Hi,
this is my /etc/modprobe.d/zaptel:
options torisa base=0xd
alias char-major-196 torisa
install tor2 /sbin/modprobe --ignore-install tor2 /sbin/ztcfg
install torisa /sbin/modprobe --ignore-install torisa /sbin/ztcfg
install wcusb /sbin/modprobe --ignore-install wcusb /sbin/ztcfg
Hi Waldo,
Doesn't * record to .gsm file initially and then convert these to .wav
later? I may be totally off the mark here, and if I am, I welcome the
correction.
In that case, why not leave the files in .gsm format instead of
translating them into another lossy format? Obviously if * records
my iax.conf:
[callshopcompany]
type=peer
host=213.61.187.150
username=X
secret=X
disallow=all
allow=gsm
--- Angelito Manansala [EMAIL PROTECTED] wrote:
can you paste you iax.conf
On 11/8/05, chawki hammoud [EMAIL PROTECTED]
wrote:
Hi:
I have been having this problem for
Ok, I hope finally it will arrive to the list...I posted it twice...-- Forwarded message --From: Gabor Horvath
[EMAIL PROTECTED]Date: 2005.11.06. 10:35Subject: differences between chan_capi and chan_capi-cmTo: Asterisk-Users list asterisk-users@lists.digium.com
Can you tell me
Does it say I use them? I only said that voipjet comes through at 19ms,
so I disagree about the TOS. (didn't know about it anyway :)
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, November 07, 2005 5:42 PM
To:
Yes but i want to enable access for all users from that ip address. I
don't want to write every user in sip.conf.
greetings
mk
2005/11/7, Peter Petrov [EMAIL PROTECTED]:
Miloš Kocbek wrote:
I want to enable access to some context in asterisk without authentication.
In sip.conf:
[username]
Kernel 2.6 + CentOS 4.1
All work perfectly but Hangup() dont work
in log/asterisk/full
Nov 5 11:58:04 DEBUG[8299]: zt_hangup(Zap/1-1)
Nov 5 11:58:04 DEBUG[8299]: Set option AUDIO MODE, value: ON(1) on Zap/1-1
Nov 5 11:58:04 DEBUG[8299]: Hangup: channel: 1 index = 0, normal = 9,
callwait =
Hello!
We consider purchasing Digium Wildcard for E1 connectivity. Wildcards
are pretty expensive pieces of silicon for small shop like ours. And we
have no previous experience with E1 communications.
What Wildcard do we need? How can we estimate our needs? How many
clients (approx.) can
hi all,
can i have a softphone which will showing the activate users and their
sip number(sort of phone book for globally use)??? does xten provides
such a feature? thanks
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users
Hello All,
I have a bri and iwsh to get CID w/name, however, even though Verizon
has told me that CID/Name is on the circuit, I still only get ANI. No
cid or cid/name.
Anyone know if it is possible to get cid over bri?
I am not sure if the issue could be in the eicon firmware or something
else,
Thanks for the replies. I have realised that I can catch the execution
after the agi statement (if it fails) in the h priority, which I then
use to play an error message to the caller.
As you suggested, I am setting a variable in the agi script so that the
h priority knows whether the agi
Branko Samardzic wrote:
Any idea on how to enforce native format into read and write streams?
In the peer definition (iax.conf or sip.conf) put:
disallow=all
allow=CODEC_YOU_WANT
where CODEC_YOU_WANT is something like gsm, g729, ulaw etc (keep it to one
entry at both ends and you're
Hi,
Digium have Wildcard for FXO/FXS connections (i.e., telephone lines)
and E1/T1 cards such as the TE110p. There're a few things you might
want to consider:
1) TE110p is much more expensive
2) it is too much for a small shop. Concurrently supports upto 15
incoming and 15 outgoing calls (or 30
Hello,
When I try to use this:
http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html
for sensing a fax (putting a file sample.call in the
/var/spool/asterisk/outgoing/) the call is made but after picking it up,
asterisk disconnects. What can be a reason? I'm using 1.0.9
Dear All,
I am facing a problem in compiling the add-ons for the mysql, though the files
are downloaded correctly and checked and I tried different mirrors even the cvs
but yet I get those errors :
app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory
cdr_addon_mysql.c:38:19: mysql.h:
Hello,
As Asterisk do not work with SRTP, i'm finding a SRTP/RTP proxy.
Any idea?
thanks
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Asterisk-Users@lists.digium.com
Hi,
While a104d install on asterisk 1.2 and CVS-HEAD
patch for zaptel.c failed.
Is it avaiable not yet?
Thanks.
__
Yahoo! Mail - PC Magazine Editors' Choice 2005
http://mail.yahoo.com
___
Hi!
We are running an * with 3 sip providers. Provider 1
works perfect, provider 2 also. But the 3:rd one is a problem. All seems normal
until we try to make a call. The phone rings by the called party and picks is
up and hear only silence. The caller (local extension on the *) still
Bartosz Piec wrote:
Hello,
When I try to use this:
http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html
for sensing a fax (putting a file sample.call in the
/var/spool/asterisk/outgoing/) the call is made but after picking it up,
asterisk disconnects. What can be a
It's a problem with bristuff that has been there for quite some time.
If you load the modules in the wrong order it will kernel panic the box.
I have been bitten by it many times, very frustrating if you are working
on a remote box
I manually load all modules now too
On Tue, 8 Nov
shenanigans wrote:
I was interested in getting feedback from current mail group users.
There is a limit to the number of times you can post this...
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
On Tuesday 08 November 2005 11:34, Hugh Jackman wrote:
Hi,
Digium have Wildcard for FXO/FXS connections (i.e., telephone lines)
and E1/T1 cards such as the TE110p. There're a few things you might
want to consider:
1) TE110p is much more expensive
2) it is too much for a small shop.
Sorry. Forgot to say that
if I connect an ip phone directly to the provider it works without problwm
Anders
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Anders Svensson
Sent: den 8 november 2005 11:09
To:
asterisk-users@lists.digium.com
Subject:
nobody has an answer here !!
--- harry gaillac [EMAIL PROTECTED] a écrit :
Hello,
Where may i find documentation about SIP domain
support and dnsmgr.conf ,
Harry
___
Appel
nobody has an answer here!
--- harry gaillac [EMAIL PROTECTED] a écrit :
Hello,
I configure Polycom ip300 for presence but when
status
change notify is no sent to subscriber !?
Why ?
Regards
Harry
Can you tell me what are the main differences between chan_capi
(http://www.junghanns.net/en/chan_capi.html), and chan_capi-cm (
http://sourceforge.net/projects/chan-capi)
chan_capi-cm is directly derived from the last development source of
chan_capi.
It does contain lots of fixes and
Hi,
Is there a way to detect (in the dialplan) if a SIP peer is registered
with the server ?
I am using macros to dial to extension, becuase i dont want to define
each extension in the dialplan, and, for example, my numbers are 8xx , i
want to know if a peer exists/registered before
Anders Svensson wrote:
Hi!
We are running an * with 3 sip providers. Provider 1 works perfect,
provider 2 also. But the 3:rd one is a problem. All seems normal until
we try to make a call. The phone rings by the called party and picks
is up and hear only silence. The caller (local
Matt Riddell napisał(a):
Did you install spandsp?
Yes, I have installed libtiff, spandsp, txfax and rxfax.
The problem now is that asterisk doesn't disconnect but when I try to
receive the fax, nothing happens. Fax (PSTN) is just waiting for receive
and after some time it finishes the call.
Hi,
basically chan_capi-cm is a fork from chan_capi. But since chan_capi is not
developed any more, chan_capi-cm has more features, is more stable and works
with newer versions of Asterisk too.
The main difference for the user is the change in capi.conf and the dial()
syntax, which is shown
There is no implementation in chan_capi-cm for CIDName yet, but if it is
available via CAPI messages we can add this.
Can you provide a log?
(A log of chan_capi-cm with 'set verbose 5' and 'capi debug', as well as a
mlog from Eicon card)
Armin
On Tue, 8 Nov 2005 [EMAIL PROTECTED] wrote:
Hello
Do you have silence suppression enabled on your clients? Asterisk can't work with silence suppression. Take a look at http://bugs.digium.com/view.php?id=5374 , the patch works fine to me, now I'm able to set SS and save bandwidth.
On Mon, 2005-11-07 at 19:41 -0800, Chris Tracy wrote:
I
Bartosz Piec wrote:
Matt Riddell napisał(a):
Did you install spandsp?
Yes, I have installed libtiff, spandsp, txfax and rxfax.
The problem now is that asterisk doesn't disconnect but when I try to
receive the fax, nothing happens. Fax (PSTN) is just waiting for receive
and after some
Elmar Haneke wrote:
I would suggest chan-capi-cm for any configuration.
You know which quadbri cards it works with?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free
Miloš Kocbek wrote:
Yes but i want to enable access for all users from that ip address. I
don't want to write every user in sip.conf.
So use no secret and host=x.x.x.x where x.x.x.x is the IP address you want to
allow from.
--
Cheers,
Matt Riddell
harry gaillac wrote:
nobody has an answer here!
Actually someone asked for you config details.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
I posted to the list with this issue a few weeks ago, but nothing really
came of it. Either I'm missing something obvious (for which I apologize in
advance) or this is a pretty serious issue between Asterisk and the SIP
devices connected to it.
I have 12 SIP phones at a particular site all
I recently resurrected an old athlon system and put CentOS 4.2 on
it to play with asterisk. First I tried asterisk-1.0.9, now I'm using
1.2.0-b2. Both have the same audio issues that have me stumped.
I looked through all the lists and forums and the closest I could
get
all you have to do is manually apply the patch before Setup and it
will patch fine(apply the patch in the 'zaptel' directory of wanpipe's
source directory to your zaptel source).
MATT---
On 11/8/05, Jason Kim [EMAIL PROTECTED] wrote:
Hi,
While a104d install on asterisk 1.2 and CVS-HEAD
We are running an * with 3 sip providers. Provider 1 works perfect, provider
2 also.
But the 3:rd one is a problem. All seems
normal until we try to make a call. The phone rings by the called party and
picks is
up and hear only silence. The caller (local
extension on the *) still gets
On Wed, 9 Nov 2005, Matt Riddell wrote:
Elmar Haneke wrote:
I would suggest chan-capi-cm for any configuration.
You know which quadbri cards it works with?
Any which support CAPI interface.
- Eicon Diva Server (all)
- AVM C4
- ...
Armin
--
Cheers,
Matt Riddell
Bartosz Piec wrote:
Matt Riddell napisał(a):
-- Attempting call on SIP/[EMAIL PROTECTED] for application
txfax(/root/testfax.tif) (Retry 1)
Channel SIP/yyy-3c49 was answered.
Faxing on a VoIP channel is not recommended. Read the following:
Yes but if i write
[test-user]
host=x.x.x.x
then only users test-user will able to make calls i want that every
username is allowed to call
greetings
mk
2005/11/8, Matt Riddell [EMAIL PROTECTED]:
Miloš Kocbek wrote:
Yes but i want to enable access for all users from that ip address. I
Hello,
Sorry here are my sip.conf and extensions.conf
in fact when polycom ip300 send subscribe to buddies
these one send back notify but nothing else when
status change
Regards
Harry
--- Matt Riddell [EMAIL PROTECTED] a écrit :
harry gaillac wrote:
nobody has an answer here!
Actually
Chris Bagnall wrote:
If 2 calls come in only a second or two apart, the first one will cause the
dial command to be executed, and when the second call comes in, it'll go to
voicemail because *all* the SIP phones report themselves as busy (because
they're ringing for the first call).
Is there
Since this is my DID, I want the line to ring as normal but allow a user
to breakout and ultimately get an outgoing line.
In this way the DID line would function as a normal telephone line. A
point lost on many responders!
I don't want to have to go into voicemail to breakout since I don't
Hi friends,
during the installation, I receive that problem, but I've installed
both Flex and, of course, C/C++ libraries.
My OS is CentOS 3.6, completely updated.
Any ideas???
Thanks
-
Compiling WANPIPE WanCfg Utility ...
Failed!
!!! WANPIPE WanCfg Compilation
Don Pobanz wrote:
I just checked out asterisk 1.2b2 for zaptel, libpri, asterisk and
asterisk-sounds. Zaptel and libpri compile fine with a 'make clean'
and 'make install'. However even after a make clean, the asterisk
'make install' does not finish on my redhat 7.3 system.
I would suggest chan-capi-cm for any configuration.
You know which quadbri cards it works with?
I'm using an Eicon-Diva-Server 4BRI.
Elmar
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Hi,
FaberK wrote:
during the installation, I receive that problem, but I've installed
both Flex and, of course, C/C++ libraries.
My OS is CentOS 3.6, completely updated.
Any ideas???
Thanks
-
Compiling WANPIPE WanCfg Utility ...
Failed!
!!! WANPIPE WanCfg
Title: Hiss
Whoo Hoo! I managed to get * up and running last night.
My most pressing problem is that there is a considerable amount of hiss heard by the called party when using a SIP phone (xten or GXP-2000). Ive tried two different computers with two different headsets, and the hiss still
Hello,
Is it possible to add a frontend groupware with
asterisk in order to Provide send receive fax to mail,
sms to mail, voice messages .
Asterisk or openpbx could be the server of the unified
messagerie .
click to dial contact in address book ,...
Harry
Drop the incoming calls into a call queue.
Is it not the case that in order for calls to go into a queue, they must be
answered first? Is it possible to drop calls into a queue before they're
answered (by asterisk)?
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is
harry gaillac wrote:
Is it possible to add a frontend groupware with
All is possible, you're only limited by your imagination. (always wanted
to say this :p)
I'm not sure there's a(n Open-source) project like this already.
Cheers..
___
Jeffrey Macko wrote:
Whoo Hoo! I managed to get * up and running last night.
My most pressing problem is that there is a considerable amount of
hiss heard by the called party when using a SIP phone (xten or
GXP-2000). I’ve tried two different computers with two different
headsets, and the
On 11/08/05 20:53 FaberK said the following:
Any ideas???
i believe the answer is in your email.
Please contact Sangoma Tech. at 905 474-1990
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
On 11/08/05 20:54 Robert Stanford said the following:
What version of gcc are you using ?
though this is documented in the UPGRADE.txt file, i believe it should have
been highlighted much more clearer. this bugbear has bitten quite a few
people who're unaware that gcc 3.x is the minimum
Is it possible to get Asterisk to issue a Playtones when an outgoing call is
answered? The examples indicate what happens when an incoming call is answered.
/Obelix
This message was sent using IMP, the Internet Messaging Program.
Hi Florian,
yes, I have Flex available:
whereis flex
flex: /usr/bin/flex /usr/share/man/man1/flex.1.gz
other ideas?
2005/11/8, Florian Overkamp [EMAIL PROTECTED]:
Hi,
FaberK wrote:
during the installation, I receive that problem, but I've installed
both Flex and, of course, C/C++
i appear to misplaced my password for my cisco 7960 SIP Phone. Does anyone know the procedure to recover this? I have read in the past that you can use cisco or Cisco but this does not appear to work.
Thanks in advance.
___
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Paul wrote:
I get the hiss and noise with softphones using all headsets I have tried
so far. I don't get it with grandstream budgetone 101 phones or phones
connected to ata's.
Then it's likely to be your sound card. Try using a nice usb headset (not the
cheapest you can find)
--
Cheers,
Ok. What does sip show subscriptions from the CLI show you?
On 11/8/05, harry gaillac [EMAIL PROTECTED] wrote:
Hello,
Sorry here are my sip.conf and extensions.conf
in fact when polycom ip300 send subscribe to buddies
these one send back notify but nothing else when
status change
Regards
Obelix wrote:
Is it possible to get Asterisk to issue a Playtones when an outgoing call is
answered? The examples indicate what happens when an incoming call is
answered.
It would have to be done by the remote machine. Unless you want to play a
sound to callee once connected:
Some Dial
Rich Adamson wrote:
I recently resurrected an old athlon system and put CentOS 4.2 on
it to play with asterisk. First I tried asterisk-1.0.9, now I'm using
1.2.0-b2. Both have the same audio issues that have me stumped.
I looked through all the lists and forums and the closest I
What about egroupware !
Harry
--- Kristof Hardy [EMAIL PROTECTED] a
écrit :
harry gaillac wrote:
Is it possible to add a frontend groupware with
All is possible, you're only limited by your
imagination. (always wanted
to say this :p)
I'm not sure there's a(n Open-source) project like
Hi Folks,
After doing extensive reading it seems that I am more confused than when
I started out.
I have an Asustek ISDNLink (P-IN100-ST-D) BRI card. I know that it has a
Winbond W6692 chipset, but there is much confusion in my head regarding
whether to use bristuff (which seems to work with the
Hi,
FaberK wrote:
Hi Florian,
yes, I have Flex available:
whereis flex
flex: /usr/bin/flex /usr/share/man/man1/flex.1.gz
Hmm, nope sorry :P. You can try to mail or call Sangoma, their support
is pretty good from what I've seen so far.
Florian
___
Is the ambient noise in the room high?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Tuesday, November 08, 2005 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Anyone written an LCDProc client for Asterisk?
It occurs to me that as many of these systems run headless in the back
of a closet a small LCD display could tell you what's going on at a glance.
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
harry gaillac wrote:
nobody has an answer here !!
Where may i find documentation about SIP domain
support and dnsmgr.conf ,
The problem is that dnsmgr is new and not finished, so there is not much
documentation yet.
Re the SIP domain support, I don't know, there is the announcement here (
chawki hammoud wrote:
Hi:
I have been having this problem for sometime that I am
not able to solve and I hope someone can help.
I can make VOIP calls between my Asterisk box and my
VOIP provider using sip channel without a problem. But
when I attempt to make a call using IAX, the call
Connected to Asterisk 1.2.0-beta2 currently running on
serveur1 (pid = 1553)
Verbosity is at least 3
serveur1*CLI sip show subscriptions
Peer UserCall ID Extension
Last state Type
192.168.0.21 86 2127e5fd-5f 84
Idle
harry gaillac wrote:
What about egroupware !
We use it, although there is no simple click to install installation package
for Asterisk integration.
The idea is to use flash operator panel to load a url when each extension is
dialed. And for click to dial, I use call files.
--
Cheers,
Matt
Hello,
I have a problem with my Sipura 2000.
The problem is that it does not accept any change in the configuration.
When I access to it, via browser or phone, and make any change, after
clicking submit all changes all the changes I made dissapear and teh
configuration remains with the
Ultimately this turned out to be a red herring as well. dialparties.agi
just does a database dip to figure out which extensions are forwarded
and then builds a dialstring based on whats left. It then returns to the
Asterisk dialplan and the extensions are still dialed in the normal way.
I stopped
Polycom User wrote:
i appear to misplaced my password for my cisco 7960 SIP Phone. Does
anyone know the procedure to recover this? I have read in the past
that you can use cisco or Cisco but this does not appear to work.
Thanks in advance.
Is this phone setup using tftp? If so, I
I just posted a few addons for the AMP users ...
These are several routines I found necessary for my system 1: Speed Dials
revised my way (AMP front end into DB), 2: Intercom in business, 3: Group
Paging in business, 4: Cisco phone display (XML) of internal directory list
from AMP extensions DB.
Get a Duxbury PCI ISDN card that has the HFC-S chipset, its type approved TE2003/013 and there is enough support on the wiki at
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+zaphfc+installdiff=24Cant find much reference to winbond+asteriskCost is also ±R200 each.Rob
On 11/8/05,
Chris Bagnall wrote:
Drop the incoming calls into a call queue.
Is it not the case that in order for calls to go into a queue, they must be
answered first? Is it possible to drop calls into a queue before they're
answered (by asterisk)?
Yes,
But your problem is stemming from the
Hamish Whittal wrote:
I have an Asustek ISDNLink (P-IN100-ST-D) BRI card.
[..]
This is not a card compatible with the bristuff.
I don't know about the availability of the hfc-cards in your part of the
world, but they are very inexpensive in Germany (around 30 EUR ~ 30 USD)
I am loathe to
Mark Phillips wrote:
Anyone written an LCDProc client for Asterisk?
It occurs to me that as many of these systems run headless in the back
of a closet a small LCD display could tell you what's going on at a glance.
Yes, I have :)
Still ironing out some bugs, but fire away if you have
Last night, my Atlas 550 failed big-time. Adtran, to their great credit, is
overnighting me a new one even though it is out of warranty. I am working
around this on my Asterisk box by plugging in my PRI directly into my TE110P
and receiving faxes with Asterisk where they used to be received with
Why the estension s dont' start?
In extensions.conf
[default]
exten = s,1,Answer
exten = s,2,Playback(invalid)
exten = s,3,Hangup
In sip.conf
[general]
context=default
___
Yahoo! Mail: gratis 1GB per i messaggi e
thanks Matt for your answer
Does asterisk-1.2-stable will provide this features ?
Harry
PS:
Who are the main developpers for the sip channels ?
--- Matt Riddell [EMAIL PROTECTED] a écrit :
harry gaillac wrote:
nobody has an answer here !!
Where may i find documentation about SIP domain
Ok. It looks like you've got most of the basic configurations setup
correctly. Let's setup a trace and then have you repeat your steps so
we can see in better detail what might be wrong.
In your logger.conf file make certain you have the following line:
full =
how to use #include to all files in /etc/asterisk/customdir ?
in v1.0.9
#include /etc/asterisk/customdir/*.conf
doesnt work
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The only way is if you are using DHCP to get an IP address
to the phone. If you are, then you can have it point the phone to a TFTP
server with config files with a new password. If you are using a static
IP, then you are out of luck. I opened up a TAC case about a year ago, and
that is what
i had ( or still have ) the same problem. Im running asterisk as
asterisk:asterisk, but dont know why, the new voicemails are saved as
root:root with 700 permissions, so i made a quick workaround, i added
the following line in sudoers file:
%lighttpd ALL=(root)NOPASSWD: /usr/bin/chmod -R 755
After rebooting my asterisk server (1.2B2) I could still call my Sipura
SIP phones from outside (via cell phone). But I have a customer with
two Cisco SIP phones...I don't know the exact model...those two phones
could not be reached. The message in the console:
Nov 8 10:03:34 NOTICE[4207]:
It is set by your SIPMAC.cnf file.
phone_password: password ; Telnet/Console Password
On Tue, 2005-11-08 at 08:51 -0500, Polycom User wrote:
i appear to misplaced my password for my cisco 7960 SIP Phone. Does
anyone know the procedure to recover this? I have read in the past
that you can
Fabio Montemaggiore wrote:
Why the estension s dont' start?
Do you get an error in the Asterisk console?
A good thing to read is the Asterisk Book which you can download for free from
one of the mirrors provided here:
http://www.sineapps.com/news.php?rssid=1044
--
Cheers,
Matt Riddell
What do you see on asterisk console?
(asterisk -vc)
El mar, 08-11-2005 a las 15:38 +0100, Fabio Montemaggiore escribió:
Why the estension s dont' start?
In extensions.conf
[default]
exten = s,1,Answer
exten = s,2,Playback(invalid)
exten = s,3,Hangup
In sip.conf
Hilton,
AFAIK, you can optionally record in gsm. However, I think * won't do
it natively. It will do -in and -out wav files, soxmix them together
and then convert them to gsm. I'm offloading all of that to a
different machine and just leaving * to create the raw -in and -out
wav files.
Matt Riddell napisał(a):
Maybe you could make an extension that you can dial which will run txfax for
you. Then you can call it with a phone and see if you hear the fax tones.
I tried this :( I hear the fax signal but then nothing happens.
These lines are in asterisk console:
Nov 8
Asterisk in console don't show not all
--- José Luis Gómez [EMAIL PROTECTED] ha scritto:
What do you see on asterisk console?
(asterisk -vc)
El mar, 08-11-2005 a las 15:38 +0100, Fabio
Montemaggiore escribió:
Why the estension s dont' start?
In extensions.conf
Looks good except one problem I am having. The AMP script does not store
the info. It adds a blank speeddial. If I edit the database the AMP
script will show the correct info, but it never updates the fields.
- James
Paul wrote:
I just posted a few addons for the AMP users ...
These are
I use a newsreader pointed at gmane.org.
It is agregated and only uses my internet connection when I tell it to.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - --
On Tue, Nov 08, 2005 at 09:12:27AM +0100, gincantalupo wrote:
Hi,
this is my /etc/modprobe.d/zaptel:
Those are probably useless:
options torisa base=0xd
alias char-major-196 torisa
Those won't help you a bit if you run a ztcfg in your zaptel init script
anyway.
install tor2
After you place one in you MUST submit.
That is only when it is saved
Paul
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Armstrong
Sent: Tuesday, November 08, 2005 10:42 AM
To: Asterisk Users Mailing List -
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Armstrong
Sent: Tuesday, November 08, 2005 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] New package posted to Sourceforge
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