On Mon, 2004-06-28 at 16:16, Rodrigo P. Telles wrote:
I've been using kphone (http://www.wirlab.net/kphone/) with success.
It's simple and works fine :-)
kphone only supports inband DTMF and so will only support DTMF when
using ulaw or alaw.
--
Eric Wieling * BTEL Consulting * 504
On Mon, 2004-06-28 at 20:08, Jean-Yves Avenard wrote:
I've only been watching this list for the past 2 days.
And it seems to be an one way street:
- -Tell about your problems and what you would like to do.
Usually no answer.
Personally I've gotten tired of answering questions over and
ignore the new
'rport' parameter (RFC3581) and continue to function.
CVS this evening had an option added called nat=never option for phones
like the Uniden.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
that says Asterisk-Cvs CVS
Updates to Asterisk and the Core Components. Subscribe or just browse
the archives.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss
and extension when picked up.
Any comments on your experiences would be very much appriciated.
http://www.google.com/search?hl=enie=UTF-8q=site%3Alists.digium.com+doorphonebtnG=Google+Search
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has
by
callprogress=yes or busydetect=yes (the beaten, bloody, dead horse is in
the archives). If your card was sharing IRQs then the expected symptom
would be poor audio quality, not dropped calls.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has
:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss
On Tue, 2004-06-29 at 09:53, Isamar Maia wrote:
I'm trying to do the following:
exten = i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
At that pint ${EXTEN} is i. Try using ${INVALID_EXTEN}
--
Eric Wieling * BTEL
.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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to me that you are using CVS -stable (which seems to support
G726 PASSTHRU) and not CVS -head (which supports G726 TRANSCODING, which
is what you need). What does show version at the CLI show.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently
/asterisk-users
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
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On Sat, 2004-07-10 at 10:11, [EMAIL PROTECTED] wrote:
How I can do to use the g.726 on asterisk ?
Use Asterisk CVS -head.
http://www.asterisk.org/index.php?menu=download
--Eric
--
Useful Asterisk Docs (BOOKMARK THEM!):
http://www.digium.com/index.php?menu=documentation (look at the
Unofficial
On Mon, 2004-07-12 at 11:00, Stefan Rosik wrote:
Hi can anyone help me on this error msg??
dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames
You are using inband DTMF on a call that's not using ulaw or alaw
codec. Don't do that. Use RFC2833 DTMF mode.
--
Eric
and everyone else that needs this functionality
does it.
--Eric
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss
--
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In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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:
http://lists.digium.com/mailman/listinfo/asterisk-users
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In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss
was not detected)
--
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.
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Holger Schurig wrote:
With my current hardware PBX, I do:
get a call from a customer
press transfer, then the number of my collegue
my collegue talks with me
when I press transfer again, I am back to my customer
when my collegue hangs up, I am back to my customer
when I hang up, my customer is at
Tony Nichols wrote:
After calling a bank, or cc processing center; you have to enter your
social security number, or the cc number - followed by the # key.
The lovely * voice responds transfering I'm sorry that was an invalade
selection. Sometimes the IVR on the other end still gets the digits
I'm having an oddball issue with a Polycom SoundPoint IP 500. As you
can see below Asterisk thinks there are 2 SIP channels active, but show
channels tells me there are no calls active. Anyone have any idea why
this is happening? The Polycom occasionally stops accepting calls and
requires a
a message.
--
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In a related story, the IRS has recently ruled that the cost of Windows
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in both: SIP - CHAN_H323 and CHAN_H323 - SIP...
when it will be solved?
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss
On Mon, 2004-07-19 at 10:00, Sebastian Nocetti wrote:
I WANT TO USE G729, I HAVE TO USE IT...
Not while testing you don't. Once you get it working with ULAW ONLY
then see if you can get it working with G729.
--
Useful Asterisk Docs (BOOKMARK THEM!):
On Mon, 2004-07-19 at 12:42, Wiley E. Siler wrote:
Thank you!
Can you tell me more about the dial plan feature? How do you setup the
correct digitmap?
It is all in the Admin Guide you can download from the Polycom web site.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
On Mon, 2003-08-18 at 04:13, Dan wrote:
P.S. I think that for the moment, the cheaper option is to use ATA with some
good and cheap DECT phones (in Europe) without any other feature than Caller
ID (name and number). It can cost you less than 120EURO per port (about 75
EURO for 1/2 ATA and
And break their voicemail.conf stuff as well.
On Mon, 2003-08-18 at 16:11, Tilghman Lesher wrote:
On Monday 18 August 2003 04:06 pm, Mark Spencer wrote:
Does anybody have any reason why I should *not* permenantly replace
app_voicemail with app_voicemail2? If so, speak now or forever cvs
MOH requires that Asterisk transcodes (It also has to transcode to for
PSTN calls and voicemail and playing any sound files). Asterisk can't
transcode to or from G723. Nope. Doesn't work. May very well never
work. Use a different codec.
On Wed, 2003-08-20 at 03:47, Andrew Joakimsen wrote:
. I'd be
happy to pay $10/channel just like I have for the G729 license.
On Wed, 2003-08-20 at 12:28, Andrew Joakimsen wrote:
And if one cannot use a different codec?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Wednesday
On Thu, 2003-08-21 at 13:46, Anton Tinchev wrote:
Don't use FXO cards at all. Just convert your lines into ISDN. Isdn cards
are cheaper and has more features. Any ISDN card that is supperted by isdn4linux
must work, but I recommend you Sedlbauer chipset based.
Digium FXS cards are great.
BRI (more correctly called ISDN BRI) is a digital service.
On Thu, 2003-08-21 at 14:12, Anton Tinchev wrote:
Eric Wieling wrote:
On Thu, 2003-08-21 at 13:46, Anton Tinchev wrote:
Don't use FXO cards at all. Just convert your lines into ISDN. Isdn cards
are cheaper and has more features
Have someone using a SIP device with RFC2833 signaling call you, now
have the press and hold down one of the dialing keys. You'll hear a
short tone then nothing.
On Fri, 2003-08-22 at 11:05, Low, Adam wrote:
Maybe its just me but I find this question a little confusing, the tone duration
The real question is: How much?
On Fri, 2003-08-22 at 23:44, Brian West wrote:
And it runs linux.
http://www.zip4x4.com/ZIP4x4.htm
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504-595-3916 x2111 (Experimental)
877-552-0838 (Backup Phone)
___
The first section basically waits for the user to dial an extension.
The second part is called if the extension they entered is invalid, then
it will go to the extension (at this point ${EXTEN} = i at priority 1
(which basically creates a loop) and the record is never called.
Try:
exten =
On Mon, 2003-08-25 at 09:09, Mark Spencer wrote:
It bugs me not having intercom/paging features. It also bugs me not
being able to look at my phones to see who's on/off.
You can do overhead paging and even stream mp3's using nbsd
Cool! Uh, what's nbsd?
--
BTEL Consulting
850-484-4535
On Mon, 2003-08-25 at 18:40, Adam Roach wrote:
As a side note, I'll point out that the Pingtel phones let
you provision client-side digitmaps. Based on asterisk-like
pattern matching, you get to say how long a digit string
should be matched, and the phone will automatically dial
when it
When you modify the Zap config you have to stop/start asterisk or do a
restart (not reload)
On Thu, 2003-09-04 at 14:25, Mickey Binder wrote:
Just have that zap channel in the pickupgroup = callgroup of the sip
phones
Hmm...I must be stupid ;O), can't get it to work.
In zapata.conf I
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
1:
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
Try this:
Dial(Zap/g4/123456,20,t)
I've found that mixing | in a Dial() can cause a problem. I don't know
why and I don't know when, but I had a problem.
or this:
Dial(Zap/1/123456,20,t)
Just in case you have some problem with your group= settings.
On Mon, 2003-09-08 at 02:16, Surajee
You will want a T100P card with a Channel Bank with FXO ports in it. A
channel bank will convert each of the FXO ports into a single DS0 on the
T-1 side going to Asterisk.
On Mon, 2003-09-08 at 05:34, Dan wrote:
Hi,
max we've got running happily without issues is 4 x X100P
and 3 X100P +
There is no reason to use E-1 if he's just going Channel
Bank-Asterisk. E1 channel banks tend to be a lot more expensive than
T-1 channel banks as as long as he's not planning on connecting directly
to the telco (without a channel bank) it should not be a problem.
On Mon, 2003-09-08 at 07:32,
Yes, depending on the channel bank. GENERALLY, people can have problems
with FXS ports on channel banks (no FSK while on hook, etc), but not
usually with FXO ports, it would depend on the channel bank. You want
your channel bank to support callerID (for your country) and support
disconnect
IAX doesn't use RTP. All voice and control data goes via the one UDP
port.
On Mon, 2003-09-08 at 21:17, Wade J. Weppler wrote:
You need to make sure that the RTP traffic is passing through. This
sounds like your problem.
Ports are specified in rtp.conf
-wade
-Original
That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint. Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
endpoints use the same protocol (SIP) AND the same codec.
On Tue, 2003-09-09 at
It would have to do some kind of trascoding, but it's a non-issue since
G729 is not involved and the CPU overhead is minimal.
On Tue, 2003-09-09 at 15:26, Mike Ciholas wrote:
On Tue, 9 Sep 2003, Eric Wieling wrote:
Transcoding would be required for access to ANY of the asterisk
sound
The source code for your kernel is not the same as the kernel actually
running on the machine.
On Tue, 2003-09-09 at 20:38, Rich Adamson wrote:
After following your suggestions, I get
[EMAIL PROTECTED] zaptel]# /sbin/modprobe ztdummy
/lib/modules/2.4.20-18.9/misc/zaptel.o: kernel-module
Yes, on ISDN PRI. On analog you can try the busytetect and progress
detect but that always disconnects my calls at random times.
On Wed, 2003-09-10 at 00:37, Chee Foong wrote:
Yes you are right, Sorry my mistake.
So, is there a way to detect busy, answer, or no answer call?
Foong
-
In /etc/asterisk/zapata.conf:
busydetect=no
callprogress=no
On Wed, 2003-09-10 at 02:44, Surajee Ratnayake wrote:
hi,
I hav a softPBX setup. Our set up has 2 servers, one is connected to
an ISDN PRI E1 coming from PSTN central office and the other server is
connected to another E1 which
I'm pretty sure the info has been posted to the mailing list several
times and should be in the searchable archives.
On Wed, 2003-09-10 at 14:28, Peter Pauly wrote:
On Tue, Sep 09, 2003 at 02:38:01PM -0500, Eric Wieling wrote:
That would be reinvite= and canreinvite= in the user entry for each
You can't use music on hold with G729 unless you buy a G729 license from
Digium. I doubt you can use the r option either since that needs
Asterisk to generate a ringing sound and unless you have the G729 codec
for Asterisk that will fail.
Unless you buy a G729 license you will not be able to use
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_book09186a00800c31f5.html
On Fri, 2003-09-12 at 06:11, Julien wrote:
Hi,
I use * fine with my ata186 but i can't find on the web an answer to my
question, in the configuration menu you can choose the caller
Does anyone know how you specify MD5 auth on a register = line?
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I have a parking.conf:
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
parkingtime = 45 ; Number
): zt_hangup(Zap/5-2)
DEBUG[1167306032]: File chan_zap.c, Line 1593 (zt_hangup): Hangup: channel: 5 index =
0, normal = 15, callwait = -1, thirdcall = 17
On Fri, 2003-09-12 at 14:32, Eric Wieling wrote:
I have a parking.conf:
[general]
parkext = 700 ; What ext. to dial
More followup:
If I call echotest it doesn't work, but if I call a real phone (Zap or
SIP via a remote Asterisk Server via IAX it works.
On Fri, 2003-09-12 at 14:32, Eric Wieling wrote:
I have a parking.conf:
[general]
parkext = 700 ; What ext. to dial to park
Make sure you don't have progressdetect=yes in your zap config.
On Fri, 2003-09-12 at 18:11, Dan Fernandez wrote:
Ive been having this same problem for a few weeks now.
WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event):
Ring/Off-hook in strange state 6 on channel 1
I get this
http://www.nufone.net/downloads/
On Sat, 2003-09-13 at 17:31, Senad Jordanovic wrote:
Any knows a link where to download:
Open H323 v1.11.7
and
PWLib v1.4.11
Thanks
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
[Any alias name here for x.com]
hostname=x.com
auth=md5
On Friday, September 12, 2003, at 08:47 PM, Eric Wieling wrote:
Does anyone know how you specify MD5 auth on a register = line?
--
BTEL Consulting
850-484-4535 x2111 (Office)
504-595-3916 x2111 (Experimental)
877-552
canreinvite=no and reinvite=no in the [exten/username] entry in
sip.conf.
On Mon, 2003-09-15 at 10:08, Shaun Ewing wrote:
Hi,
Today I was assigned a block of IPs by my ISP and decided to move the
Asterisk box onto a public IP.
I am also using 10.1.0.0/16 on my LAN, with the IP phones
Have your tried switchtype=5ess?
On Mon, 2003-09-15 at 17:13, tony mancill wrote:
I have a T1 PRI from the telco and have been trying to get it to work
with the wildcard T400P.
I know framing and coding is esf and b8zs. what else should I be
putting into the zaptel.conf and
Have your tried setting busydetect=no and callprogress=no in
/etc/asterisk /zapata.conf?
On Tue, 2003-09-16 at 12:19, John Sellens wrote:
| From: Dan Fernandez [EMAIL PROTECTED]
| Date: Tue, 26 Aug 2003 16:30:40 -0400
|
| All of a sudden I am getting the following warning Ring/off-hook in
If your Asterisk server is on the same machine as your DHCP server then
you should not need the .cnf file. My 7910 (running SCCP/Skinny) finds
my Asterisk server just fine. If your DHCP server is not running on the
same machine as your Asterisk server you may fine this URL to be
helpful. If
See my changes below
On Thu, 2003-09-18 at 12:20, Lars Fredriksson wrote:
[7101]
context=sip
type=friend
host=dynamic
dtmfmode=inband
mailbox=7101
--
Sample configs and more: http://www.fnords.org/~eric/asterisk/
BTEL Consulting
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On Fri, 2003-09-19 at 10:04, Sean Rodger wrote:
Can anyone tell me if they have had any problems using the Digium X100P
cards and the
Cisco ATA186 together with asterisk??
Yes. The only codec that is compatable with Asterisk without additional
non-free codecs is the ULAW or ALAW codec. See
I have inbound IAXtel calls working, but they come into the wrong
context.
I have a context= line in general above the register line in iax.conf
Does anyone have any ideas what might be happening?
--
Sample configs and more: http://www.fnords.org/~eric/asterisk/
BTEL Consulting
Are you using SIP or H323? If SIP, what are the allow= and disallow=
lines in your sip.conf?
On Mon, 2003-09-22 at 03:08, Chee Foong wrote:
IC, does that means they are not compatible?.
Funny thing is, call make from asterisk to AS5300 is fine using codec
G.729.
But call from
I have found that mixing the Dial() format with | can cause problems.
Does Dial(H323/ip$12.34.56.78,120,r) work as expected?
On Mon, 2003-09-22 at 03:09, Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 22 September 2003 04:02, Jeremy McNamara wrote:
You have
for my network
Any with AS5300 manage to get the digium's g.729 working
Foong
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, September 22, 2003 4:10 PM
Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300
Are you using SIP or H323
since they are different.
Thanks
Foong
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, September 22, 2003 5:30 PM
Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300
add a disallow=all above the allow=g729 line.
On Mon
):
ParseHeader ():
Using Eric Wieling [EMAIL PROTECTED] as reply-to address.
Using [EMAIL PROTECTED] as errors address.
Got MIME info: _disposition=inline, _encoding=7bit, _type=text/plain, boundary=, charset=iso-8859-1
Finished parsing message header.
Forcing message to be multipart/mixed
)
This message has been 'sanitized'. This means that potentially
dangerous content has been rewritten or removed. The following
log describes which actions were taken.
Sanitizer (start=1064266495):
ParseHeader ():
Using Eric Wieling [EMAIL PROTECTED] as reply-to address.
Using [EMAIL
-877-677-9643 x2111 (Toll Free)
This message has been 'sanitized'. This means that potentially
dangerous content has been rewritten or removed. The following
log describes which actions were taken.
Sanitizer (start=1064309189):
ParseHeader ():
Using Eric Wieling [EMAIL PROTECTED] as reply
4 = G.711 u-law
256 = G.729A audio
So what Asterisk is saying is that the H323 device is using G729 and the
MGCP device is using ulaw/G711.
Since Asterisk can't translate between those two codecs unless you have
a G729 license installed the call fails.
Do you specify the allowed codecs in your
It's not a general problem. Can you give us more info like the Dial()
command and any output on the Asterisk console. As good as some people
on this list are at solving problems, I'm not aware of any mindreaders
on the list.
On Tue, 2003-09-23 at 09:55, Thomas Haeger wrote:
Hi all,
can
In the USA and Canada there are no specific area codes for cell .vs.
landline. In the USA most places have have flat rate calling so the
caller doesn't really care if they are calling a landline or a cell
phone. There are cellphone prefixes in each area code, but I'm not
aware that anyone uses
I have posted a link to the tarball of my rather simple AGI script that
will call back the number the AGI script receives as your CallerID
number. Currently only calls back 4 digit extensions (2000 thru 8999)
and IAXTel 700 numbers.
The tarball, other scripts, and various ither info can be found
Run the command festival
Give it the command (SayText Would you like to play a game?)
Does it say anything? If not, then there's a problem Festival.
Type (Quit) to quit the festival app.
On Wed, 2003-09-24 at 08:22, Bryan Nolen wrote:
I am trying to use festival (latest version 1.4.3)
I have
ztdummy doesn't need any zap config. just remove the zaptel.conf
On Thu, 2003-09-25 at 14:38, Senad Jordanovic wrote:
Hi,
I have enabled ztdummy in order to have * compile it.
I can modprobe ztdummy with no problems.
The sole reason, i need ztdummy is to heve musiconhold and meetme
There are several SIP aware NAT routers. Any Cisco router with a
firewall load has SIP aware NAT. There is at least one other brand of
SIP aware NAT router out there, but I don't recall the brand.
On Sun, 2003-09-28 at 17:26, Uriel Carrasquilla wrote:
Nikotel has a solution and one
I would say that the advantage of an e-mail list is that the reader can
decide which mail reader they want to use rather than being forced into
what ever interface the web site wants them to use. I, for one, would
not be on the mailing list if I did not have the great filtering and
sorting
On Mon, 2003-09-29 at 09:40, Mark Spencer wrote:
2) if you build your application as a module that loads into a stock
asterisk server, you do not have to disclose your source
This is FALSE. Even modules for Asterisk MUST be released under GPL,
unless you obtain a license to release them
Check /proc/interrupts to make sure the cards are not shareing IRQs with
anything.
On Thu, 2003-10-02 at 11:43, The Traveller wrote:
Yo all,
I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P.
The call wil sound OK at first, but after 10-20 minutes, the audio will
start to
auth=plain
On Thu, 2003-10-02 at 12:36, Olle E. Johansson wrote:
WipeOut wrote:
Olle E. Johansson wrote:
I still can't get Windows messenger to register with a secret to
Asterisk.
Anthony - do you connect without registering or does Windows messenger
register properly with your
That sound you hear is the sound of mpg321 running. Do an ls -l
/usr/bin/mpg123 if it's a symlink to mpg321 then you have found your
problem.
On Thu, 2003-10-02 at 14:54, john lawler wrote:
I'm trying to get musiconhold to work w/ my Asterisk system, and I'm not
having much success yet.
On Fri, 2003-10-03 at 13:28, Chris Hirsch wrote:
failed checksum
NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2
(Ring/Answered)...
I've only seen this message when using callprogress=yes and/or
busydetect=yes. Set them to no.
--
Sample configs and more:
This entire idea will only work for the USA (and the few other countries
that have unmetered local calls). HOWEVER, in most countries ALL calls,
even if you call the person across the street are billed by the min.
--Eric
On Sat, 2003-10-04 at 10:33, sip wrote:
But it would be a free call to
that I haven't set up my sound card on this
computer and I'm seeing
WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read
error on sound device: Resource temporarily unavailable
Is that necessary to somehow read the caller id burst?
Eric Wieling wrote:
On Fri, 2003-10
Two phones for $130 at
http://www.sipphone.com/tiki-index.php?page=Order%20Now
On Mon, 2003-10-06 at 11:42, Joe Dennick wrote:
Where does one find a hard-phone for $65?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leif Madsen
Sent: Monday,
Each card needs it's own IRQ, not shared with any other device (not even
shared with other Digium cards). Adding a PCI backplane gives you more
slots, but not more IRQs.
On Tue, 2003-10-07 at 16:23, Dennis Gearon wrote:
I am wondering if it's possible to use a bunch of cards in a PCI
cat /proc/interrupts to make sure that the cards are not shareing IRQs
with other devices or each other.
On Tue, 2003-10-07 at 17:34, Ariel Batista wrote:
-- Original Message --
From: Ryan Tucker [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Tue,
try trunking=yes
On Wed, 2003-10-08 at 11:15, duncan wrote:
Im having problems setting up a trunk between two locations. Heres the
setup I have:
Server A is connected to the PSTN at my datacenter
Server B is connected to a clients e1 line at his datacenter
I only want to route calls
Look at the iax.conf.sample in the Asterisk source directory. Maybe the
Handbook is wrong. If so, report it to bugs.digium.com.
On Wed, 2003-10-08 at 11:55, duncan wrote:
try trunking=yes
i already have trunk=yes (which is correct according to the version of the
handbook i have) and it
Since Asterisk isn't converting from one codec to another you should not
need a G.729 license.
On Wed, 2003-10-08 at 12:05, Nicolas Gudino wrote:
Hi List,
I'm new to asterisk. I think it's great! I'm interested in terminating calls
via a SIP provider. I want to know if I need to license
Does festival work outside of Asterisk? Are you running the
festival_server script?
On Wed, 2003-10-08 at 12:09, Juan J. Sierralta P. wrote:
Hi,
Im trying to get app_festival to work. I got the source from the
Debian woody package of festival-1.4.2 and applied the patch that came
Does this work?
festival_client --tts_mode Do you want to play a game?
On Wed, 2003-10-08 at 15:46, Juan J. Sierralta P. wrote:
On Wed, 2003-10-08 at 13:31, Eric Wieling wrote:
Does festival work outside of Asterisk? Are you running the
festival_server script?
What do you mean
Uh, ignore that last email. 8-)
What I mean is can you make festival say things outside of Asterisk,
even it it just generates a .wav file of the speech.
On Wed, 2003-10-08 at 15:46, Juan J. Sierralta P. wrote:
On Wed, 2003-10-08 at 13:31, Eric Wieling wrote:
Does festival work outside
You can send a fake ring by using something like:
exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],20,r)
Assuming the ATA is in the sip.conf as [1234]
However, this does NOT solve the underlying problem.
On Thu, 2003-10-09 at 15:29, Steve Dolloff wrote:
Here is my Configuration
PSTN - Cisco
In the sample zaptel.conf file (see the asterisk source directory) there
is the following:
;
; If you are having trouble with DTMF detection, you can relax the
; DTMF detection parameters. Relaxing them may make the DTMF detector
; more likely to have talkoff where DTMF is detected when it
;
Translation: Asked to transmit frame type G.711 A-law, while native
formats is G.711 u-law (read/write =G.711 u-law/G.711 u-law)
Looks to me like you need a disallow=all in your sip.conf and allow=
lines for the codecs you want to allow, then make sure that the IP
phones you are using support at
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