you need to just specify these things.
1. specify dbhost=ccxcxc(say)
dbname=d(say)
dbuser=cddd(say)
dbpass=sjdhjas(say)
in the [general] section in voicemail.conf
2. create table user in the databse specified above.
You can get the
structure of
Hi,
On Wed, 22 Oct 2003 at 15:44, Chris Albertson wrote:
Also do remember that PCI card's config registrs are little endian
and you will have to mantally byteswap when you read the hex dump.
... or simply use lspci -nv to get the IDs instead of the textual
translations.
cu
Reinhard
On the subject of Asterisk in Australia, does anyone want to test my patch
for Australian ringtones? (as we all hate USA one). It's been sitting there
for over a month waiting for testing.
http://bugs.digium.com/bug_view_page.php?bug_id=259
You'll also need to change your
Is it possible to generate indications based on the context? And what
abou SIP devices, they generate their own tones
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Adam Hart
Sent: Thursday, October 23, 2003 2:20 AM
To: [EMAIL
Steven Critchfield wrote:
I'm sorry, either I didn't explain myself well enough or you
misunderstood. I have no connection to the telco at my home. I have a
T100p and a channel bank making extensions in my home. I have a cable
modem connecting me to the outside world.
ohh.
That's different.
Hi!
Does * do DNS-lookups when outgoing calls are placed to a different
SIP domain? Can I call from sip:[EMAIL PROTECTED] to sip:[EMAIL PROTECTED]?
Can * work as a regular SIP proxy in that aspect?
Can * handle SIP URI:s that are complete SIP URI:s (sip:[EMAIL PROTECTED]) instead
of numbers only?
Hi
I'm running exactly the same setup. Asterisk is running on my FW/NAT/Router
with two interfaces. My local phones are situated behind the NAT and connects
to the outer interface of the */FW/NAT/Router. * is then connected to my
SIP providers (since I'm only using the SIP-part of *, PSTN
Matthew Simpson wrote:
The number of codecs is overwhelming to me.
What do ya'll consider the best codec for conserving bandwidth? [I realize
at the cost of quality]
Secondly, what do you think the best codec for voice quality is?
Yours,
Matthew
Its hard to tell you which codec is best
There's a fix I'd like :
When you pick up the phone and press callers then send, any standard
human being would like to have the number shown sent, not another one ...
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
From my experience iLBC is unbeatable on lossy and slow links. I have
been in situations where no other codec (GSM, Speex, G.729) worked and
iLBC was still fairly usable.
Jan.
On 22-10 23:29, Matthew Simpson wrote:
The number of codecs is overwhelming to me.
What do ya'll consider the best
Hello.
I have this issue, when I pickup a call that is ringing in a SIP
Phone, it keeps ringing.
There is bug #116 that mention something about these, but it does
not seem to be resolved , at least, not yet.
Anybody else has seen it behavior ?
Thank's.
Ing. Angel Gomez Garcia wrote:
Hello.
I have this issue, when I pickup a call that is ringing in a SIP
Phone, it keeps ringing.
There is bug #116 that mention something about these, but it does
not seem to be resolved , at least, not yet.
Anybody else has seen it behavior ?
Hi Jean-Christophe,
Yes i think you do :)
If you put the canreinvite=no then * won't try to connect the two sip phones
together but indeed will behave as proxy taking in the outside stream and
passing it on to the phone on the inside and vice versa.
Greetings,
Tjardick
--
Tjardick van der
WipeOut wrote:
Ing. Angel Gomez Garcia wrote:
Hello.
I have this issue, when I pickup a call that is ringing in a SIP
Phone, it keeps ringing.
There is bug #116 that mention something about these, but it does
not seem to be resolved , at least, not yet.
Anybody else has seen it
Wow, how soon do you think it will take for them to actully get the ilbc-
enabled firmware into our hands?
On Wed, 22 Oct 2003 20:53:57 -0600, John Brown (CV)
[EMAIL PROTECTED] wrote:
Grandstream and Global IP Sound have inked a deal in which
Global IP Sound will provide its royalty free
I am currently trying just about the same exercise - getting asterisk to
work in a uClibc environment. I've gotten it to compile by removing the
enum support - some warnings but else a clean compile. It can also run and
everything seems ok - except that when I try to connect a sip useragent -
Ing. Angel Gomez Garcia wrote:
WipeOut wrote:
Ing. Angel Gomez Garcia wrote:
Hello.
I have this issue, when I pickup a call that is ringing in a SIP
Phone, it keeps ringing.
There is bug #116 that mention something about these, but it does
not seem to be resolved , at least, not
WipeOut wrote:
Ing. Angel Gomez Garcia wrote:
WipeOut wrote:
Ing. Angel Gomez Garcia wrote:
Hello.
I have this issue, when I pickup a call that is ringing in a SIP
Phone, it keeps ringing.
There is bug #116 that mention something about these, but it does
not seem to be resolved ,
Thomas Dingermann wrote:
WipeOut wrote:
Ing. Angel Gomez Garcia wrote:
WipeOut wrote:
Ing. Angel Gomez Garcia wrote:
Hello.
I have this issue, when I pickup a call that is ringing in a
SIP Phone, it keeps ringing.
There is bug #116 that mention something about these, but it
does
Working with X-Lite, iLBC is unuseable. The sound is completely scrambled,
even without using Asterisk between 2 clients. While trying to use SPX,
X-Lite connects to Asterisk, but no sound at all.
Else, the GSM 06.10 is quite fair and works for everybody.
Jean-Christophe
- Original Message
Hi all
I've been trying to make * work with IAXtel to no avail, all seems ok in
the config but am not getting anywhere
This is what I'm getting from console (user/pass/dest # changed for
obvious reasons):
DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT
on RTP to 0
Hi!
I am trying to music on hold but I am having all sorts of problems with it.
I am running RH9 and the latest version of Asterisk as of yesterday.
Here is what I did to test it:
1. I manually deleted the mpg123 softlink to mpg321.
2. I downloaded mpg123-0.59r-1.n0i.src.rpm, compiled and
Hi
anyone has a suggestion about what do I need to do to solve this? Any
comments appreciated.
You can solve this in one of 2 ways.
1. put ppp back into your kernel.
2. remove ppp references from zaptel.
You should use #2. Edit Makefile and comment out the PPP option.
Steven
For anyone running RH9 with a recent version of *, if you are using
music on hold
I would be interested in what version you installed or compiled. The
version described
below is not working properly and leaves core files in the mohmp3
directory for me. :(
Clif Jones wrote:
I am trying to
...
Still: When I call my Asterisk box (which has a fixed IP and is located
within a university network) using X-Lite I get choppy sound to say the
least. In fact I can hear only the first half second of what I am
supposed to hear followed by permanent silence. Note that this * box has
no
We are seeing high utilization (70% +) with astersisk when idle on Debian 3.01 with a
Fritz PCI chan_CAPI.
The PC is a P900 256MB and it is only used for asterisk. The problem seems to start
within a few hours of asterisk starting and the fix is to kill and restart asterisk.
Any ideas
On oh323.conf I have:
codec=G711U
frames=20
But while connecting it gives me in log:
1:18.636 H225 Caller:8111de8 H245 Capability merge result:
Table:
G.723.1(5.3k){hw} 1
Set:
0:
0:
G.723.1(5.3k){hw} 1
Which I don't have, so the connection is dropped. Any known solutions?
Hello,
How to turn off native bridge in
Asterisk.
Is it possible ??
Bart
Hi,
I have just set up IAXTEL connectivity and I get a similar response.
I have tried to call 1800 and the * says that a connection to
IAXTEL is made but I get no ringing or anything from the remote end.
Does anyone have a 1700XXX number I can call, or can somebody call mine,
Hi all:
I've no response for the last question with the same subject. Please excuse
me for the extreme length of this mail, but I send 2 SIP traces.
I have problem with * and 5300, when the incoming and outgoing call are
routed thru the same SIP gateway (AS5300). Do I need to set an special
hi guys, i got a TDM400P FXS card an everything is
fine except for this when i do modprobe wcfxs
, the linux shows 2 TigerJet Network Inc Model 300
128k. i don't know why it is showing 2 of them. or is
that what it is?
ERROR:
Freshmaker version: 63
Freshmaker passed register test
ProSLIC on
Have you another ISDN card in your system ?
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von C M
Gesendet: Donnerstag, 23. Oktober 2003 14:06
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] wcfxs error
hi guys, i got a TDM400P FXS card an everything
Hi,
-Original Message-
Does anyone have a 1700XXX number I can call, or can
somebody call mine,
17008188820.
You can try my IAXtel number: 17005821001
I'm not at my desk right now, but the number has a little IVR with some
options to test.
Best regards,
Florian Overkamp
Hi,
The Win32 binary of Gastman crashes on Windows 2000
SP4. Same case on all my machines, no error, no log.
Although, the CVS version works great on
Linux.
Is it anybody who knows how to compile it with
mingw32 ? Or better, could anyone, who already has mingw32 installed, make a
binary
Here are some ideas for anyone with some extra time on there hands.
SIP phones on call pickup either use a special REGISTER or you can
place a call with the magic extension and have the switch hang up
on you and immediately call you back. With the second option, you
could dial *8, Asterisk could
Hello, all
How can I write sound file with external
G723.1 codec ( actually I have CISCO that can make H323 call to Asterisk box
with G723.1or G729 codec ) I am trying to start Record application by
specifying inextensions.conf
[writesound]
exten = s,1, Answer
exten =
Can anyone please point me toward the source/binary (linux and Win32) for
Gastman??
Robert
Hi,
The Win32 binary of Gastman crashes on Windows 2000 SP4. Same case on all
my machines, no error, no log.
Although, the CVS version works great on Linux.
Is it anybody who knows how to compile it
Is Gastman at a usable level now? Have there been recent modifications?
Last time I tried using it, it was causing strange errors on asterisk
(in combination with the quad span TDM card and 2 PCI FXO's)
-Original Message-
From: rnc Info Lists [mailto:[EMAIL PROTECTED]
Sent: Thursday,
If you mean how to get the CVS version you just have to do a checkout from digium.
export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
cvs login
password: anoncvs
cvs co gastman
regards
Mickey Binder
-Original Message-
From: rnc Info Lists [mailto:[EMAIL PROTECTED]
Sent: 23.
by using the usual CVS:
# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
# cvs login - the password is anoncvs.
# cvs co gastman
or the released source and binary is available at:
ftp://ftp.asterisk.org/pub/telephony/gastman/
Jean-Christophe
- Original Message -
I didn't have any major trouble. Some functions seem unsupported by now, but
I did play for more than 1 hour by monitoring calls, forcing redirections
and connections, and it seems to be allright for such jobs. Although, I
don't use a TDM card, but SIP and CAPI.
Jean-Christophe
- Original
Totally awesome, sounds like something I want to do myself :)!
-Original Message-
From: Robert Hajime Lanning [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 22, 2003 11:21 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Is the X100P a WinModem?
quote who=Steve Sobol
Ok. And
Hello,
Can somebody tell me what does it means
?
I just installed my codec g729 with two
channels.
[codec_g729b.so] = (Annex B (floating point)
G.729/PCM16 Codec Translator) == Detected 2 licensed G.729
transcodersWARNING[16384]: File translate.c, Line 219 (calc_cost):
Translator
On Mon, Oct 20, 2003 at 03:55:47PM -0500, Tom Walsh wrote:
Also trunking requires that some sort of timing device (digium card or
ztdummy) be in place for trunking. Otherwise trunking is disabled.
What does ztdummy require to work? kernel compile options? Does it work
on SMP systems?
okay someone find me on IRC where I can ssh in and i'll really try to fix
this.
Mark
On Thu, 23 Oct 2003, WipeOut wrote:
Ing. Angel Gomez Garcia wrote:
Hello.
I have this issue, when I pickup a call that is ringing in a SIP
Phone, it keeps ringing.
There is bug #116 that
Hi. I am using #include to include a file in
extensions.conf. I have an agi perlscript which modifies the #included
file and then forces an asterisk reload with 'system("asterisk -rx
reload")';
After the reloadI use set_context,
set_extension and set_priority to tell asterisk where I want
hello * users,
i am using a zplex 10b(8fxosand 16fxs) connected to a T100 digium card,
RH8.0.
soon after running * i get this warning
file chan_zap.c,line 4155 (ss_thread):CallewrID returned with error on
channel 'zap/7-1'
then executes the other applications itself.
--executing
Ben,
I have an X100P used, at present, largely for outgoing calls. It shares the
single incoming POTS line with a number of analog phones. Is it possible to
talk the X100P (Zap/1) to answer a ringing call only if I ask it to? I'd
like to use only the SIP phone in my office, but let the
What global variable?
I am also trying to deal with this issue
Thanks
Lee Goodman
- Original Message -
From: Surajee Ratnayake [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 22, 2003 10:45 PM
Subject: Re: [Asterisk-Users] Artificially Limiting IAX Calls
-
Same here. Could someone who has the latest tarball post a mirror?
thanx
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 23, 2003 1:07 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] A software FAX modem
Steve Underwood wrote:
You can
Hi everyone,
Anyone knows if it is possible to remotely delete a specified
message from voicemail storage. I would like make it possible to delete a voice
message that was forwarded to the uservia email after he finishes
listening on a pc. He could click on a link in the email message body
On 23/10/2003 12:38, David J Carter wrote:
I have tried to call 1800 and the * says that a connection to
IAXTEL is made but I get no ringing or anything from the remote end.
Does anyone have a 1700XXX number I can call, or can somebody call mine,
17008188820.
Hey likewise. I
Hey likewise. I can only seem to ring myself (1 700 873 7731). I just
tried
you and got nothing.
I'm able to call IAXtel numbers.
If you want, give 1 700 625 4069 a call. You'll get our IVR and won't
disturb anybody. If you want an echo test, enter '7899' when prompted for an
extension.
On Thursday, October 23, 2003 8:52 AM, [EMAIL PROTECTED]
[SMTP:[EMAIL PROTECTED] wrote:
hello * users,
i am using a zplex 10b(8fxosand 16fxs) connected to a T100 digium
card,
RH8.0.
soon after running * i get this warning
file chan_zap.c,line 4155 (ss_thread):CallewrID returned with
On Thu, 2003-10-23 at 08:44, Muhammad Nasim wrote:
Hi. I am using #include to include a file in extensions.conf. I have
an agi perl script which modifies the #included file and then forces
an asterisk reload with 'system(asterisk -rx reload)';
After the reload I use set_context,
On Thu, 2003-10-23 at 09:07, Edwin Silva wrote:
Same here. Could someone who has the latest tarball post a mirror?
thanx
Done, These are from when I downloaded them for my use. These are the
originals, and not the slightly modified versions to make it work on my
system.
Clif,
For anyone running RH9 with a recent version of *, if you are using
music on hold
I would be interested in what version you installed or compiled. The
version described
below is not working properly and leaves core files in the mohmp3
directory for me. :(
We brought up a stock
I managed to get inbound IAXtel working by setting it up the wrong way
(i.e. [iaxtel] as the last entry, etc).
You can call my IVR system at 700-923-3645. Extension 2101 is for
interactive services including talking clock, and callerid readback.
Extension 2102 is for system services like echo
Thanks Steven :)
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 23, 2003 10:40 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] A software FAX modem
On Thu, 2003-10-23 at 09:07, Edwin Silva wrote:
Same here. Could someone who has
-- Original Message --
From: Shaun Ewing [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Fri, 24 Oct 2003 00:38:12 +1000
Hey likewise. I can only seem to ring myself (1 700 873 7731). I just
tried
you and got nothing.
I am able to dial your number
hi guys, i got a TDM400P FXS card an everything is
fine except for this when i do modprobe wcfxs
, the linux shows 2 TigerJet Network Inc Model 300
128k. i don't know why it is showing 2 of them. or is
that what it is?
Freshmaker version: 63
Freshmaker passed register test
ProSLIC on
I have tried to call 1800 and the * says that a connection to
IAXTEL is made but I get no ringing or anything from the remote end.
Does anyone have a 1700XXX number I can call, or can somebody call mine,
17008188820.
Hey likewise. I can only seem to ring myself (1 700
Thanks for your reply. It seems I may be proceeding in the wrong direction.
I have a context with time arguments e.g. include
somecontext|18:00-9-00|mon-fri So the system behaves according to standard
office hours.
I want to be able to leave the office in office hours e.g at 13:00, dial a
number
Eric,
I managed to get inbound IAXtel working by setting it up the wrong way
(i.e. [iaxtel] as the last entry, etc).
You can call my IVR system at 700-923-3645. Extension 2101 is for
interactive services including talking clock, and callerid readback.
Extension 2102 is for system
Hi,
Thanks all for help.
Working on most 1700XXX numbers now in and out, but still no go on the
18X numbers, just tried the HP sales number for a test.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson
Sent: 23 October 2003
On Thu, 2003-10-23 at 10:51, Muhammad Nasim wrote:
Thanks for your reply. It seems I may be proceeding in the wrong direction.
I have a context with time arguments e.g. include
somecontext|18:00-9-00|mon-fri So the system behaves according to standard
office hours.
I want to be able to
Bartosz Jozwiak wrote:
Hello,
How to turn off native bridge in Asterisk.
Is it possible ??
Bart
canreinvite = no
in each entry your sip.conf
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http://lists.digium.com/mailman/listinfo/asterisk-users
Take the output of lspci -v with a grain of salt. It is doing a
lookup in a local file to translate the numbers it reads off the
card into words like Tiger Jet Network Inc..
lspci is using only the first two of the four ID numbers to do the
lookup. and for whatever reason the 400P and 100P use
Help, I'm stuck. Lost in the woods.
I have one Asterisk running on FreeBSD outside on the Wild Internet.
One on the safe inside, behind a NAT firewall.
The inside server registers with IAX to the outer one and can place calls.
The outside one can't register to the one on the inside, since it
Louis-David Mitterrand wrote:
On Mon, Oct 20, 2003 at 03:55:47PM -0500, Tom Walsh wrote:
Also trunking requires that some sort of timing device (digium card or
ztdummy) be in place for trunking. Otherwise trunking is disabled.
What does ztdummy require to work? kernel compile options? Does it
Hello, are there any plans to support VoATM (AAL2) in Asterisk or is any work
being done in that domain? Thanks - DL
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Yes
Ing. Angel Gomez Garcia wrote:
Hello.
I have this issue, when I pickup a call that is ringing in a SIP
Phone, it keeps ringing.
There is bug #116 that mention something about these, but it does
not seem to be resolved , at least, not yet.
Anybody else has seen it behavior ?
what their costs are or what makes them successful. Armchair
businessmen are a dime a dozen; it doesn't help that everytime you
post to the list, you advocate products which will undercut Digium's
source of revenue.
Isn't this what Linux is about?
Every asterisk box helps to cause things
Title: Number of TDMoE Channels?
I was trying to establish a TDMoE span of 4 channels between two Asterisk servers. Machine A has a T100P to our PBX. Machine B has no Zaptel hardware. With 4 channels (em signalling) the red alarm never clears, and eventually machine A panics. With 24
WipeOut wrote:
Olle E. Johansson wrote:
Help, I'm stuck. Lost in the woods.
I have one Asterisk running on FreeBSD outside on the Wild Internet.
One on the safe inside, behind a NAT firewall.
The inside server registers with IAX to the outer one and can place
calls.
The outside one can't
WipeOut wrote:
http://www.voip-info.org/wiki-Asterisk+timer
This will not work on SMP systems (Multiprocessor), where the RTC clock
is used for SMP support.
Symetrical Multi Processing
Fixed. Thank you!
And maybe SMB file sharing needs timers too ;-)
/O
Johnson, Randy wrote:
-Original Message-
From: WipeOut [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 23, 2003 2:12 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX peers and NAT
Olle E. Johansson wrote:
Help, I'm stuck. Lost in the woods.
I have
Thanks Steve I'll try it with the global variable first and then later have
a go at the DBput etc.
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 23, 2003 5:00 PM
Subject: Re: [Asterisk-Users] agi script forcing asterisk reload
Hi. Can anyone help me with the
following:
[globals]
OFFICEHOURS
[internal]
exten =
*80,2,SetGlobalVar(OFFICEHOURS=100)
exten =
*80,2,SetGlobalVar(OFFICEHOURS=200)
[incoming]
exten =
Someone has proven chan_skinny with Cisco 7910?
I've got some problems with dtmf relay:
Oct 23 14:58:30 WARNING[-1533859520]: File
chan_skinny.c, Line 1710 (skinny_indicate): Don't know how to indicate condition
14
Thanks in advance,
Gus
On Thursday 23 October 2003 13:32, Ethan wrote:
what their costs are or what makes them successful. Armchair
businessmen are a dime a dozen; it doesn't help that everytime
you post to the list, you advocate products which will undercut
Digium's source of revenue.
Isn't this what Linux
I've tried to list various files and applications in Asterisk that includes passwords.
http://www.voip-info.org/tiki-index.php?page=Asterisk+password+files
If you know any other file or application with passwords, add to the Wikipage or mail
me
offlist so I can update.
Sometime in the future,
--- Ethan [EMAIL PROTECTED] wrote:
what their costs are or what makes them successful. Armchair
businessmen are a dime a dozen; it doesn't help that everytime you
post to the list, you advocate products which will undercut
Digium's
source of revenue.
Isn't this what Linux is about?
I'm about to download Festival source, apply the astrisk diff's, and
initiate basic testing. Thoughts are to download v1.4.3 (latest per
the fesitval website.
If anyone has an existing how-to, install notes, tips, or any suggestions
I'd greatly appreciate it. Direct email is fine if you'd rather
Hello,
How to go back with asterisk CVS
?
and what happens when then turn right around and expect digium to
support this?
personally, i think the $99 is high, but hey, i'm not the one
who's invested all my time/energy/$$ into this like Mark has.
just remember, the free/linux/etc *about* stuff is the base
software, don't start applying
On 23/10/2003 21:16, Rich Adamson wrote:
I'm about to download Festival source, apply the astrisk diff's, and
initiate basic testing. Thoughts are to download v1.4.3 (latest per
the fesitval website.
If anyone has an existing how-to, install notes, tips, or any suggestions
I'd greatly
I had some trouble getting festival 1.4.2 to compile and run on RH9. The issue
was (if I recall correctly) with the gcc (3.2.2) compiler. Something about
Templates has changed in the new C++ compiler (making it more conformant to
standards) causing the festival not to compile.
Anyways, I then
WipeOut wrote:
Olle E. Johansson wrote:
Here is basically the way mine is setup.. names changed to protect the
innocent.. :)
Maybe you can spot what you are missing..
PBX1- insidepbx (behind NAT)
---iax.conf--
register = user:[EMAIL PROTECTED] ; Server on static IP
[outsidepbx]
;For
BTW, the issue in my last email is also applicable to the speech_tools source.
Deepak
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 23, 2003 4:16 PM
To: Asterisk-users-list
Subject: [Asterisk-Users] Festival on RH9?
I'm about to download
I agree with the above 100%. In fact the best thing that could
happen to Asterisk would be for someone to figure out how to make FXS
cards priced at $10 per line. I'm thinking all that is really
required is full duplex sound card and a ringer. Ringers can be made
with a 555 timer IC, a
Bartosz Jozwiak wrote:
Hello,
How to go back with asterisk CVS ?
Use a -D switch..
For help try..
cvs -H checkout
or
cvs -H update
Later..
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
It's already been done. The X101P is a $10 winmodem, tested by me as of
last night.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Thursday, October 23, 2003 4:12 PM
To: [EMAIL PROTECTED]
Subject: Re:
My interrest is radio. I'd like to use Asterisk as a N-way audio
switch between a set of ham radios and to act as a transcoder between
a few of the ham-oriented VOIP systems like IRLP, Echo Lnk, Wires and
the like.
You know, radio stations pay $5000+ for Evantide units that allow call in
Coming from the [evil] Dialogic world (where even the drivers cost
money) the prices Digium is charging seem very reasonable. New
single-span Dialogic T1 interfaces cost at least three times ($1225 USD
was the best price I could find on the D/240PCI-T1) what the single span
Digium card costs.
Okay, the festival build with patches went fine, and starts fine.
I added
exten = 555,1,Festival,Testing one two three.
When I dial extension 555, the CLI indicates:
-- Executing Festival(SIP/3000-1d95, You are calling.) in new stack
== Parsing '/etc/asterisk/festival.conf': == Parsing
hell i just got a quote today for D240JCT-1T1 for $4500ish
Steven M. Sokol wrote:
Coming from the [evil] Dialogic world (where even the drivers cost
money) the prices Digium is charging seem very reasonable. New
single-span Dialogic T1 interfaces cost at least three times ($1225 USD
was
I've attached two SIP debugs in reference to bug #116. They are from
today's CVS build.
1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the
call. After which, SIP(2) rings for about 30 seconds then stops.
2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging
Rich:
Please see if festival_server is running as specified in:
http://www.marko.net/asterisk/archives/0209/0389.html
==
export PATH=$PATH:/usr/src/festival/bin
/usr/src/festival/bin/festival_server
==
Or test festival in bash...
Regards,
Gus
- Original
I have the following in my extensions.conf:
exten = 21,1,NoOp(${CALLERIDNUM})
exten = 21,2,GotoIf($[${CALLERIDNUM} = ]?21|4:21|9)
exten = 21,4,Playback(/etc/asterisk/interactive-services/no-callerid)
exten = 21,5,Wait(1)
exten = 21,6,Playback(/etc/asterisk/interactive-services/no-callerid)
exten
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