On Wed, 2004-11-03 at 18:45, Hermann Wecke wrote:
> I couldn't think any recipe to reject a call if no callerID is presented.
>
> PrivacyManager and Zapateller are not an option, as the call will be
> answered before I can drop it. I just want to "silent drop" the call: no
> callerID, no answer.
On Wed, 2004-11-03 at 10:21, Matthew Boehm wrote:
> I've added all our SIP extensions to gastman but when someone makes a call
> to another extension, a new icon gets created. Why doesn't gastman use the
> icon that is already there? The existing icon turns green so I know that
> gastman knows its
On Tue, 2004-11-02 at 21:33, Jean-Michel Hiver wrote:
> >Actually, I assume the above (2 x IAX devices behind a single NAT
> >router) would work perfectly without any special configuration EXCEPT in
> >the (perhaps most common case) where both IAX devices are talking to the
> >same IAX server.
> >
On Tue, 2004-11-02 at 14:26, Mark Phillips wrote:
> I have a brand new HP Proliant DL380 with dual Xeon 3.06 processors, 2gb
> RAM running RHEL. It has a TDM410P installed.
>
I'm not familiar with the TDM410P, did you mean a TE410P or a TDMxxB
card?
If you could re-post your question, then we ma
On Tue, 2004-11-02 at 12:46, Karl J. Vesterling wrote:
> At 06:51 PM 11/1/2004, you wrote:
> [snip for brevity[
> >
> > So the U.S. Govt has never used linux anywhere? Wow.
>
> Not in most installations, and definitely not in DoD facilities.
> The "Office of Inspector General" has deemed open so
On Tue, 2004-11-02 at 20:57, Benjamin on Asterisk Mailing Lists wrote:
> On Tue, 02 Nov 2004 13:50:07 +0400, Jean-Michel Hiver
> <[EMAIL PROTECTED]> wrote:
> > Out of interest, how would this work in a situation where two IAX
> > compliant devices (i.e. IAXy) are behind a non-configured natted netw
On Fri, 2004-10-29 at 14:10, Glenn Powers wrote:
> Why not just use voip-info.org for this? It already seems to be the site
> for all things asterisk. Is there a problem with many people
> contributing dialplans, chunchs of dialplans, or AGI scripts to it?
> "Forking" is just as bad on the doc s
On Thu, 2004-10-28 at 23:34, Eric Wieling wrote:
> Alessio Focardi wrote:
> > Hi all !
> >
> > I played around for a few hours with a polycom 500 phone and it seems me that the
> > dtmf
> > mode is not configurable, looks like it only has inband mode.
> Polycom IP phones support RFC2833.
>
> I d
On Thu, 2004-10-28 at 04:15, Michael Bielicki wrote:
> BBversion: ?
You are right, I'll have it changed to Asterisk Version tomorrow...
Regards,
Adam
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On Thu, 2004-10-28 at 04:23, Steve Kann wrote:
> Adam Goryachev wrote:
> > On Wed, 2004-10-27 at 13:37, Jim Van Meggelen wrote:
> I think that it's hard to reach a critical mass on a project like
> this. The way I see it, there's 4 places people will probably look
>
On Wed, 2004-10-27 at 16:31, Kristian Kielhofner wrote:
> Richard wrote:
> > The default user name and password is a huge issue in some cases. For
> > example, hackers can get into the server, grab the configuration, program
> > their own phone and make free calls. Another example, if you have mult
On Wed, 2004-10-27 at 15:27, Jim Van Meggelen wrote:
> [EMAIL PROTECTED] wrote:
> > See http://www.websitemanagers.com.au/asterisk/
> > I have run/maintained www.deadcat.net for a long time (6 or
> > more years I
> > think) which is basically the same sort of thing for Big
> > Brother (cross platfo
On Wed, 2004-10-27 at 06:50, Andrew Kohlsmith wrote:
> On October 26, 2004 04:15 pm, Richard Lyman wrote:
> > the issue i think is being discussed is when all participants can
> > talk. if you were simply retransmitting client side muted then
> > you could fit alot of listeners amoung one or more
On Wed, 2004-10-27 at 13:37, Jim Van Meggelen wrote:
> People will want to pay for your expertise because you wrote (or at
> least contributed to) the base platform, or language, or what-have-you.
> The more one contributes, the more their credibility is established --
> their services gain value.
On Mon, 2004-10-25 at 14:05, Paul Hales wrote:
> First you need to set up the hint function in extensions.conf:
>
> exten => 6003,hint,SIP/6003
>
> Then set the SNOM's buttons (FUNCTION KEYS) to be speed dials (DESTINATION)
> to the extensions in question.
Has anyone managed to do this with a po
Just a heads-up that asterisk is getting a mention in spam now... oh,
and make sure you NEVER EVER buy anything from this company.
[SNIP]
NEWS: VocalScape Inc. Announces DELETED for Asterisk IP PBX Users. You
can start earning money NOW ? without waiting for the fabled
stock-market recovery. How?
On Thu, 2004-10-14 at 23:10, Alex Barnes wrote:
> Hi all,
>
> I am currently trying to decide what Operating System is best to go
> for on a customer site. Server will only be running Asterisk / MySQL
> / Apache / PHP but nothing else.
>
> I have only tested Asterisk on SLES 8.1 however I do h
On Thu, 2004-10-14 at 20:09, Dido Sevilla wrote:
> On Wed, 13 Oct 2004 20:46:48 -0600, Darren Wiebe <[EMAIL PROTECTED]> wrote:
> > I have taken the astcc program which is designed for calling cards and
> > used it to create a very basic post-pay system. This allows your users
> > to make multiple
On Thu, 2004-10-14 at 11:08, Ed DeHart wrote:
> When you call my system your call is handled by the auto
> attendant. It works fine with one little problem. In addition
> to being able to dial any extension during the announcement, you
> can dial a telephone number. The system will bridge the Za
On Wed, 2004-10-13 at 17:00, Vladyslav wrote:
> Hi All.
> How to receive multiple pages with rxfax ?
>
> Here is what I have:
> exten => 10,1,Setvar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
> exten => 10,2,Setvar([EMAIL PROTECTED])
> exten => 10,3,rxfax(${FAXFILE})
> exten => 10,4,system(/
On Wed, 2004-10-13 at 15:06, James Bean wrote:
> Sorry, I explained this wrong.
>
> I am wanting the callerid of the incoming caller from my
> analogue line on the TDM400P to be passed TO the sip
> phone so the sip phone display shows the phone number of
> the incoming caler from the call on the
On Tue, 2004-10-12 at 04:11, Barton Hodges wrote:
> Trying again with a different subject...
>
> Currently, if I briefly press the flash hook on my phone, the caller
> is placed on hold. I would like for the channel to hangup if I do
> this instead, never placing a caller on hold (I'll be using
>
> Wow, that's a really sucky attitude. I would expect *Digium* to tell him
> to go away and solve his own problems. However, if the user community does
> that, then this is one of the suckiest user communities I've run across in
> the free software world, and I've been doing free software for ma
On Sat, 2004-09-25 at 09:49, Michael Bielicki wrote:
> 64bit it :)
>
> [EMAIL PROTECTED] root]# cat /proc/cpuinfo
> processor : 0
> vendor_id : AuthenticAMD
> cpu family : 15
> model : 5
> model name : AMD Opteron(tm) Processor 244
Any idea to the number of channel
On Fri, 2004-09-24 at 01:12, Kevin Walsh wrote:
> I seem to be hoarding patches, and sending them out on request. I should
> set up a website to list and share them more easily.
I did, but nobody used it... http://www.websitemanagers.com.au/asterisk/
Anyone can register and upload *their* files,
On Thu, 2004-09-23 at 15:34, el Flynn wrote:
> Lenny Tropiano / asterisk.org Mailing list wrote:
> > These taken tonight (9/22/2004) at the Expo and Reception
> > Enjoy. http://photos.tropiano.org/gallery/astricon-2004
> >
> > Lenny
>
> Anyone knows if those Snom Keypad 220s are available, and w
On Wed, 2004-09-22 at 17:13, duncan hall wrote:
> Hi,
>
> I am currently trying to find a replacement for a dinosaur PBX and want
> to replace it with a VoIP solution.
>
> We have just moved our lines over to an Optus Multiline from a Telstra
> ISDN Onramp 30 service with 100 lines.
>
> My que
On Tue, 2004-09-21 at 23:49, Matthew Boehm wrote:
> Not to flame a respond, but I only count 13 lines, not 200.
That is 13 lines that *I* quoted from your email, you quoted a lot more
from the previous email (the entire email in fact)
> Anyway, what you posted is exactly what I am trying to preve
On Tue, 2004-09-21 at 11:52, Raul Elizondo (wizardteam) wrote:
> Hi,
>
> I am a little bit confused about how extensions.conf recognice a fax tone.
> For testing i got this in my home pbx:
>
> [opciones]
> exten => s,1,wait(1)
> exten => s,2,Background(new/opciones)
> exten => 1,1,Goto(explicacio
On Tue, 2004-09-21 at 08:14, Matthew Boehm wrote:
> OK. So I removed all the callerid= from the sip.conf and Wiley's fix works
> perefectly. But I am back to where if I call out, the caller id shows up as
> my extension only.
>
> My fix, that didn't work:
>
> [global-outgoing]
> exten => s,1,Set
On Thu, 2004-09-16 at 12:41, Arinze Izukanne wrote:
> Can't locate asterisk/AGI.pm in @INC (@INC contains:
Looks like you are missing asterisk/AGI.pm which is most likely from the
Asterisk-perl package. <20 seconds on google gets me this:
http://asterisk.gnuinter.net/
Regards,
Adam
On Thu, 2004-09-16 at 12:41, Arinze Izukanne wrote:
> Can't locate asterisk/AGI.pm in @INC (@INC contains:
Looks like you are missing asterisk/AGI.pm which is most likely from the
Asterisk-perl package. <20 seconds on google gets me this:
http://asterisk.gnuinter.net/
Regards,
Adam
On Wed, 2004-09-15 at 22:37, Senad Jordanovic wrote:
> >>
> >> make a simple web interface.. where user logs in..
> >> interface tells a script on * server about users
> >> location/extension/device. then your script will re-create TFTP
> >> files, sends
> >> reboot to 7940/7792 and you are done
Just wondering if anyone has gotten instant messaging working between
two polycom phones/and or MSN messenger with asterisk in the middle? I
tried a few things but couldn't get it to work...
Regards,
Adam
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On Wed, 2004-09-15 at 04:29, Lee Howard wrote:
> On 2004.09.14 11:10 Marty Mastera wrote:
> > 2)Packet loss, etc...makes faxing over the internet unreliable
>
> I'm not sold on this theory yet. I don't think that it's so much a
> matter of packet loss (this shouldn't occur regularly), but ra
On Tue, 2004-09-14 at 13:56, bagattin jerome wrote:
> Hi,
>
> Another question about read command:
> Howto sup file option and keep maxdigits options ?
> exten => 3,1,Read(ILE,1)
> give me :
> Unable to open 1 (format UNKN): No such file or
> directory :-(
>
This one is easy, just do:
exten
On Tue, 2004-09-14 at 10:34, Phil Stevens wrote:
> Hello,
>
> I'm researching the possibility of using VOIP (SIP) with an existing
> PABX system. Ideally, the setup would be to dial an outside line through
> the PABX (that would actually link to the the VOIP gateway).
>
> At this point I would pr
On Sat, 2004-09-11 at 14:25, Kevin P. Fleming wrote:
> Adam Goryachev wrote:
>
> > PS, in case you are wondering, I (and my supplier) have spent hours
> > looking at different motherboard specs, and so far haven't been able to
> > find anything suitable (except a dual
On Sat, 2004-09-11 at 13:37, Scott Stingel wrote:
> Hi Adam-
>
> I'm wondering if the TE405P might be a better choice, since it's 5 volt PCI
> and may allow you to consider a wider selection of motherboards. Sounds
> like you may not need the latest and fastest motherboards, which often use
> the
Hi all,
I'm looking for a new system which will use the TE410P. Originally I was
going to use a dual Athlon MP system, but my supplier tells me these are
being phased out now, and so will be difficult to find replacement parts
later.
So, I am looking for suggestions of suitable motherboards with
On Thu, 2004-09-09 at 21:38, HengWee Chin wrote:
> Hi Umar,
>
> unfortunately I have not found a solution for my problem. I do not think
> that there is any problem in the dial plan. The IVR that I have is not done
> using asterisk. It is another application running on another machine with a
don't use it, is if you can't afford
it. AFAICT, the second best solution is to use chan_capi/zaphfc with
supported BRI cards.
Just my 0.02c worth
Regards,
Adam
--
--
Adam Goryachev
Website Managers
Ph: +61 2 9345 4395[EMAIL PROTECTED]
F
On Wed, 2004-09-08 at 07:52, Kris Boutilier wrote:
> I'm using a T100p to interface to a channel bank and from there to analog
> PSTN lines. Because of my particular setup I have to do post-connect inband
> DTMF dialing - which takes up to 5 seconds for a 10 digit number, assuming
> 0.5/sec per di
DAA mode is 'FCC'
Found a Wildcard FXO: Wildcard X101P
Registered tone zone 1 (Australia)
I haven't actually had anything quite like this before, usually asterisk
'just works' ...
Thanks,
Adam
On Wed, 2004-09-08 at 00:21, Adam Goryachev wrote:
> Well, it has be
Well, it has been a long time since I used my NetJet S cards with
asterisk (moved to the quad PRI card) but I am trying to get this
working again for home use.
Basically, what I am using is this:
Linux 2.6.8.1:
config lines:
# ISDN subsystem
CONFIG_ISDN=y
# Old ISDN4Linux
CONFIG_ISDN_I4L=y
# CONFI
On Tue, 2004-09-07 at 21:20, John Howard wrote:
> I'm using the Zyxel 660H-61 with fantastic results.
>
> It supports SIP out of the box, and ive been able to set up the bandwidth
> shaping for SIP (it supports this natively), and iax2 as well.
>
> It cost be about Â60 GBP.
>
> Well worth the mo
On Wed, 2004-09-01 at 04:39, Deon Rodden wrote:
> All of my phones use sip, their accounts are in the sip.conf file and
> they have the context of 'company' or whatever. These phones need to be
> able to call each others extension, as well as dial outside to the real
> world. So in that context
On Tue, 2004-08-31 at 21:42, Atif Rasheed wrote:
> this is from my extensions.conf, the first three patterns are for
> toll-free numbers, and fourth pattern is for other numbers, where an AGI
> is called for authentication.
> now when I dial 011448000664327 if falls into the fourth pattern, where
> - Original Message -
> From: "Greg Hulands" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <[EMAIL PROTECTED]>
> Sent: Thursday, August 26, 2004 9:31 PM
> Subject: Re: [Asterisk-Users] TDM400P Problems
>
>
> > Even when the tdm400 is the only card
On Wed, 2004-08-25 at 10:00, Robert Rozman wrote:
> Hi,
>
> I'm writting some Perl AGI scripts. I got lucky on some working, but try to
> debug other.
>
> I put a lot of verbose() or print STDERR commands into them, but no output
> appears on asterisk CLI (I run asterisk -vvvrgc after normal
On Wed, 2004-08-25 at 03:02, Chris Shaw wrote:
> Check out my ATA idea though, with a regular cheap analog doorphone and a
> HTX86 or even Sipura, you can program the ATA to dial an extension as soon
> as the button on the intercom is pressed and then with some extension logic
> you can do neat thi
On Mon, 2004-08-23 at 15:53, Imran Akbar wrote:
> Thanks,
> I seem to have done the zaptel installation - what am I missing - i
> still don't have a chan_zap.so file?
>
In the asterisk src directory, do make clean install
It won't even build the zaptel stuff unless you have installed zaptel
On Mon, 2004-08-23 at 13:02, [EMAIL PROTECTED] wrote:
> I am writing a call center application.
> I do not want to use Queues to manage my incoming calls and connect
> them to the operators for a few reasons which I wont go into here.
It would be interesting to see why this wouldn't work for you..
On Fri, 2004-08-20 at 14:21, Shaun Ewing wrote:
> Hello all,
>
> I was wondering if anybody knows where one might obtain a PCI ISDN
> card supporting a single BRI for use with Asterisk in Australia (and
> using something like chan_capi).
www.atp.org.au have been nothing but helpful and knowledgea
On Fri, 2004-08-20 at 09:42, Lee Allen wrote:
> Okay, now my script...
>
> It creates an outoing call in /var/spool/asterisk/outgoing, pulling
> information from a database (assuming I learn some perl and mysql, or
> something!)
> THAT file (outgoing call queue) would have to...
> - call the given
On Thu, 2004-08-19 at 18:26, lenz wrote:
> I updated the file at the same URL -
> http://demo.xcept.it/xc-ast/XC-AST.htm - with a new version that I tested
> with Firefox too.
> There are new statistics available that show agent's averages and activity
> and lets you analyze more than a queue
On Thu, 2004-08-19 at 17:42, Nicola Murino wrote:
> I would like to have some more informations on asterisk,
Read more of the mailing list, more of the various websites like
www.digium.com and www.voip-info.org
> I want to setup a linux based pbx and asterisk seems to be the best solution,
Of c
On Thu, 2004-08-19 at 07:01, Seth Remington wrote:
> On Wed, 2004-08-18 at 14:45, David Filion wrote:
> > Does anyone know of an alternate source for spandsp? opencall.org is
> > down and all the links returned by Google just point to the dead site.
> >
> > Thanks
> > David Filion
>
> I threw a
> For those that have been selling telecomm for awhile, its fairly well known
> the business purchasing decision is based primarily on "cost" followed by
> "features". Its also fairly well understood that many businesses will
> list a feature or two as "required" to ensure their favorite vendor is
On Fri, 2004-08-13 at 22:06, Craig Guy wrote:
> Hi,
>
> now that these cards have approval in Australia, has anyone had any luck in
> connecting them to a Telstra Onramp 10 service? Are these configured as a
> PRI, bchan=1-10, dchan=16 in zaptel.conf and switchtype=euroisdn in
> zapata.conf? Bet
On Fri, 2004-08-13 at 01:38, Dana Nowell wrote:
> This is interesting but did you check space consumption? On my system it
> appears that each directory is about 4K bytes. One million phone numbers
> at 10 digits (or 1 number) is 10 million directories (or one million
> 'number' directories) at 4
On Thu, 2004-08-12 at 22:49, Rich Adamson wrote:
> What you're looking for is key system functionality, and asterisk is a
> pbx. Fully understand that many of the competitive products cross the
> old line between key systems and pbx's by providing the most often
> wanted features in a single box. N
On Thu, 2004-08-12 at 20:17, Steven Critchfield wrote:
> On Wed, 2004-08-11 at 23:53, Adam Goryachev wrote:
> > On Wed, 2004-08-11 at 20:42, Steven Critchfield wrote:
> > >
> > > So before you hamstring your small office into having unnecessary
> > > growing p
On Wed, 2004-08-11 at 20:42, Steven Critchfield wrote:
> On Wed, 2004-08-11 at 06:17, Jeremy Lowery wrote:
> > I am currently a new asterisk user and new to telephony in general. I
> > have been looking around to implement a solution with asterisk that has
> > many of the nice features of a propr
On Wed, 2004-08-11 at 21:09, Ben Merrills wrote:
> Hi,
> I have a problem with a Digium quad E1 card. It seems when I make
> outgoing calls to any party, when that person talks on the line, they
> hear scratching and static (thereâs also background static, but less
> of it). The person making the c
On Thu, 2004-08-12 at 03:35, Steven Critchfield wrote:
> On Wed, 2004-08-11 at 11:32, Chris Shaw wrote:
> > That is a matter of opinion and not in any way factual SQL, just as
> > everything else, is as secure as YOU make it... As you said, it's a language
> > for querying relational databases,
Can anyone say, with the current state of asterisk (CVS version, or any
other older version), using any combination of hardware, is it possible
to display the status of one extension onto another telephone?
ie, say I am extension 510 and have a polycom IP600 with 6 lines, and
associated lights. Wi
On Wed, 2004-08-04 at 01:46, john lawler wrote:
> > Finally, can I turn off the '#' to transfer, since we're using the
> > hook-flash (albeit manually) instead? ISTR an option to do this but have
> > spent the morning trying to find it again unsucessfully...
>
> I think you might want to look at t
On Tue, 2004-08-03 at 21:30, Christopher Lee wrote:
> The only change I believe I had to make was under
>
> /usr/src/asterisk/channels/chan_zap.c
>
> #define DEFAULT_CIDRINGS 2
>
> The default is 1
>
This is one of 2 patches I make to asterisk every time I download. It is
needed to make the c
Gee, I think the subject says the right thing.
I've looked at the wiki at the various phones available, and basically I
think what I am after is one of the following phones:
Polycom
Aastra 480i
Zulty's Zip 4x4
Although after just reading through the review linked from the wiki, and
the polycom we
On Thu, 2004-07-22 at 23:16, Deon Rodden wrote:
> What do you mean does not support? I mean, it connected to the system fine
> and can place and receive calls.
No, I meant it will work, but won't be work optimally. (ie, probably you
will have bad echo).
> As far a legal issues, that would figure.
I've recently enabled monitoring (recording) of incoming calls that
arrive in the queue (all calls come in through the queue) using the
config options in queues.conf.
However, it seems that as soon as the call is placed on
hold/transferred, the monitoring stops. I would like to know if it is
possi
On Wed, 2004-07-21 at 02:03, Deon Rodden wrote:
> I installed a server in Australia with a Wildcard X100P in it. From my
> server in the U.S, I pushed a call via IAX to the server in Australia which
> then pushed it out that card. Severe echo, only I could hear it though. The
> remote side heard no
On Tue, 2004-07-13 at 12:50, jurgen wrote:
> Hi all,
>
> I've been lurking here and reading the Wiki for a month or so now,
> getting information on the suitability of Asterisk for my
> installation.
Great start...
> I'm responsible for the technical stuff at a mostly-inbound call
> centre in Me
On Wed, 2004-07-14 at 05:19, Zarjazz wrote:
> Basically I have a few people on the * box using the current agent /
> queue system where I create a call file into the spool directory so they
> can make outbound calls to customers from a simple web page. But now I
> need to add the feature of transfe
On Wed, 2004-07-14 at 00:42, Adam Goryachev wrote:
> In either case, I would suggest you discuss your options with the folks
> at www.atp.org.au, I've found them to be quite helpful, and definitely
> quite knowledgeable
PS, most digium resellers seem to follow the standard di
On Tue, 2004-07-13 at 12:28, Kimble Young wrote:
> David,
>
> If you go the analogue route:
>
> * You'll get poor audio compared to ISDN which is crystal.
> * Each number will act like a seperate line unlike with an ISDN card where
> you can receive two calls simultaneously on the same line.
Act
> 3) When I try to dial by generating a call file in the proper outbound call
> directory, I still get stuck on the dial command.
>
>
> Any ideas? Am I just not understanding something critical?
>
>
> Thanks for any help! I've search the archives and the WIKI for
On Sat, 2004-06-19 at 19:27, Storer, Darren wrote:
> Hi Kevin,
>
> KW> By the way, it's useful to map 911 and 112 onto your 999
> KW> route for the benefit of foreigners who don't know any better.
>
Well, while you are at it, you might as well add-in 000, because that is
what we use.
(BTW, why
On Mon, 2004-06-21 at 11:40, Nik Martin wrote:
> Adam Goryachev wrote:
> > On Sat, 2004-06-19 at 06:13, Stephen R. Besch wrote:
> >
> >
> >>So, if someone could brief me on the GPL issue, and (perhaps someone
> >>else) offer a distribution point, it
On Sat, 2004-06-19 at 14:10, Randy Bush wrote:
> i am using the freebsd port, which seems to not yet have WaitExten(),
> which i kinda want to use thusly
>
> [ext-666]
> exten => _.,1,SetVar(areacode=666)
> exten => _.,2,Background(zz-in-who) ; give them list of extns
> exten => _
I like asterisk, I use it, I try to contribute where I can.
OK, I'll shut up now.
--
Adam Goryachev
Website Managers
Ph: +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au
__
On Thu, 2004-06-17 at 04:45, Michael Sandee wrote:
> >
> >
> >Am I dreaming?
> >
> Yes.
>
> Community based development is too unreliable.
>
> Just to refer to ongoing projects... Look at the farfon
> (www.farfon.com), It's an active project in the final stages of development.
> It offers the be
On Wed, 2004-06-16 at 04:35, Robert Withrow wrote:
> I've searched the lists but I didn't find anything exactly like this.
>
> I have two Grandstream BT101 phones connected to an Asterisk.
> Periodically, for reasons that I can't determine, one or the other (or
> both) of the BT101s decide(s) to
Look at show application queue
It includes a timeout option, which after the call is not answered for
some period of time will drop back to the dialplan and hence your
voicemail.
Regards,
Adam
On Tue, 2004-06-15 at 18:35, Matt wrote:
> Hi all,
>
> I'm stuggling with how to present calleds to a s
Has anyone considered some or all of the following possibilities:
A) The rates change so frequantly that by the time they make it to the
website, they would be out of date?
B) Maybe they have different rate sheets for different people (based on
number of minutes etc being bought)
C) Maybe the re
On Fri, 2004-06-11 at 05:59, Nik Martin wrote:
> Can the Manager Redirect command transfer a ringing SIP extension? I'm
> trying to implement a Camp On feature, and having failed to do it in Dial
> Plan logic, am trying to do it with manager logic. If an arbitrary Sip
> extension is ringing, I ne
I've been avoiding commenting on this thread because I haven't studied
the code enough, or my current problem, but anyway, here is my 0.02c
worth...
I found the documentation to be OK, and the app seems to do some
fantastic things, which the current call parking can't do. However, the
real reason
On Thu, 2004-06-10 at 23:27, Rich Adamson wrote:
> This email is intended to document an issue for anyone searching the archives.
>
> We had a problem yesterday with _all_ iax2 and sip sessions; no reasonable
> conversation could be established due to extremely choppy audio in one
> direction only
Also check the first line of the file points to your location of perl.
Usually /usr/bin/perl or /usr/local/bin/perl
PS, "chmod 755 makering.pl" to make it executable.
Regards,
Adam
On Tue, 2004-06-08 at 18:52, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
> Simon <[EMAIL PROTECTED]>
> -- Executing Dial("SIP/PHONE2-d557", "Modem/ttyI0:v0413xx") in
> new stack
The only thing I can see which is a bit strange is this, why do you have
a v in the dial string??
It has been a while since I used the traverse card, but I never needed
to use the v in front of the number.
Try
On Mon, 2004-06-07 at 13:27, Randy Bush wrote:
> > Exactly my point, by ***DEFAULT*** Asterisk won't use SRV records,
>
> is this a feature or a bug?
A feature of course, IMHO, nothing should be enabled by default. Let's
not do a MS Windows product...
In fact, I think it would be nice if all mod
On Thu, 2004-06-03 at 11:40, Tony Hoyle wrote:
> Adam Goryachev wrote:
> > Well, actually they are. Sure, for $20 you can buy an analog phone, for
> > $150 you can buy a grandstream, big difference. However, for a PBX class
> > telephone, you are looking at prices > $500
On Thu, 2004-06-03 at 10:21, Tony Hoyle wrote:
> John Fraizer wrote:
> > your desk has to be able to TELL the PBX you want to transfer a call.
>
> No it doesn't there's a universal (?) standard for this - hit recall, dial new
> number. Heck it even works on PSTN lines if you pay for the right s
On Wed, 2004-06-02 at 12:45, Brian D'Arcy wrote:
> Hi Adam,
>
> I appreciate your feedback, and understand totally where you're coming
> from as far as the database perspective goes.
>
> For the first "draft" of the app, I think I'm going to let it default to
> using a conf file for two reasons.
Does anyone know what VoIP phones are available in australia for
'decent' prices (ie, decent for me is under $250 - $300 max)
So far I've found the following:
D-Link DPH-100S $620
Various Cisco phones at > $500 each
ZyXEL Prestige 2000W$380
Haven't been able to find much else around.
On Wed, 2004-06-02 at 06:14, Brian D'Arcy wrote:
> Hello all,
> Have a .conf file (findme.conf?) which contains multiple contexts, each
> context's name should match the naming convention used with sip, or
> iax.conf. For example, if I have [bdarcy] as one of my sip peer
> entries, in findme.conf
Hmmm, perhaps I am the only one who doesn't trust their telco (I doubt
it) but...
I have the rates that I currently pay my telco, and would like to
extract my CDR's and add an additional field displaying the actual price
paid for the call. I would like to do this based on destination phone
number,
On Tue, 2004-06-01 at 11:37, Andrew Kohlsmith wrote:
> Anyone? I can't believe something this simple should be giving this much
> trouble...
>
> -A.
>
> On Friday 28 May 2004 12:17, Andrew Kohlsmith wrote:
> > -- Starting simple switch on 'Zap/2-1'
> > [ I dial *8# here and get fast busy ]
On Fri, 2004-05-28 at 10:36, Adam Hart wrote:
> cts-au.freshtel.net sorry, it's hosted at comindico in sydney.
Nicer...
I get over 100 pings
min 14ms
avg 34ms
max 234ms
with one packet dropped.
(bad my end, I have bursty traffic for SMTP/POP server)
I suppose I could do QoS on outbound, which sh
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