On Fri, 2003-09-05 at 01:40, John Todd wrote:
> [nonpedanticexample]
> exten => s,1,DigitTimeout(5)
> exten => s,2,ResponseTimeout(20)
> exten => s,3,Background(type-your-selection)
> exten => s,4,Background(silence/3)
> exten => s,5,Background(type-your-selection)
> exten => s,6,Background(silenc
Hello all!
I've talked recently to the head of R&D dept. of Telkom Telos
(www.telos.com.pl) - a big Polish company specialised in making
phones. I gave them the idea of creating a cheap (cost-effective)
hardware IP phone. The phone we discussed would include hardware
support for IAX (though pr
Is there an easy way to change the realm used for authentication from "asterisk" to
anything else e.g. mydomain.com ??
Thanks
--
__
http://www.linuxmail.org/
Now with e-mail forwarding for only US$5.95/yr
Powered by Outblaze
__
If there was a native IAX phone with GSM support and was around $70, I'd
buy a few, and I know several people in my social groups would get them.
I could even make a business case to get them for the office.
On Fri, 2003-09-05 at 02:56, Grzegorz Nosek wrote:
> Hello all!
>
> I've talked recently
Sounds like a good idea..
My suggestion on looks and features would be to look at somthing like the Snom200,
features like the ability to connect a standard pc type head set to the phone a great
cost cutting features..
As for codecs I would look and G.711, G.729, GSM, iLBC and Speex.. That way
Hello,
I am building an asterisk PBX with some stuff to make a usable VOIP /
PSTN Gateway. I use the following devices:
SIP Phones from GrandStream for VOIP side
OpenLine4 from voicetronix for PSTN Side
I am building things step by step with some priorities.
I have now a working system able to
I must say that I would be EXTREMELY interested in distributing such
phones here in Switzerland ... We see a lot of demand here ... I am even
willing to beta-test if needed.
For hardware/software infos, have a look at :
http://www.tuxscreen.net/
This is a completely open-source and open-hardwa
These are probably more issues for grandstream.. Maybe mail [EMAIL PROTECTED] with the issues about dropping both calls when the phone is hung up..
Later
Hello,
I am building an asterisk PBX with some stuff to make a usable VOIP /
PSTN Gateway. I use the following devices:
SIP Phones from Grand
GS phone does blind transfer only. Afer pressing transfer, you will
hear dialtone and then dial the number, after dial the whole number,
either wait more than 5 seconds or press "redial/send" button, then
hangup, it should work.
--- "WipeOut ." <[EMAIL PROTECTED]> wrote:
> These are probably more
This works only if transfering to a phone wich is onhook. If it is off
hook (busy), it doesn't work
Is there any possibiliy to simulate transfert with dial plan?
Regards,
Daniel
William Zhang a écrit:
GS phone does blind transfer only. Afer pressing transfer, you will
hear dialtone and th
The problem I thought he was refering to was that if phoneA is in a call with phoneB,
then phoneA uses "flash" to put phoneB on hold and call phoneC then..
Problem 1
If phoneC hangs up then both the calls from phoneA to phoneC and phoneA to phoneB are
disconnected.
Problem 2
PhoneA has no way o
It is exactly that and noway for PhoneA to connect PoneB and PhoneC
each other.
Daniel
WipeOut . a écrit:
The problem I thought he was refering to was that if phoneA is in a call with phoneB, then phoneA uses "flash" to put phoneB on hold and call phoneC then..
Problem 1
If phoneC hangs u
At 00:56 5-9-2003 -0400, you wrote:
I saw your posting about the cisco ata186 I2 vs I1 and the simple vs
complex impedance.
I ordered a cisco ata186 i2 for use in Canada by mistake, didn't know that
I needed the I1
version.
Will the I2 version work in Canada with regular anlog phones, or will I
On Thu, 2003-09-04 at 19:20, Dave Alan Caruana wrote:
> has anyone got G729 and SIP working together?
> some config examples would help :)
This configuration works for me:
sip.conf:
[grandstream]
type=friend
username=grandstream
insecure=yes
host=dynamic
context=sip-out
nat=yes
canreinvit
I'm no where near an expert (or even very knowledgable on some of this stuff),
but a fair number of machines (regardless of whether its a 7960 or whatever)
will not fail over to secondary/backup gateways unless the primary is totally
non-responsive. That usually means if the proxy responds with eve
Title: Carlos Fernández Puente
Hi,
I have a
problem when i try to set an asterisk environment variable while asterisk
is running an AGI aplication.
I post the
few code lines (in C).
printf ("SET
VARIABLE agisel = %s\n\n",agiselected);fprintf(stderr,"SET VARIABLE agisel =
%s\n\n",agiselected);
hi great gurus of asterisk :)
could somebody remind me the key combination to send a call
into the parking queue ?
while you're at it, are there any other key combinations I should know??
eg. put a call on hold etc.
thanks
Dave
___
Asterisk-Users ma
That makes perfect sense. It works perfectly.
Thanks to you and Matteo who suggested the same solution.
-Original Message-
From: Alastair Maw [mailto:[EMAIL PROTECTED]
Sent: 4 septembre, 2003 11:28
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Call script after hangup
Frank N. wrot
To park a call you simply transfer the call into extension 700 (this is the default
and can be changed)..
To get the call back you just dial the parked location.. If you are using an IP phone
this is a problem becasue it will not tell you the location of the parked call so you
will not know whe
On Fri, 2003-09-05 at 03:25, Marcel Prisi wrote:
> I must say that I would be EXTREMELY interested in distributing such
> phones here in Switzerland ... We see a lot of demand here ... I am even
> willing to beta-test if needed.
>
> For hardware/software infos, have a look at :
>
> http://www.t
Steven Critchfield wrote:
On Fri, 2003-09-05 at 03:25, Marcel Prisi wrote:
I must say that I would be EXTREMELY interested in distributing such
phones here in Switzerland ... We see a lot of demand here ... I am even
willing to beta-test if needed.
For hardware/software infos, have a look at :
On Fri, 2003-09-05 at 08:58, Marcel Prisi wrote:
> Steven Critchfield wrote:
>
> > On Fri, 2003-09-05 at 03:25, Marcel Prisi wrote:
> >
> >>I must say that I would be EXTREMELY interested in distributing such
> >>phones here in Switzerland ... We see a lot of demand here ... I am even
> >>willi
hello,
i have problem with oh323 channel driver (tryied differnet versions).
when i try to make call between oh323 - sip, oh323-isdn, oh323-capi
asterisk crash with segmentation fault. Channel driver was compiled with
pwlib 1.5.0 and openh323 1.12.0 libs.
Does anybody know solution ?
WrapH323Conne
> Examples I'd like to see:
>
> 1)
> ${FOO} contains 12345#
> ${HASH} contains #
something like this:
exten => 123,1,Gotoif($[${FOO} : 12345#]?2|102)
>
> If ${FOO} contains the contents of ${HASH} anywhere, go to 2. If not, goto 102
>
> exten=> 123,1,GotoIf($[...???...]?2|102)
>
>
> 1.1)
>
You could use ResponseTimeout together with Background instead of playing
silence files.
Martin
On Thu, 4 Sep 2003, John Todd wrote:
>
> As has been noted before on this list, the Wait() application does
> not listen for keystrokes from users. Many of you, like me, have
> looping Background(),
what i'm asking is what is the key sequence
you have to dial for the transfer ..
it was something like *7# if I remember
well, I know I had it working, but the client
lost the paper I wrote it on for him, and I can't
trace the email I got it from!
cheers
Dave
- Original Message -
From: "
Well, on the other hand Release Notes for software 4.2
(http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/sip/relnote/phnrn42s.htm#58498)
says:
"The SIP phone can register with a backup proxy to support Survivable
Remote Site Telephony (SRST). If the main prox
It's defined in /etc/asterisk/parking.conf
and set by deafult as 700
Martin
On Fri, 5 Sep 2003, Dave Alan Caruana wrote:
> what i'm asking is what is the key sequence
> you have to dial for the transfer ..
>
> it was something like *7# if I remember
> well, I know I had it working, but the clie
On Friday 05 September 2003 08:21 am, Rich Adamson wrote:
> I'm no where near an expert (or even very knowledgable on some of this stuff),
> but a fair number of machines (regardless of whether its a 7960 or whatever)
> will not fail over to secondary/backup gateways unless the primary is totally
>
If you are using ulaw codec, try change it to alaw.
oh323 currently has some problems with ulaw codec.
Michael
On Friday 05 September 2003 10:22 am, Marian Danisek wrote:
> hello,
> i have problem with oh323 channel driver (tryied differnet versions).
> when i try to make call between oh323 - s
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Dave Alan
> Caruana
> Sent: Friday, September 05, 2003 9:37 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] call parking -- what was the key
> combination?
>
>
> what i'm asking is what is the
Hi all together,
i have following configuration:
ISDN Phone ---> ASTERISK1/PRI ---> ASTERISK1/IAX ---> INTERNET --->INTERNET
ROUTER (Port 5036 nat) ---> ASTERISK2/FXO/ANALOG DEV
The call flows fine, but no sound will be transfered.
On ASTERISK1 a message like "stopped sounds" occurs
It just dawned on me as I was playing with the manager interface - it
can't be very difficult at all to write an Win32 app that serves as a
"lamp field". Between 'Newchannel', 'Newstate', and 'Hangup' events, all
of the information is there.
I've heard several requests for line appearances, but m
If you put "Tt" in your dial statement you can type "#" some number to
transfer to.
Of if you can send flash hooks that will work as well.
Dave Alan Caruana wrote:
what i'm asking is what is the key sequence
you have to dial for the transfer ..
it was something like *7# if I remember
well, I kn
Can you turn up the gain on sip calls?
I would like the calls to be louder but
the phones are already at max volume.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
I am new to this forum. As well as a new user
of Asterisk. My vendor installed the system and we are still trying to get
all the bugs out of it! I have a few questions about configuration and a
program to view who is on what extensions.
I am looking for a program that will work on my
Re
Does any one have any experience setting up asterisk with polycom IP
phones? All i have been able to figure out about them is that they connect
to an FTP site on "boot". I tried going to the site to see what files are
there but it seems they deny directory browsing.
Any one have any clues as to
Hello all,
I have a newbie question about the CDR.
Does "billable" seconds equal "end time minus the time that a human actually picks up
the phone"?
thanks,
Steven
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listi
It should if we can preciselly know on the given interface.
So that might not be true with analog interfaces ...
regards
Martin
On Fri, 5 Sep 2003, Steven Poelmans wrote:
> Hello all,
>
> I have a newbie question about the CDR.
> Does "billable" seconds equal "end time minus the time that a hum
yes, is to say that's equal to end time minus the ring time
before the remote party picked up.
matteo.
Il ven, 2003-09-05 alle 20:32, Steven Poelmans ha scritto:
> Hello all,
>
> I have a newbie question about the CDR.
> Does "billable" seconds equal "end time minus the time that a human actuall
Hi Steven.
I have done as you suggested and I'm still getting the same problem.
/proc/interrupts lists the following:
0: 45489 XT-PIC timer
1: 235 XT-PIC keyboard
2: 0 XT-PIC cascade
5: 335816 XT-PIC wcfxo, Intel ICH2
8
As you can see wcfxo is still sharing an IRQ. It won't work well if it
shares an IRQ.
On Fri, 2003-09-05 at 19:39, Zak wrote:
> Hi Steven.
>
> I have done as you suggested and I'm still getting the same problem.
> /proc/interrupts lists the following:
>
> 0: 45489 XT-PIC timer
>
Martin Pycko wrote:
What's the Spain busy tone ? x ms tone, y ms of silence etc ...
If I remember correctly, 0.2 ms on 0.2 ms off repeated. All tones
are 425 Hz, -10dBm
It may also add 0.4ms off after every 3 on/off cycles
--
If you have 0.4 ms silence every 3 cycles then try to uncommnet
BUSYDETECT_TONEONLY in asterisk/Makefile and recompile.
regards
Martin
On Fri, 5 Sep 2003, Norberto Garcia Prieto wrote:
> Martin Pycko wrote:
>
> >What's the Spain busy tone ? x ms tone, y ms of silence etc ...
> >
> >
> >
> If
"Samy Touati" <[EMAIL PROTECTED]> writes:
>Will the I2 version work in Canada with regular anlog phones, or will
>I need to change it.
No idea... I'm not certain what Canada does with analog phones. I
suspect they're the same as the US ones.
--Michael
__
Hi,
Just got hold of Ericsson webswitch 100 G4 (4 FXO ports).
IT uses H323 as codec. The plan is to use it for incoming/outgoing calls on
two PSTN lines.
I have ATA 186 which is using SIP to use asterisk services.
I can not figure out:
1. where in asterisk do I edit conf files so it uses webswit
hi, what does "tr" means at the end of line?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
Gillham
Sent: 05 September 2003 06:29
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960
Andrew Joakimsen wrote:
>>>exten => 1000,
A friend and I have recently added the ability to announce the callers
position in the call queue every x seconds.. or even just inject an
anouncement every x seconds. All setup in queues.conf and can be setup
per queue.
My next project is to add the ability to announce the callers estimated
wait
If I was calling I would like to know either how long the the person
that's been in the queue the longest has been waiting OR an average of
how long the callers were in the queue before they were answered (over
the last X (where x in a config option) mins
On Fri, 2003-09-05 at 14:05, Brian West wr
In reading the source for the CDR_CSV module, I understand that it should
use the SIP username as the account code for calls made from SIP devices.
However, nothing is being recorded in the csv file for that field (i.e.
blank value). Is there any way to add an account code for SIP users? I can
On Fri, 2003-09-05 at 14:05, Brian West wrote:
> A friend and I have recently added the ability to announce the callers
> position in the call queue every x seconds.. or even just inject an
> anouncement every x seconds. All setup in queues.conf and can be setup
> per queue.
>
> My next project i
On Fri, 2003-09-05 at 14:00, Senad Jordanovic wrote:
> hi, what does "tr" means at the end of line?
There is documentation, it is even within quick access.
>From issueing a "show application dial" at a asterisk cli prompt I see
the following.
The option string may contain zero or more of the fol
At 12:21 PM 9/5/2003 -0700, you wrote:
In reading the source for the CDR_CSV module, I understand that it should
use the SIP username as the account code for calls made from SIP devices.
However, nothing is being recorded in the csv file for that field (i.e.
blank value). Is there any way to add an
> under 20 minutes
> If I ever heard a time over 20 minutes I'd hang up and call back later,
> or stop doing business with the company. This limits down your number of
> prompts and lowers the expectation of wait time accuracy.
Sprint PCS comes to mind on that longer than 20 min hold times! :P
bk
On Fri, 2003-09-05 at 01:40, John Todd wrote:
[nonpedanticexample]
exten => s,1,DigitTimeout(5)
exten => s,2,ResponseTimeout(20)
exten => s,3,Background(type-your-selection)
exten => s,4,Background(silence/3)
exten => s,5,Background(type-your-selection)
exten => s,6,Background(silence/3)
e
On Fri, 2003-09-05 at 14:41, Brian West wrote:
> > under 20 minutes
> > If I ever heard a time over 20 minutes I'd hang up and call back later,
> > or stop doing business with the company. This limits down your number of
> > prompts and lowers the expectation of wait time accuracy.
>
> Sprint PCS
Title: RE: [Asterisk-Users] app_queue input needed...
There is one thing you have to look out for. Wait time is affected only by the number of calls in front of you, not total calls, the number of agents answering, and the length of calls.
I say this because if you are going to update the ann
thanks very much...
do you know of any other links to documentation, guides, manuals etc.
(Digium site
does not offer much).
The biggest problem so far, I find is lack of docs.
To produce information one does need data.
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PR
On Fri, 2003-09-05 at 14:41, John Todd wrote:
> >On Fri, 2003-09-05 at 01:40, John Todd wrote:
> >
> >> [nonpedanticexample]
> >> exten => s,1,DigitTimeout(5)
> >> exten => s,2,ResponseTimeout(20)
> >> exten => s,3,Background(type-your-selection)
> >> exten => s,4,Background(silence/3)
> >> e
If I was calling I would like to know either how long the the person
that's been in the queue the longest has been waiting OR an average of
how long the callers were in the queue before they were answered (over
the last X (where x in a config option) mins
On Fri, 2003-09-05 at 14:05, Brian West w
So announce this when they first enter once and only once. That
sounds like a reasonable idea.
bkw
On Fri, 5 Sep 2003, McAughan, Matt wrote:
> There is one thing you have to look out for. Wait time is affected only by
> the number of calls in front of you, not total calls, the number of age
> Steven Critchfield wrote:
>
> > On Fri, 2003-09-05 at 03:25, Marcel Prisi wrote:
> >>This is a completely open-source and open-hardware hardware phone
based
> >>on Linux on an ARM embedded platform ... they already had lots of
> >>experience ... but might need some different software ...
> >
>
Message: 2
Subject: Re: [Asterisk-Users] Re: Asterisk Jitters
From: Eric Wieling <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Date: Fri, 05 Sep 2003 11:54:19 -0500
Reply-To: [EMAIL PROTECTED]
As you can see wcfxo is still sharing an IRQ. It won't work well if it
shares an IRQ.
I have changed the pci
Hi all,
I would be most grateful if someone would review my plans for my
new phone system and comment on areas of expected trouble and
advice on what to do better. Instead of moving our Panasonic
KX-TD1232/TVS200 system (ugh...) to our new location, we've
decided to jump into IP telephony with *
> The second method, where a "sliding window" average of wait times in
> the last X minutes is used as the sample base is a bit more
> difficult, but after some thought I am think it will provide a more
> accurate number. Note that an unanticipated result of this method
> may be that some callers
When I call in from an outside POTS line to a Zap channel, and the call
ends, it seems like the hangups are very "sloppy." I see Asterisk give
the hangup command, but on my phone there's lots of clicks and the line
acts like it's staying open for several seconds, then I hear a phone
ringing so
I have discovered something quirky in our Asterisk. If I call in to a
Zap channel (from an outside POTS line), then transfer the call around
several times, I get the above error, after which it will hangup. I
believe Asterisk may issue a SIP CANCEL to the extension it was starting
to dial.
N
I have about a dozen SIP phones up and working, now I want to connect the
asterisk box to our Fujitsu 9600 PBX. I currently have two dial-up servers
conencted to the Fujitsu PBX that I built with mgetty/pppd and have the
lines provisioned the same way as those dial-up server, ESF, B8ZS, and E&M
wi
Would anyone mind emailing me, or maybe posting somewhere their music
on hold .so file?
thx
-ben
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Any recommendations on a hardware based SIP phone to use with *?
I'm looking for something that would be common, as well as quick and easy to source,
somthing relatively quick and easy to configure.
Side note, is SIP automatically enabled in *, or do I have to add a channel driver as
I do with
Sean,
> Any recommendations on a hardware based SIP phone to use with *?
>
> I'm looking for something that would be common, as well as quick and easy
> to source, somthing relatively quick and easy to configure.
I'm very new to this as well, but with 20+ years of telephony and data
network per
This is an oddity of how the POTS works, and has nothing
to do with asterisk.
For almost all domestic switches in the world, the called
party can hang up the handset without disconnecting
the call. If the phone is picked up before a timer
pops (on the order of 10-30 seconds), then the call
continu
The
Vonage service is offered with a SIP Cisco ATA device for connection to an
analog phone.
Is
it possible to connect the Vonage service directly to the Asterisk PBX
bypassing the ATA and FXO card? Are
there other services that offer this capability or something similar to IP
dialtone
No. You can use packet8 if you slightly
modify the asterisk source code (outgoing calls only) or you can use the
service provided by nufone.net
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Friday, September 05, 2003
Title: Fax tone detection problem on bridged (PRI -> SIP) calls
Greetings,
[w_pstn]
exten => 8478106205,1,record,w_faxmsg:pcm
exten => 8478106206,1,Dial,SIP/w_sip
exten => fax,1,Goto(w_fax,8478106207,1)
[w_fax]
exten => 8478106207,1,Dial,Zap/g1/8478106207
We send faxes (from the same fax
More info:
cat /etc/zaptel.conf |grep -v "^#"
span=1,1,0,esf,b8zs
e&m=1-24
loadzone = us
defaultzone=us
cat /etc/asterisk/zapata.conf |grep -v "^;"
[channels]
context=default
signalling=em_w
group=1
channel => 1-24
lsmod:
Module Size Used byNot tainted
wct1xxp
Hi All,
i found an article that explains SIP NAT woes.
http://www.sipcenter.com/files/SIPNATtraversal.pdf
It is a great read for all people in the mailing list that have problems
with SIP when * is behind NAT or when there is NAT between asterisk and a
SIP phone.
Serge
Alex Zarubin wrote:
Greetings,
[w_pstn]
exten => 8478106205,1,record,w_faxmsg:pcm
exten => 8478106206,1,Dial,SIP/w_sip
exten => fax,1,Goto(w_fax,8478106207,1)
[w_fax]
exten => 8478106207,1,Dial,Zap/g1/8478106207
We send faxes (from the same fax machine) to 8478106205 and 8478106206.
On 847810620
Why on earth don't you just compile it?
bkw
On Fri, 5 Sep 2003, Ben Bloomberg wrote:
> Would anyone mind emailing me, or maybe posting somewhere their music
> on hold .so file?
>
> thx
>
> -ben
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
At 08:38 PM 9/5/2003 -0500, you wrote:
Why on earth don't you just compile it?
Thank you! I was going to ask but didn't want to look stupid :)
--Ernest
bkw
On Fri, 5 Sep 2003, Ben Bloomberg wrote:
> Would anyone mind emailing me, or maybe posting somewhere their music
> on hold .so file?
>
> t
I have a few questions regarding SIP and NAT that you may be able to
answer. In both cases, I'm "assuming" that the customer will use SNOM
phones and/or xten soft-phones.
Q1: I know that it is possible to use a STUN server to handle SIP over NAT.
Does this require any special configuration of t
At this point, I'm just trying to save what I have. The installation
I'm using is the one that comes with debian and it doesn't include any
files relating to music on hold. (I'm also a complete and total newbie)
So, if there is a way to recompile the package with music on hold, that
would be aw
Does * support ISDN
Primary Rate Interface (PRI) - 2B Transfer Capability for
T-1/PRI?
In other words
the ability to take a call and join it to another call and then drop off letting
the CO-switch take over.
-Kevin
Kevin Fjelsted, PresidentAltiCom CTI,
Inc.
Track Me Down!One nu
I am having Yet Another Regular Expression problem, but this one
might not be my fault, or at least it might not be obviously my
fault. :-)
exten => 2212,1,SetVar(FOO=123456**)
exten => 2212,2,SetVar(BAR=$[${FOO:-1:1} = *])
This script continues with a value of 0 in BAR.
Similarly, none of the
No. You can use packet8 if you slightly modify the asterisk source
code (outgoing calls only) or you can use the service provided by
nufone.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, September 05, 2003 7:21 PM
T
On Friday 05 September 2003 01:52 pm, Steven Critchfield wrote:
> Just a request, but if you post examples like this, please make it
> complete enough to show best practices.
I definitely agree with your statement Steve; and yet certainly there will be
people who are new to Asterisk (as I am) who
Does * support ISDN Primary Rate Interface (PRI) - 2B Transfer
Capability for T-1/PRI?
In other words the ability to take a call and join it to another
call and then drop off letting the CO-switch take over.
-Kevin
Kevin Fjelsted, President
AltiCom CTI, Inc.
Track Me Down!
One number Access,
On Fri, 5 Sep 2003, Zak wrote:
> I have changed the pci slot of the fxo so that it won't share IRQ with
> another device but the jittering is still
> there. check the interrupts list below
>
>CPU0
> 0:1158022 XT-PIC timer
> 1:807 XT-PIC keyboard
>
John Todd wrote:
Vonage is a silly way to do VoIP with Asterisk - you would have to hook
their box up to an X100P card on your system, which is preposterous.
Not necessarily preposterous; I would certainly allow that its
optimality is arguable.
I agree that Vonage holds a heavy hand over their
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of John Todd
> Sent: Friday, September 05, 2003 11:54 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] VONAGE or IP Dialtone
>
> >No. You can use packet8 if you slightly modify t
Currently, the voice prompts are stored in GSM
format. Is there a way to play other formats, like WAV files? Or can we play the
GSM other than the current compressed format? Maybe a less compressed GSM format
(currently, isn't the GSM mode 8k voice)
Lee Goodman
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