap phone has no
> previsions
> > > for external power.
> > >
> > > John Novack
> >
> >
> > _______________________________________________
> > --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
>
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Message: 9
Date: Sat, 22 Apr 2006 16:41:57 -0400
From: "Davi-Ann" <dthurn at tstt.net.tt>
Subject: Re: [Asterisk-Users] How can I get a recording from a CD to
myasterisk digital assistant
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <003501c6664d$32817e20$b4dbdbdb at infosys200>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=response
Is there any special encoding that I have to use?
----- Original Message -----
From: "Alberto Sagredo" <asagredo at peoplecall.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Saturday, April 22, 2006 4:15 PM
Subject: Re: [Asterisk-Users] How can I get a recording from a CD to
myasterisk digital assistant
> You will need them in one of asterisk supported formats.
>
> wav, slin,gsm, g729, g723...
>
> Davi-Ann escribió:
>> I got someone to record the messages we want for our auto-attendant menu
>> on a CD.
>>
>> All I have to do not is to upload the files into the asterisk box,
>> however the format is not recognized by the Asterisk box.
>>
>> Question 1) What formats should the sound file be, so I can upload it to
>> my asterisk box?
>>
>> Thanks
>> --Davi-Ann
>>
>> _______________________________________________
>> --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
>
------------------------------
Message: 10
Date: Sat, 22 Apr 2006 21:54:08 +0100
From: Wayne <Wayne at planetWayne.com>
Subject: Re: [Asterisk-Users] RE: SPA 3000 - UK Replacement
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <444A97F0.1090705 at planetWayne.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
tom wrote:
>
> You think that's bad, I ordered one on the 10th of march from redstore,
> that was showing a 3-5 day. They still haven't despatched the unit and I
> have been trying to call them now (on their 0870 number) for about a
> week, during the past 3 weeks I have been sending them email after email
> that hasn't been responded to.
>
Hiya!
I had that too with RedStore. The order tracking was saying for
absolutely AGES that it was waiting to come into stock. I did manage to
get to speak to someone and was assured that they were awaiting
delivery. Eventually (took about a month (or two??)) it turned up! - Had
it now up and running since March and works fine (after figuring out
that I needed Mod Taps to hook a phone into it to make it work!)
admittedly RedStore did give me the option to cancel the order - but I
stuck with it as it was nearly half the cost from anywhere else (about
£50). Like I say though - this was about 6-8 weeks or so ago since I
took delivery - I haven't checked to see if they are still selling them.
Wayne.
------------------------------
Message: 11
Date: Sat, 22 Apr 2006 22:01:13 +0100
From: Wayne <Wayne at planetWayne.com>
Subject: Re: [Asterisk-Users] Sipura SP3000 question
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <444A9999.3000405 at planetWayne.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hiyall,
I don't suppose anyone has the elusive 'administrators' manual for these
things - I've got the users manual but would still like the full suit so
to speak.
Cheers
Wayne.
------------------------------
Message: 12
Date: Sat, 22 Apr 2006 18:14:30 -0400
From: "Steven Totaro" <stotaro at bluehippo.com>
Subject: RE: Pinouts for T1/E1 crossover cable WAS "RE:
[Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ?"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<2C2595A0F39ADC4E84623DA5CC0DB7A41A2F41 at exchange.qfirst.com>
Content-Type: text/plain; charset="iso-8859-1"
The "telco guys" probably did something non-industry standard and reversed send and receive in the jack that they plugged the CAT5 into. Sure it works, sure it is easier, sure it is not the correct way of doing things.
Thanks,
Steve
________________________________
From: asterisk-users-bounces at lists.digium.com on behalf of Lacy Moore - Aspendora
Sent: Sat 4/22/2006 2:55 PM
To: Paul Mahler; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ?"
at&t (formerly SBC, formerly Southwestern Bell, formerly AT&T) just came out and installed my PRI. FYI, they used Cat 5e cable. No special T1 cabling that costs a fortune to buy somewhere, just plain old Cat 5e cable. Guess what guys? If they are using this as customers' sites, they are probably using it elsewhere. It's only as good as the weakest link, so you can go out and spend lots of money on "T1 cable", or just use Cat 5e like the telco guys do.
--
Lacy Moore
Aspendora, Inc.
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Message: 13
Date: Sat, 22 Apr 2006 18:18:03 -0400
From: "Steven Totaro" <stotaro at bluehippo.com>
Subject: RE: Pinouts for T1/E1 crossover cable WAS "RE:
[Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<2C2595A0F39ADC4E84623DA5CC0DB7A41A2F42 at exchange.qfirst.com>
Content-Type: text/plain; charset="iso-8859-1"
I have used cross-connect wire from the spool to make T1 crossover cables with RJ45 ends. All that matters is that pin one goes to four and two goes to five on both ends.
________________________________
From: asterisk-users-bounces at lists.digium.com on behalf of Andrew
Sent: Sat 4/22/2006 2:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?"
Alexander Lopez wrote:
>I have not in my experience seen any problems with using a Good Quality
>Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you
>should be fine. As far as the shielding goes, I use UTP cables and
>Connectors all the time and some of my X-connects run over 100 feet
>
>
I have used cat-5 for everything communications. serial printers, dumb
terminals, DS1 and even 10/100 ethernet. :-) It's easier to have it
installed as a network jack and then use for whatever you need.
...
Andrew McRory
LinuxSystems
Tallahasse, FL
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
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To UNSUBSCRIBE or update options visit:
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Message: 14
Date: Sat, 22 Apr 2006 22:55:59 +0000
From: broadbandvoice at comcast.net
Subject: RE: [Asterisk-Users] Don't see my post
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<042220062255.26782.444AB47F000217B90000689E220700095308010B020E9B02 at comcast.net>
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Subject: RE: [Asterisk-Users] Don't see my post
Date: Wed, 19 Apr 2006 01:33:17 +0000
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Message: 15
Date: Sat, 22 Apr 2006 23:34:29 +0000
From: broadbandvoice at comcast.net
Subject: RE: [Asterisk-Users] How to restrict simultaneous phone
registrations
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<042220062334.2677.444ABD850006363400000A75220588617208010B020E9B02 at comcast.net>
Content-Type: text/plain; charset="us-ascii"
disable three-way calling, restric channels to one per call.
-------------- Original message --------------
From: "Bill Gibbs" <bgibbs at edurotech.com>
> I say just bill the user at extension 333 it's his responsibility to
> keep the login info private. If he disputes it, refund the first time
> then change the password to something really complicated then start
> billing him if it keeps happening after that!
>
> Bill
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bryan
> Mahin
> Sent: Wednesday, April 05, 2006 10:50 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] How to restrict simultaneous phone
> registrations
>
> :) I should rephrase my question. And included a bit more information on
> what I am trying to accomplish.
>
> Solution 1 (preferred)
>
> I am working on an asterisk installation where most end users will use
> softphones. If I am not able to lock down calling to one phone at a
> time, the end users will share their login information with friends,
> family, neighbors, and the some girl they meet on myspace.
>
> Currently, I am able to register two phones at separate locations with
> the same account on each phone and make concurrent calls.
>
> For example, If I login extension 333 at location A, and 333 at location
> B, simultaneous calls can be placed from both phones at the exact same
> time. I only want calls placed from extension 333 to work from either A
> or B not A and B concurrently.
>
> Here is my ideal solution. Location A wants to make a call, but location
> B has a call in progress. Location B has to either close their phone, or
> hang up before Location A can make the call.
>
>
> OR.. Solution 2. :)
> A way I can distinguish in my CDR the IP address or some other
> recognizable difference between the two locations when they make
> concurrent calls using the same extension. The complication here is; I
> can currently the log IP addresses, but as the end phones are on the
> internet, Nat'd, and I am using a siparator for traversal. As a result,
> my logs show the IP address of the siparator and I don't have any other
> data to distinguish the end phones.
>
> OR.. Solution 2.5
> One thought I've had is to send logs from the siparator to a syslog
> server, parse them, hunt for simultaneous calls placed by the same
> accounts from different locations, and bill the end users accordingly.
> But I really dislike this idea as no one likes to be hit with
> surcharges.
>
> Any help or insight is greatly appreciated.
>
> Thanks again,
> Bryan Mahin
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric
> "ManxPower" Wieling
> Sent: Wednesday, April 05, 2006 7:50 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] How to restrict simultaneous phone
> registrations
>
> Bryan Mahin wrote:
> > Hello all,
> >
> > I am looking for a way to restrict users from logging in two separate
> > phones with the same authorization name/password at the same time.
> > Meaning, I only want users to be able to place a call from one phone
> in
> > one location, but have the ability to move from computer to computer.
> > Has anyone found any sort of solution for this type scenario?
>
> This is a non-issue, because a second registration to the same account
> will override and previous registrations for that account.
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Message: 16
Date: Sun, 23 Apr 2006 00:42:57 +0000
From: broadbandvoice at comcast.net
Subject: RE: [Asterisk-Users] No DTMF
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<042320060042.3633.444ACD910003ECD900000E31220073483008010B020E9B02 at comcast.net>
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From: "Mark Edwards" <mark at switchnet.com.au>
Subject: RE: [Asterisk-Users] No DTMF
Date: Thu, 9 Mar 2006 04:47:59 +0000
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Message: 17
Date: Sat, 22 Apr 2006 21:25:38 -0400
From: "Steven Totaro" <stotaro at bluehippo.com>
Subject: RE: [Asterisk-Users] Don't see my post
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<2C2595A0F39ADC4E84623DA5CC0DB7A41A2F45 at exchange.qfirst.com>
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Also, this is really a biz list question.
________________________________
From: asterisk-users-bounces at lists.digium.com on behalf of broadbandvoice at comcast.net
Sent: Sat 4/22/2006 6:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Don't see my post
Gafachi can, I've been using them with you problems.
-------------- Original message --------------
From: <billy at kersting.com>
First of all, try sending it to the asterisk-biz list.
________________________________
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John Rich
Sent: Monday, April 17, 2006 10:53 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Don't see my post
Hi Folks,
I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes.
Thanks
John.
________________________________
Yahoo! Messenger with Voice. Make PC-to-Phone Calls <http://us.rd.yahoo.com/mail_us/taglines/postman1/*http:/us.rd.yahoo.com/evt=39663/*http:/voice.yahoo.com> to the US (and 30+ countries) for 2¢/min or less.
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