On April 1, 2009 01:32:18 pm Jason Aarons (US) wrote:
> I don't think a off the shelf modem has the necessary DSPs to convert
> voice to codecthat is why a Voice Gateway/Analog Telephony Adapter
> or FXO/FXS cards exist instead of modem having a second life.
There are no DSPs in any of the tel
On March 30, 2009 12:48:59 pm randulo wrote:
> Except for roaming and in particular international roaming, isn't the
> best plan to forward calls the iPhone. It is a phone, too isn't it? Or
> just a game platform, browser and GPS?
That's pretty much what I do; I use siax (I have a jailbroken iPhon
On January 2, 2009 01:44:14 pm David wrote:
> 2007
>
> 2006
> ====
> Andrew Kohlsmith 290
> 2005
> ====
> Andrew Kohlsmith 731
Damn... I'm slipping! 2nd place in 2005.
-A.
___
-- Bandwidth and Colocation Provided
On December 17, 2008 06:59:19 pm David fire wrote:
> you are soamming my mail box whit this useless discution
> the solution is doble posting (top and bottom)
It's a public mailing list. If you're having trouble managing it, you may
want to try a digest version, or perhaps a moderated list.
-A.
On December 17, 2008 05:03:00 pm Eric "ManxPower" Wieling wrote:
> To me top posting is like people talking about "SIP Trunks". There is
> no such thing as a "SIP Trunk". There are SIP connections, peers,
> friends, etc. The term is simply a marketing buzzword to make people
> that don't know mu
On December 4, 2008 08:31:40 pm Matt Gibson wrote:
> 1st place: An APSTel dial plan (professional license) donated by -- you
> guessed it - APSTel!
> 2nd place: An Aastra 57I IP telephone donated by Ottawa Phone Systems and
> Flewid Inc!
> 3rd place: An APSTel dial plan (standard license) donated b
On December 4, 2008 02:14:52 pm Erik (Caneris) wrote:
> Thanks. Unfortunately no SIP/IAX access at this time, only by dialing one
> of the TNs. However, I'll bring it up with the client and see if they'd
> want us to configure that.
Definitely would be cool, you don't lose any ad revenue and I don
On December 2, 2008 07:55:00 pm Erik (Caneris) wrote:
> Nuance would say "no" :)
> I'd say "maybe". Call up +14164854854, it's a recent project we did for a
That's pretty cool! Is there any SIP or IAX access to this (aside from
dialing a POTS number) ?
-A.
_
On December 1, 2008 07:21:33 pm Doug wrote:
> Hmmm. When our users are pounding the network
> with BitTorrent traffic, we just shut them down
> and wait for them to complain. It's against our
> Acceptable Use Policy, and causes all sorts of
> VOIP headaches.
As someone who is the technical lead
On October 29, 2008 10:19:36 am Bill Michaelson wrote:
> I'm wondering how prevalent the practice of physically segregating voice
> and data networks is in the Real World.
>
> What are the factors that typically lead to such a decision?
> DIscussions of pros and cons are most welcome by me.
>
> Exp
On October 28, 2008 12:58:25 pm JD wrote:
> The folks that devloped the fax V.protocols took into acount typical
> copper problems like noise or echo. But what they never conceived of as
> even being possible is that a call might shift around in the time
> domain. Thanks to jitter/latency, the de
On October 27, 2008 02:01:43 pm Jeff LaCoursiere wrote:
> Speaking of fring, I just got my brand new iphone 3G. Anyone have any
> comments on how well fring or any other sip client (siphon?) works on
> iphone?
I do not like fring. It's buggy, it's unstable, it looks "goofy" -- but I
have to say
On October 5, 2008 12:22:37 pm Philipp Kempgen wrote:
> Thunderbird could probably render his text/html part just fine but
> I don't want it to. (Nothing is wrong with preferring text/plain in
> the MUA.)
> Thus it renders his text/plain part which lacks line breaks.
> I posted some links to the li
On October 3, 2008 08:56:34 pm Philipp Kempgen wrote:
> I could live with 1 or maybe 2 of these issues but 5 is a bit
> much. You didn't even notice these problems, so, ok, sorry for
> being rude. But for people who are used to email in ages it feels
> like a punch in the face. It's a real culture
On October 3, 2008 04:15:26 pm Tariq .. wrote:
> it is FRING i'm sorry for the mistype...
> www.fring.com
I just downloaded it for the iphone... it's pretty cheap looking, crashes
occasionally and appears to force all audio through their server, but I have
to say that yes, it does have potential
On September 25, 2008 10:41:45 am Drew Gibson wrote:
> Once CDMA has gone the way of the dodo in North America, I really will
> miss one of my favourite scenes:-
>
> Visiting Brit steps off plane and checks phone for messages...
>
> Puzzled look appears as they ask "Why doesn't my phone work? It wo
On September 25, 2008 09:01:52 am Dean Collins wrote:
> Yep you got it world coverage includes all the countries of the
> world like USA, Canada and Mexico, and not something like USA and 212
> other countries globally.
>
> BTW I hear that Iraq also now uses CDMA (some senator shoe-horned it
>
Good morning,
I have a Bell Canada PRI here (switchtype=national) and I am trying to perform
a call-forward-unconditional on one of the DIDs.
The idea is that when DID 5551234 receives a call, Asterisk redirects it back
out the same PRI to some external number.
This is simple enough to do with
On August 23, 2008 07:57:33 pm Alex Balashov wrote:
> Yes, indeed. Encapsulation protocols such as IPSec/GRE won't work at
> all over high RTT latency (>= 400 ms).
Why not? Is there some kind of timing involved in encapsulating data that I'm
not aware of?
-A.
___
On August 11, 2008 06:59:23 pm JR Richardson wrote:
> So my question is this: Can I setup Asterisk to only allow t.38 pass
> through from these ATA's, without the need to use the #99 in every dial
> string from the fax machine?
Can you use disallow/allow with UDPTL? I'm not sure, I've never play
On July 24, 2008 04:42:42 pm David Cook wrote:
> Have the Norstar programmer send all 3 digit, unused extensions to the PRI.
> Then Asterisk will see 221, etc. and can handle at your dialplan sees fit.
Yes, this works, but you won't be able to treat those as regular extensions;
the Nortel will tr
On July 17, 2008 11:44:07 am Dean Collins wrote:
> 1/ R&D costs v's number of units manafactured per annum.
That's bullshit; There are many more office phones than office desktops out
there, and the research has been paid for many times over. Think of how long
the Meridian 1 has been around.
>
On June 23, 2008 08:08:53 am OCG Technical Support wrote:
> I little more digging and I confirmed that cell phone VM and FAX waiting
> icons are in fact controlled by a proprietary SMS message format. Here's
> what I found:
Yes; this is the same sticking point I hit; you can't use an SMS email
g
On June 17, 2008 01:45:43 am randulo wrote:
> The screwdriver is reversible, it swings both ways, pull out the shank
> and stick it in the other way, it becomes a Phillips. I'm tellin ya,
> there Digium engineers are good!
Most every pocket screwdriver that is sold as a promotional item is like th
On June 16, 2008 07:22:18 pm Mark Hamilton wrote:
> How come he has it, and he's in Paris! I'm in Toronto, and I don't have it?
Yeah, me too. I even got a mention in the book, but no screwdriver? :-(
-A.
___
-- Bandwidth and Colocation Provided by htt
On June 15, 2008 12:04:01 pm randulo wrote:
> Moving day, everything packed. Including tools! But wait, there in the
> jar with pens and pencils... it looks like. Yes, it's the Digium
> Asterisk tweaker!
>
> THANKS Digium!
>
> Before you ask, it's 1.0 I think.
?
-A.
_
On June 4, 2008 06:20:57 pm Joe Carroll wrote:
> Interestingly enough, on the syslog messages from the TNT we are seeing
> "Called = 911, Q850 Cause = 28, SIP Response = 484"
That really looks like the switch that the TNT is talking to is rejecting the
number, not the TNT...
-A.
___
On May 5, 2008 01:58:42 pm Tilghman Lesher wrote:
> > Hmm. Haven't found any Digium Stockholm office to discuss with ;-)
> That hasn't stopped any of the Canadian employees. :-)
That's because nothing stops Canadians, short of Hockey Night in Canada :-)
-A.
On May 4, 2008 08:40:10 pm Jay R. Ashworth wrote:
> > > Customer's insistence. We didn't have a choice, really.
> > Nothing wrong with that, it just adds more billable hours. :-)
> As long as it does.
I don't know about you, but whenever a customer wants me to do work and does
not want to follo
On May 4, 2008 07:24:45 pm Rob Hillis wrote:
> Customer's insistence. We didn't have a choice, really.
Nothing wrong with that, it just adds more billable hours. :-)
-A.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
aster
On May 2, 2008 03:13:40 pm Norman Franke wrote:
> enter the four-digit extension of the person you are trying to reach
I would suggest breaking that up
"Please enter the"
"digit"
"extension of the person you are trying to reach"
then you can use the individual numbers and fill in 2 digit, 3 digi
On May 1, 2008 11:39:52 am Tony Mountifield wrote:
> Does anyone know if the Digium PRI cards can be configured or modified
> to have a high-impedance input on the RX pair? I would be interested in
> this in order to build a bi-directional PRI audio sniffer using two
> E1/T1 ports per trunk to be m
On April 12, 2008 03:12:31 am Col Ferguson wrote:
> Hello,
> I have a system in a motel that needs call billing data output through its
> serial port so the existing motel management software can collect the call
> billing info.
> Is there any easy way to redirect the data that goes into the
> cdr_
On April 7, 2008 02:01:08 am Alex Balashov wrote:
> A Lucent TNT Max outfitted with _plethoric_ VFCs might work okay. Apex
> too, perhaps. Haven't tried to see how much it can handle when TDM->RTP
> translation is required.
I'm curious; are the cpu/tdm/dsp requirements for 672 g729 rtp streams t
On April 6, 2008 11:12:33 am Steve Totaro wrote:
> I cannot recommend the Adtran MX2800 M13, it has redundant everything
> and is very easy to setup and not very expensive either.
Agreed; I've set these up and they are rock effing solid. We did have a shelf
controller die and without the backup s
On March 25, 2008 02:15:42 pm Lacy Moore wrote:
> I think that is one of the biggest things that businesses overlook
> when switching to Voip. It's hard to get in the directories.
I have to say that it's been many years (well before voip) that I've gone to
the directories. Google and yellow pag
On March 24, 2008 02:38:03 am mark morreny wrote:
> What I need to do is to try to route called based on the dialed number as I
> have multiple DIDs on my line. Is this something that can be done? Is
> this something to do with the hardware that I am using? If so, what kind
> of hardware do I ne
On March 20, 2008 02:33:52 pm Anselm Martin Hoffmeister wrote:
> Am Donnerstag, den 20.03.2008, 16:59 +0200 schrieb Tzafrir Cohen:
> > And what happens if at the time of the shutdown there was a
> ROTFL
> Trafrir, you made my day.
Oh god, I didn't realize that wasn't a typo until you wrote that..
On March 19, 2008 05:05:05 pm Bill Andersen wrote:
> CentOS release 4.4 (Final)
> Kernel 2.6.9-34.0.2.ELsmp (SMP)
> Asterisk 1.4.16.2
> Dell SC440 w/RAID 1
> Digium TE120P
>
> The GUI is a commercially available product, to remain un-named at this
> point.
Ok, and what specifically are the types o
On March 19, 2008 07:00:20 pm Steve Totaro wrote:
> I would not consider a "Dell SC440 w/RAID 1" "Server Grade" you can
> pick them up for $250 on sale.
Why not? Is the price not high enough, or is there some technical reason? I
ask because your only explanation as to why it's not server grade
On March 19, 2008 12:43:21 pm Bill Andersen wrote:
> I'm a USER of Asterisk. We purchased 3 commercially available
> "Asterisk Based" PBXs a little over a year ago. (I won't mention
> which one at this point - I don't want to bad mouth them - yet!)
> Two of the systems are very small (5 SIP lines/
On Friday 30 November 2007 04:17:36 Philipp Kempgen wrote:
> With SIP you can "attach" custom variables to calls (using
> X-... headers).
> IAX (Inter-Asterisk eXchange!) can't do that (yet).
With IAX2 you can share variables too. I believe Tilghman had supplied a
patch to do exactly that severa
On Monday 12 November 2007 07:54:42 Dave Fullerton wrote:
> From what I've heard, I think your best bet is to buy a multi-port
> T1/E1 card for asterisk, put your E1 in one port and a channel bank in
> the other port, then plug your fax extension into an FXS port on the
> channel bank. Since both
On Tuesday 16 October 2007 15:25:13 Philipp Kempgen wrote:
> Michael Collins wrote:
> > I don't know if it's relevant or not, but I do know that at least one
> > legacy PBX vendor (NEC) has a 'solution' that helps with some of the
> > sillier CDR's that could get generated. They have what they cal
On Tuesday 16 October 2007 03:49:37 Atis Lezdins wrote:
> Well, as far as i have tried - i never get ANSWERED in DIALSTATUS. Only
> thing that continues is h extension.
You must of course use 'g' in the Dial flags so that it continues on in the
dialplan after hangup...
-A.
_
On Monday 15 October 2007 17:18:00 Andreas van dem Helge wrote:
> On 10/11/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote:
> > Yes, see the t1e1override module parameter in wct4xxp/base.c. IIRC, it's
> > 0xff to hard code to E1 mode, and set it to 0 for T1 mode. -1 is to use
> > the jumper sett
On Monday 15 October 2007 19:50:03 Philipp Kempgen wrote:
> I'd basically just Dial() 2 times:
> Dial(SIP/...);
> Dial(Zap/...);
>
> No need for priority jumping. And not need to check if
> the ChanIsAvail(). Just Dial().
Why not just do it the correct way?
Dial(SIP/,,g)
GotoIf($[${DIALSTATUS} =
On Tuesday 09 October 2007 14:32:38 Matt wrote:
> http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
And your point, precisely, is what?
Someone who has a criminal record can't be a technical authority? Someone
can't have a criminal record without being a scumbag? Or perhaps that you
prefe
On Tuesday 09 October 2007 10:14:23 Matt wrote:
> Before you put any work into this... ask yourself... what exactly are you
> hoping to accomplish? There is no way one system can handle a DS3s worth
> of traffic... therefore, what good would this do?
Whatever gave you the notion that a modern PC
On Saturday 29 September 2007 18:43:59 Andrew Joakimsen wrote:
> That's horrible. I don't buy too many IP phones these days, but can
> anyone suggest a place better than the scumbags at VoIP supply?
I don't know about you, but I've had nothing but very good results with
VOIPSupply. I didnt do hu
On Thursday 30 August 2007 9:49:57 am Matt wrote:
> I want to reply to this my initial comments were not trolls.
> I think, however, my initial comments reflect what alot of the
> asterisk community is experiencing.WE support asterisk for people.
> WE also sell phone systems based somew
On Thursday 23 August 2007 11:22:23 pm Stephen Bosch wrote:
> > dial(SIP/polycom-on-my-desk&Local/5551212,15,tr)
> Will this work even if the Local is pointing to a Zap channel?
> As far as I know, this only works with SIP or IAX outgoing.
I'm not sure where you are getting that assumption from, a
On Thursday 16 August 2007 2:57:06 pm Barry L. Kline wrote:
> As far as tutorials, just pick up a copy of "Asterisk: The Future of
> Telephony." Most of the "howto" for compilation is there, albeit
> somewhat dated until the newer version of the book hits the press.
I'd wait a couple of weeks, t
On Thursday 09 August 2007 1:18:17 pm Stephen Bosch wrote:
> > Why would anyone want a Collingwood DID? I don't answer calls from
> > Collingwood simply because I am plain old not interested in the free
> > vacation weekend I keep winning. :-)
> Are there lots of boiler rooms in Collingwood?
..
On Thursday 09 August 2007 8:15:09 am Zeeshan Zakaria wrote:
> Does anyone provide 705441XXX, 705444XXX or 705446XXX DIDs? This is for
> Collingwood area in Ontario.
Why would anyone want a Collingwood DID? I don't answer calls from
Collingwood simply because I am plain old not interested in the
On Wednesday 08 August 2007 1:39:34 pm Mike wrote:
> exten => 12345,1,AGI(agi-helloworld.agi)
AGI is an application, and you've called it.
> exten => 12345,1,Noop(${AGI(agi-helloworld.agi)})
AGI is not a function. You cannot "nest" applications like that. The NoOp
application cannot call anot
On Wednesday 08 August 2007 1:17:24 pm Jay R. Ashworth wrote:
> > Digium has taken the stance that it's better to set arbitrary variable
> > names to arbitrary values rather than allow what many would consider the
> > perfectly accepted method of using a $? type of return code in addition
> > to an
On Wednesday 08 August 2007 12:10:47 pm Mike wrote:
> I can be a bit slow sometimes, but you said that it's not possible, and on
> the other hand told me to write my own function (which appears to
> contradict the first statement).
That's because I'm a little slow today... I thought you were aski
On Wednesday 08 August 2007 11:41:38 am Mike wrote:
> But what if I wanted to write my own custom application for one specific
> purpose, I can't set a return value? It's not possible at all?
Not possible, to my knowledge.
> Let me put it this way then, if I needed to have some processing all do
On Wednesday 08 August 2007 11:24:45 am Mike wrote:
> Is it possible to write a function in Asterisk, that returns a value? Sort
> of like any programming language allows?
Digium has taken the stance that it's better to set arbitrary variable names
to arbitrary values rather than allow what many
On Thursday 26 July 2007 12:51:06 pm Eric Chamberlain wrote:
> Could you elaborate on how you configure the MWI of the mobile device to
> use asterisk voicemail?
yes, please explain. SMSing the phone doesn't light MWI, unless you get
access to the "raw" SMSC, as all the email gateways just mangl
On Thursday 05 July 2007 2:38 pm, Doug Lytle wrote:
> Already did that. I use ASSP for filtering. Digium and associated
> mailing lists are white listed. There was only 1 attempt for deliver
> and there were no delays. I subscribe to 10 mailing lists (Including
> the dev list) and they are not
On Tuesday 03 July 2007 9:47 pm, Joe acquisto wrote:
> We get to do that, because, back in the late 1700's . . . we won.
Hey man, I'm Canadian... We've got our own set of funny accents, and don't get
us started on the Quebecois. Not even the Parisians can understand
THEM! :-)
-A.
___
On Tuesday 03 July 2007 7:20 am, J. Oquendo wrote:
> (again) Dell. We know based on someone's accent and lack of proper
> use of grammar, they are not speaking to us from a location in
> the USA. How can we "validate" that such instance is illegal. It
You obviously have not been around any city ce
On Wednesday 06 June 2007 3:33 pm, Jared Smith wrote:
> Hopefully in the future we'll have the RTCP reports logged (either as
> part of the CDR records, or in a Call Quality log of some kind).
> Until then, I'm pretty sure you can listen for RTCP events through the
> Asterisk Manager Interface, and
On Tuesday 05 June 2007 3:25 am, F6HQZ wrote:
> I am using Kirk DECT/SIP 600V3 every day.
> This system run very very well behind an Asterisk, with transfert feature,
> caller ID display and so...
> Seen as an IP-Phone running a separate SIP account for each handset.
> Consider the 600V3 server as
On Monday 04 June 2007 10:28 am, Paul Hayes wrote:
> Looking at the OP's requirements list in the first post, there is
> nothing currently on the market which will cover anything like all those
> features (and do it well!).
I've got the WIP300 and 330 on my list, with the latter being the more lik
On Monday 04 June 2007 8:24 am, Bryan Laird wrote:
> - Physically the phone feels very light and cheap, that if you were
> to drop it that it might not survive very many of them. The buttons
> feel more
> like a toy than anything else but once you get beyond that it works.
How are they for big
On Sunday 03 June 2007 4:30 pm, Alex Crow wrote:
> No frills, specs look good, price seems excellent!
> http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519
That's a terrible phone. I've tried them. the screen is pretty much useless,
the buttons are *TINY*, the battery life horrib
I've tested a few different wifi SIP phones for office/factory use, and
generally have been underwhelmed. Before I grab another few and test, I'd
like to ask around here about the candidates.
My requirements are relatively simple:
- WEP/PSK should be supported WITHOUT dragging the phone down
-
On Friday 01 June 2007 9:24 am, Rob Schall wrote:
> comcast high-speed, thinking that would be more than enough. Turned out
> though, with most high speed solutions, there is some limited packet
> loss and its just to be expected. You internet browsers, etc, would
Limited packet loss != **EIGHT SE
On Saturday 26 May 2007 1:21 am, Edgar Guadamuz wrote:
> Very good... by the way, I'm studing electrical engineering and I've
> chosen asterisk scalation as my final graduation project. I hope do a
> similar work within and asterisk cluster.
I've been working as an EE, and I've got to ask... what
On Thursday 24 May 2007 11:30 am, Steve Edwards wrote:
> As I remember, the "key" was to add code to conf_run() to take the user
> out of the conference, play the custom sound file, and put them back into
> the conference. These in/out steps are needed to keep that user in sync
> with the conferenc
On Sunday 20 May 2007 11:36 am, Jon Pounder wrote:
> how many cable feet were you ever able to actually get various speeds at ?
Depended on the hardware and wire gauge. I was able to do 1250kbps
symmetrical on a 4kmish loop very reliably.
> around here it might just be the geography but I think
On Monday 21 May 2007 3:38 pm, Doug Lytle wrote:
> Doing a 'man sox' does wonders:
The question, however, is is Asterisk playing them louder than normal, or are
they recorded too loudly to begin with?
Adjusting volume gains on these files is the LAST thing you should do.
Determine what the nat
On Wednesday 16 May 2007 1:07 pm, Alex Balashov wrote:
> You would need two 4-port FXO cards. One to take the 3 outside POTS lines,
> and one to generate the 3 FXO lines toward the legacy PBX pretending to be
> the far end. Produce a simple dial plan that basically forwards nearly
> everything in
On Wednesday 16 May 2007 1:00 pm, François Delawarde wrote:
> Thanks again for your help, and sorry if I was not 'that' convinced on
> your first answer and sent a mail to Xen user mailing list to check if
> they knew that issue (no answer yet). Now I almost believe you a lot. If
> I understand wel
On Wednesday 16 May 2007 11:47 am, Olivier wrote:
> Do you mean nobody has ever done this before (as I thought before asking
> this question to the list) ?
> So which tool KDE users are using for this ?
I am a KDE user, although on Slackware. Have been for many, many years.
Typically you will fi
On Tuesday 15 May 2007 3:31 pm, Jeremy Mann wrote:
> 1.Is it feasible to use asterisk as a Man in the Middle for a T1
> PRI system? The idea is to intercept outbound calls from the Nortel PBX
> and redirect them via VoIP to another asterisk box at another branch
> transparently(thus saving
On Monday 14 May 2007 10:41 am, [EMAIL PROTECTED] wrote:
> I want to ask you if asterisk, when I use the command park(), gives me for
> example a variable that contains the slot position where it parks the call
> or if it only tells me (audio) in the channel this position number? In
> other words,
On Friday 11 May 2007 7:46 pm, Jon Pounder wrote:
> well actually there is dialtone on the unprovisioned pairs for the
> most part, but you can only dial repair, the telco office or 911 on
> them. I am not sure if its all pairs or just pairs that had a line
> provisioned at one time. ANAC just repl
On Friday 11 May 2007 5:45 pm, Jon Pounder wrote:
> again, I'm interested to know anyone whose actually done this, and what
> the results were, since I have been thinking of the same thing for a
> while.
I'd run about two dozen of these things using a variety of equipment.
Pairgain SDSL modems (
On Wednesday 09 May 2007 8:26 pm, Robert Augustyn wrote:
> I understand that these sets are digital but what about connecting
> Asterisk fxs to Nortel fxo and keep sets connected to existing Nortel?
Yes you can do that; I have. No you don't want to; it doesn't work worth a
shit. You lose so ma
On Sunday 06 May 2007 6:42 pm, Forum wrote:
> Can someone recommend a good quality 24 or greater port channel bank?
For FXS: I have personally used Adit600, Access Bank I and IIs. They all work
great, and the AB1 and AB2 products are *cheap*.
For FXO: Adit600. The AB1/2 work, but have no CPD c
On Thursday 03 May 2007 10:18 am, Chris Bagnall wrote:
> It seems that more and more phones these days are coming with XML
> mini-browsers. I'd like to have a go at developing something useful to use
> on them, but in all honesty, most of our customers use their phones to make
> and take calls and
On Wednesday 02 May 2007 3:04 pm, Goke Aruna wrote:
> I have a set up that answer my customer. and its working well,
> however, the number of call to technical dept is what i want to reduce.
> I want all call to get to voice prompt except that that enter when
> minutes is 15, 30, 45, 60(in multiple
On Wednesday 02 May 2007 11:49 am, Danish Samad wrote:
> [salesivr]
> exten => _X.,1,NoOp(Incoming call from user ${EXTEN} and caller id
> ${CALLERID})
> exten => _X.,2,Playback(emptyy)
> exten => _X.,3,Background(Main_Sales)
> exten => _X.,4,WaitExten(2)
> When I press a digit in _X,3 or _X,4 it
On Monday 30 April 2007 4:14 pm, bails wrote:
> > I'm still looking for a miniPCI ADSL chipset that Linux can use, or an
> > actual "raw" ADSL non-PCI chipset that I can design into an embedded
> > system. If anyone has any leads, please don't hesitate to contact me!
>
> Any chance we can get to se
On Saturday 28 April 2007 11:22 am, Chris Bagnall wrote:
> Thanks to all who replied to my thread a few days ago "SIP devices with
> packet loss tolerance". One of the suggestions that came out of that thread
> was to replace routers at users' premises with ones that support QoS.
Sangoma S518 (int
On Thursday 15 March 2007 12:32 am, shadowym wrote:
> Hard to expect the business community to take Asterisk seriously when this
> sort of stuff happens IMHO. I can't understand how 3 of 4 hard drives
> could just suddenly fail simultaneously. There must be more too it. No
> UPS? Someone spilled
On Wednesday 28 February 2007 7:53 pm, Lee Howard wrote:
> The problem, however, as we all know, is that the Asterisk maintainer,
> Digium, requires undue retribution in the form of "disclaimers" before
> it will accept any contribution into the code repository - and in this
> case the author of th
On Wednesday 28 February 2007 3:45 pm, Alan Chandler wrote:
> I am trying to setup an arrangement whereby clients on machines A, B, C
> and D can talk to each other on Softphones. A,B,C are are all Windows
> XP machines, machines D and S are linux. This has to include A talking
> to B and ultimate
On Saturday 24 February 2007 6:48 pm, Matt wrote:
> Now.. back to your issue.
> Setup a crontab to restart asterisk every night. Use a version of
Nonsense. Set up proper monitoring of system resources (memory is only one
resource you should be watching) and help the community out if you're
de
On Friday 23 February 2007 8:35 pm, Zoilo Gomez wrote:
> However, faxing (B410P ISDN bridged to TE410P PSTN) seems to work fine
> with EC at 256 taps on the B410P.
Generally speaking all modems (this includes POS machines and faxes) emit a
tone which echo cancellers recognize and disable themselv
>> On 2/15/07, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
> > This is coming right out of left field, as I've never set up an Asterisk
> > queue
> > or agent system, but is it possible to pause and unpause while in the
> > wrap-up
> > time? W
On Tuesday 13 February 2007 11:30 am, James Fromm wrote:
> Does anyone have a solution to allow an agent to selectively end his
> wrap-up time? We define a wrap-up time of 60 seconds to allow our
> agents to finish their notes from a call. In some cases, the full 60
> seconds is not needed and ou
On Thursday 15 February 2007 6:51 am, Steve Underwood wrote:
> It looks like octasic have started supplying their echo canceller as
> host software for zaptel now. I expect either canceller would work with
> the Sangoma cards, as they currently sit in the zaptel framework too.
Out of curiosity, wh
On Wednesday 14 February 2007 6:00 pm, shadowym wrote:
> Think of what you just said. You just said a "Central Processing Unit" is
> an "Application Specific Integrated Circuit".
>
> If you say so.LOL!
Do you disagree that a CPU is not an integrated circuit specialized in dealing
with the ap
On Wednesday 14 February 2007 4:12 pm, shadowym wrote:
> The algorithms may be similar but EC is an infinitely variable
> non-linear(analog) process. A CPU cannot do that. You can fake it by
> performing cpu intensive rapid calculations one after another but it is
> fundamentally not an analog pr
On Wednesday 14 February 2007 11:19 am, Matthew Fredrickson wrote:
> We noticed that it has slightly better performance characteristics than
> the Octasic, particularly in double talk scenarios, at least from our
> internal lab testing.
How has the testing been with respect to its use on FXO ports
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