Marcel van Kaam, Fonetica wrote:
I had the same problem with installing addons. I checked out in the file
cdr_addons_mysql.c what the location of the asterisk.h must be and changed
the cdr_addons_mysql.c to the location of the asterisk.h file.
After this it worked. Also to be sure do: locate
I will see to check all my notes on Asterisk and see or I can put some
manuals online.
That would also be helpful for the newbie's.
Marcel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: donderdag 30 juni 2005 8:14
To:
I want to use Asterisk registering itself to a GK.
SIP phones are registered to Asterisk
H323 are registered to the GK
I want to:
1. make calls from SIP (Asterisk) -- H323 (GK)
2. use Meetme to make a conference call for both types of phones
I got on the GK, login and password, IP of GK,
Anyone know of any good VoIP phones for about $50.
You should probably try to shell out a little more and get quality
phones, it really makes a difference. Alternatively, I've found that
SIPURA ATAs sound great (they have really good echo cancel too).
Lastly how do you fell about the 1
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jamie Carl
Sent: Thursday, 30 June 2005 2:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] X100P connected as extension to
Panasonic 616EASA-PHONE
Title: Sipura 3k answers then immediate busy signal
I have a sipura 3000 that I am using just to send calls to my mac asterisk server. When you call the phone it rings, answers, and then goes right to a busy signal. Any ideas?
Thanks for your help!
Jane
At the console in verbose mode I get:
This is my musiconhold.conf and my
folder:
[EMAIL PROTECTED]:/etc/asterisk# less
musiconhold.conf[classes]default =
quietmp3:/var/lib/asterisk/mohmp3;loud =
mp3:/var/lib/asterisk/mohmp3;random =
mp3:/var/lib/asterisk/mohmp3,-z;unbuffered =
mp3nb:/var/lib/asterisk/mohmp3;quietunbuf =
Hi,
app_pickup, app_pickupchan, app_pickdown, app_steal are your friend
in BRIstuff. ;)
best regards
Klaus
Am Mittwoch, den 29.06.2005, 10:09 -0500 schrieb Brian West:
Go get app_intercept from www.pbxfreeware.org
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan:
Extension 6000 is an extension that i created to try MusicOnHold, and the 2391
is the caller of rxtensions 6000.
I have the problem with two sip client, if one of them set on hold the other
one, i can't hear any sounds also if i see Started music on Hold clasess.
I also tryied this
Hi,
I'm using the new AMP which provides a call recording. The options of
recording call Always and Never are well working.
But how to use the On-Demand option ? Should I press a pad ? Is this
configured in the featuremap of features.conf ? Why my modifications in
that features.conf have no
Change from default to manual. I did that
and it helped.
Later I changed to madplay and set that as
default. Below my line from musiconhold.conf:
default =
custom:/usr/share/asterisk/mohmp3/,/usr/bin/madplay --mono -R 8000
--output=raw:-
Marcel
-Original Message-
I claim to be NO expert, but is there a chance that the 'ztdummy' driver is
also being loaded? I'm thinking it might cause a timing conflict of some
kind...I may be way off here, but I'd still check for something as simple as
that...
Chris Coulthurst
[EMAIL PROTECTED]
|-Original
On 06/29/05 11:51 Matthew Boehm said the following:
Hey gang,
I've been able to use sipp to produce some call volume on our asterisk
server. The server has no problems handling 50 simul calls. But then again,
no RTP is being done. I tried to use the rtp echo ability of sipp but that
i've
Anyone know if any of the ITSPs support Speex?
My initial testing would indicate not, but you never know. . .
Thx.
B.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi,
I'm trying to write a tool, which shows me the state of the current
calls. For this purpose I'm reading from Pipe the Asterisk output and
parse it... asterisk -vr | mytool
However, the problem ist how to get the information about who got this
call in the group. The Zap channels are
last I heard that feature wasn't supported
how are you getting it to work?
- Original Message -
From: Alexis F. [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, June 30, 2005 12:29 AM
Subject: [Asterisk-Users] AMP - recording call
Hi,
I'm using the new AMP which
In the ISDN case, setcallerid or hidecallerid can be
configured and I am aware that Asterisk doesn't
support that on analog line. My question is whethere
there is something like add-on script or hardware that
will do the job. The teleco company provide the
callerid service, but no private number
I have already read the referenced wiki and you are
right, Asterisk doesn't support the setcallerid or
hidecallerid . I am hoping to find some help like AGI
script or some hardware that can add this feature to
the analog line.
--- Flu [EMAIL PROTECTED] wrote:
Thank you all so much for your responses.
I would have responded sooner but for the time zone I am in.
Do we apply all 3 patches:
astcc.txt [^] (1,839 bytes) 06-06-05 19:43 [Delete]astcc1.txt [^] (1,918 bytes) 06-29-05 20:17 [Delete]astcc.patch [^] (820 bytes) 06-29-05 20:39 [Delete]
or just
It depends on your telco, in the UK on an analog line we can prefix it
with 141, so in that case yes, Asterisk can do it. You to find out from
your telco whether a caller with a standard handset can do anything to
control callerid with your telco.
Steve
-Original Message-
From: [EMAIL
I have already read the referenced wiki and you are
right, Asterisk doesn't support the setcallerid or
hidecallerid . I am hoping to find some help like AGI
script or some hardware that can add this feature to
the analog line.
The callerid service is provided on the line, but the
teleco doesn't
On Thu, Jun 30, 2005 at 09:57:36AM +0200, Martin Czarnowski wrote:
Hi,
I'm trying to write a tool, which shows me the state of the current
calls. For this purpose I'm reading from Pipe the Asterisk output and
parse it... asterisk -vr | mytool
Or use the manager API, which was meant
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Martin Czarnowski
|Sent: Thursday, June 30, 2005 12:58 AM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Resolving groupcalls
|
|
|Hi,
|
|I'm
Oops, sent that last one prematurely!
How about the accountcode setting? You could get user information from
that, right?
Maybe you could send:
Asterisk -rx 'show channels'
..and when you get the data, you'd know which channels are up and alive
(full names).
You could then re-run the command
Look for a book coming out from O'Reilly in a few months or so. It's
being offered on Amazon (pre-release). I can't remember the names of
the authors, but I'm pretty sure they are some of the Asterisk
developers.
Jeff
On Wed, 2005-06-29 at 19:37, Robert Goodyear wrote:
On Jun 29, 2005, at
Hello my name is Axel Pache and i and some kolleges are working on a
callcenter solution. We use ICD to manage skill based routing. But now
we got some problems integrating FOP, for example FOP doesnt
acknowledge the ICD-queues right. I have to use a normal asterisk
queue to get FOP
Hi,
I would like to connect my * with two
quad bri card: one to my Hipath pbx and other to the telco.
I successfully installed the cards to
asterisk patched with bristuff, now how I tell asterisk that I have 2 qud
bri card.
I searched documentation in google with
no success.
Someone can help me ?
Hi
I have been trying for a while to find a way to get an SMS send when I
receive a voicemail into my asterisk system. I don't want to send an
SMS if the caller doesn't leave a message. I have voicemail.conf set
up to email and delete.
302 = 302,Website Sales,[EMAIL
At 14:37 06/06/2005 +0300, you wrote:
hello all,
I would like to create an application where a file is played repeatedly
in the background while two parties are having a conversation. Does
anyone know of a way to achieve this?
I have been looking into the ManagerAPI to redirect the call to a
voiptalk.org seem to be pretty reliable for both incoming and outgoing
calls... I've been using them for at least 6 months for light volume calls.
Steve Foy wrote:
Hi,
I'm looking for a reliable provider to use mainly for outgoing calls in the
UK, incoming isn't so much of a worry as I
*nod* For a long time . had to be the last item in a pattern match
(if it was used at all)
Chris Modesitt wrote:
That worked perfectly, this behavior must have changed recently because I
tried that 6 months ago and it did not work:-)
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL
i tried to write a pattern to usa destination and that
was 1* it worked well but when i tried to specify the
number of digits i wrote 1NXXNXX it didnt work.
so what i must to write in case to specify the number
of digits of the destination.
__
Leon Sun wrote:
Hi, All
I would like to send SIP NOTIFY to SIP UA from Asterisk. Is it possible?
I appreciate if you can provide detail sample of message header.
To turn on MWI using an unsolicited NOTIFY you could use the following with
sipsak Replace the word 'yes' with 'no' to turn
Hi;
Have a BUDGETONE-100 and using it with asteriskProblem
occurs when I dial message centerMessage center does not accept tones (password
for example) that I dial,
Behaves as I do not dial any number and asks for the
password againChanged the DTMF Mode from in-audio to RTP(RFC2833)
it
Matt wrote:
Use account codes? How does one use account codes, and of which
account codes do you speak?
Don't use Caller*ID for billing. Use account codes, which is supported
pretty much everywhere in Asterisk.
If you want me to hold your hand I usually require dinner and drinks first.
Hi,
is there anybody who knows if asterisk writes both in Master.csv and in
MySql tables (I installed asterisk-addons) at the same time?
Or once addons are installed, asterisk doens't write Master.csv anymore?
I'd like to have them all.
Thanks
Giorgio Incantalupo
Barton Fisher wrote:
exten = 0099,5,Dial(ZAP/g2,20,r}/*${CALLERID}*${EXTEN}*)
Try:
exten = 0099,5,Dial(ZAP/g2/*${CALLERID}*${EXTEN}*,20)
or even
exten = 0099,5,Dial(ZAP/g2/*${CALLERID}*${EXTEN}*,20,r)
r is almost never needed. Asterisk will provide a ringing sound to
the caller if it
Michael Blood wrote:
I receive this error on the asterisk console and it is pretty much
ALWAYS coming up.
Sometimes there will be a break where it does not display.
This happens when Asterisk receives corrupt data from the card. The
most common cause of this is some device or driver locking
I believe this may solve your problem,
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+grandstream+budgetone
works for me.
-yair
On 6/30/05, Betül Gözlükoğlu [EMAIL PROTECTED] wrote:
Hi;
Have a BUDGETONE-100 and using it with asterisk…Problem occurs when I dial
message
I can confirm that its probably a very good idea to wait for that book,
one of the authors is also the person doing the asterisk documentation
project. (Leif Madsen aka blitzrage)
I bought one of the other books, in an e-book format and was very
disappointed with what i got. (Both the contents
I am using
rawplayer:
default =
custom:/var/lib/asterisk/mohmp3/raw,usr/bin/rawplayer
as in: http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it
However, the music is too loud.
Without having to rerecord it, is there a parameter like quietmp3 that can be
used with the above to
Always and never work. The on demand hasn't been implemented yet.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of hank
Sent: Thursday, 30 June 2005 3:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hank,
There is, look again on the [EMAIL PROTECTED] sourceforge site.
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of hank
Sent: Wednesday, 29 June 2005 11:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello Everybody,
Has anybody gone through the C code of the Asterisk? Like Dial ( ) is a
registered application of Asterisk, I wish to build an application of my own.
So how does do it in Asterisk? And also could anybody
Please tell me an IRC for asterisk and also the server that
I
Hi,
Perhaps Im wrong but if you use g729
with no translation (pass-thru) you cant hear voice mail. Set your codec
to gsm or g711 and try again.
Erdem HAKI [EMAIL PROTECTED]
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Betül Gözlükoğlu
Sent: Thursday,
On 6/30/05, Mark Charlton [EMAIL PROTECTED] wrote:
Hi
I have been trying for a while to find a way to get an SMS send when I
receive a voicemail into my asterisk system. I don't want to send an
SMS if the caller doesn't leave a message. I have voicemail.conf set
up to email and delete.
On Thu, 2005-06-30 at 12:43 +0200, Giorgio Incantalupo wrote:
Hi,
is there anybody who knows if asterisk writes both in Master.csv and in
MySql tables (I installed asterisk-addons) at the same time?
Yes
Or once addons are installed, asterisk doens't write Master.csv anymore?
No
I'd
Sorry I should have paid more attention to your post :)
You are already using fastsms!
oops.
;exten = h,1,HasNewVoiceMail(30${dialed_extn})
;exten = h,2,goto(h,100)
;exten = h,102,DeadAGI(fastsms|44000|Caller ${CALLERID} left
a new voice
mail at ${DATETIME} on Sales extn
In the USA it's possible to send a voicemail to most mobiles now via an
email gateway eg [EMAIL PROTECTED]
Is it possible in asterisk (and [EMAIL PROTECTED]) to send a voicemail
waiting email to 2 email addresses at the same time?
Cheers,
Dean
-Original Message-
From: [EMAIL
Dear All,
I am using Linux-High Availability between two Asterisk servers, everything is
fine but I do have one problem with this, When a server fails and the other
assumes the ip address and start asterisk on server 2, the ip phone must
re-register themselves again, otherwise the phones won't
On Thu, 2005-06-30 at 16:33 +1000, Terry H. Gilsenan wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jamie Carl
Sent: Thursday, 30 June 2005 2:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I am able to hear voice mail after I entered
the passwordBut the problem is I can not enter password when the
budgetone is configured as SEND DTMF in-audio
It accepts via RTP or via SIP,but this
time , I can not dial internal numbers over telephony system(santal dahilisi)
From:
Dear All,
I am using Linux-High Availability between two Asterisk servers,
everything is fine but I do have one problem with this, When a server fails
and the other assumes the ip address and start asterisk on server
2, the ip phone must re-register themselves again, otherwise the phones
are
Yes you can modify the config file. So I dont see this as a con, Ive
run AMP over Asterisk for months now, to me it was just easier, and Ive
customized it also for our environment. There is a huge argument for
dialplan programming, it can be very very cumbersome.
Cons
- you're forced to
Hello Everybody,
Can any one guide me in developing
an application in Asterisk? Just as Dial ( ) is a registered application. So
that I can develop modules of my own.If anybody has done it, kindly please let
me know.
Regards,
Bharat M. Sarvan
That is the point it is no working. How do you replace these services ?
Is someone using an old version ? From which one isn't any more support ?
Alexis F.
last I heard that feature wasn't supported
how are you getting it to work?
- Original Message - From: Alexis F. [EMAIL
Hi ,
I am trying to use a telephone Atcom AT323
Both in SIP mode and in IAX mode, I have a lot of echo on a large number of
number called (NOT ALL, it depends on the network I reach)
I see that using in /etc/asterisk/capi.conf
echosquelch=1
;echocancel=1
echotail=64
Everithing is really good
Hello Everybody,
Can any one guide me in developing an application in Asterisk? Just as Dial ( )
is a registered application. So that I can develop modules of my own.If anybody
has done it, kindly please let me know.
Regards,
Bharat M. Sarvan
On Thu, 2005-06-30 at 18:35 +0530, Bharat M. Sarvan wrote:
Hello Everybody,
Can any one guide me in developing an
application in Asterisk? Just as Dial ( ) is a registered application.
So that I can develop modules of my own.If anybody has done it, kindly
please let
Bharat M. Sarvan wrote:
Hello Everybody,
Can any one guide me in developing an application
in Asterisk? Just as Dial ( ) is a registered application. So that I can
develop modules of my own.If anybody has done it, kindly please let me know.
No. You have sent this at
And the UK although the PRI provider can either override or supply
it for you and you are normally limited (unless you've signed an
agreement) to DDI numbers directly provided by the PRI provider.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Hello, i used sipsak to send the NOTIFY message:
sipsak -H 192.168.10.1 -f sip.txt -s sip:192.168.10.11 -v
asterisk listens on 192.168.10.1 and the phone with MWI is the 192.168.10.11
i send a couple of different messages to the phone but i always get back:
received ICMP packet (type: 3,
Wow!
Thanks to everyone for the overwhelming support for hosting the Daily
Asterisk News!
It is now being kindly hosted by Josh Colp (File).
Let me know if you have any problems/questions.
--
Cheers,
Matt Riddell
___
Did u installed mpg123 0.59r ?
Giordano
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Adam
RobinsInviato: giovedì 30 giugno 2005 13.01A: Asterisk
Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List -
Non-Commercial DiscussionOggetto: RE:
You only apply astcc.patch. Feedback can be given by signing up on
bugs.digium.com and commenting there on the patch. Could you please
provide the output from the console when you run astcc? The only reason
for it to be left marked in use is if the prog. is crashing. Other than
that it
Juan Luis Moyano wrote:
Darren Wiebe wrote:
As I said I would, I have posted some screen shots and my astcc database
dump in the wiki. Please see:
http://www.voip-info.org/tiki-index.php?page=ASTCCGuide for links to the
info.
Darren Wiebe
[EMAIL PROTECTED]
Darren, I'm very thankful
[EMAIL PROTECTED] wrote:
Hi,
I would like to connect my * with two quad bri card: one to my Hipath
pbx and other to the telco.
I successfully installed the cards to asterisk patched with bristuff,
now how I tell asterisk that I have 2 qud bri card.
The driver will recognize them. Next,
Dear All,
I am using Linux-High Availability between two Asterisk servers,
everything is fine but I do have one problem with
this, When a server fails and the other assumes the ip address
and start asterisk on server 2, the ip phone must
re-register themselves again, otherwise the phones are
No, I am not using mpg123 at all.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano
GrandisSent: Thursday, June 30, 2005 9:35 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: R:
[Asterisk-Users] Music oh hold
Did u installed mpg123 0.59r ?
Thanks for the suggestions, everyone!
I'd previously tried PickUP() but it'd always give me a 503 as well,
with this familiar error:
chan_sip.c:7402 handle_request: Nothing to pick up
Almost as if I still had pickupchan= defined in features.conf, but I did
make sure to comment that out and
Does anyone in here know how to setup auto negotiation between g729 and
g711ulaw on
a cisco 5400? I would imagine it would be the same on a 3660.
The problem I am having is natively the call is setup for g729 however
when the call is transferred
to voicemail it uses ULAW so when the cisco
I created some scripts to logrotate. I am having a problem. After I do
it, I am sending kill -HUP to the process
its not using the newly created messages file again. Could someone help
me out with how I can rotate asterisk's
log's without killing the process?
Dear All,
I
am using Linux-High Availability between two Asterisk servers, everything is
fine but I do have one problem with
this, When a server fails and the other assumes the ip address and start
asterisk on server 2, the ip phone must
re-register themselves again, otherwise the
See below for the console output when Iapplyastcc.patch
[EMAIL PROTECTED] agi-bin]# patch -p0 astcc.patch(Stripping trailing CRs from patch.)patching file astcc.agi[EMAIL PROTECTED] agi-bin]#
The problem still persists, card IN-USE after each call and no record generated in astcc cdr database.
I tried using your working astcc.agi file instead of mine, but that failed to work too.Darren Wiebe [EMAIL PROTECTED] wrote:
Juan Luis Moyano wrote:Darren Wiebe wrote: As I said I would, I have posted some screen shots and my astcc databasedump in the wiki. Please
Could someone help
me out with how I can rotate asterisk's
log's without killing the process?
Does restarting the syslog service help?
# service syslog restart
or
# /etc/init.d/syslog restart
___
Asterisk-Users mailing list
If your phones are setup to connect to the asterisk box by
name, then a smart DNS server can just point phones to the backup box after
failure. However, since asterisk running on the backup box doesn't know
about the phones, this is only half the solution
From: Mohamed A. Gombolaty
Geoff Manning wrote:
Could someone help
me out with how I can rotate asterisk's
log's without killing the process?
Does restarting the syslog service help?
# service syslog restart
or
# /etc/init.d/syslog restart
Asterisk doesn't use syslog by default.
Try running: asterisk -rx logger
Title: Re: [Asterisk-Users] [Fwd: Asterisk Balancing solution]
Maybe... Wait for it... Realtime? Keep the information in a database that is shared or replicated between all servers.
- Joshua Colp.
On 6/30/05 10:54 AM, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote:
Dear All,
I am using
Damon Estep wrote:
Dear All,
I am using Linux-High Availability between two Asterisk servers,
everything is fine but I do have one problem with
this, When a server fails and the other assumes the ip address and
start asterisk on server 2, the ip phone must
re-register themselves again,
Guillermo Salas M wrote:
On Thu, 2005-06-30 at 16:33 +1000, Terry H. Gilsenan wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jamie Carl
Sent: Thursday, 30 June 2005 2:48 PM
To: Asterisk Users Mailing List - Non-Commercial
Asterisk doesn't use the syslog daemon tho does it? I thought it
did internal logging to a file.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL PROTECTED]
Hi
[EMAIL PROTECTED] uses the following file:
/var/log/asterisk/*log {
missingok
rotate 5
weekly
create 0640 asterisk asterisk
postrotate
/usr/sbin/asterisk -rx 'logger reload' /dev/null 2 /dev/null
endscript
}
/var/log/asterisk/full {
missingok
rotate 5
daily
create
That should work. Or just configure the phones to re-register every
minute.
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 30, 2005 9:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] [Fwd:
Router#conf term
Router(config)#voice class codec 99
Router(config-class)#codec preference 1 g711ulaw
Router(config-class)#codec preference 2 g729br8
Router(config-class)#codec preference 3 g729r8
Router(config-class)#end
Router(config)#dial-peer voice 2000 voip
thank you I will give that a try.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hilton
Williams
Sent: Thursday, June
Use Realtime and host the database on a separate machine.
This should solve most of your problems.
~Vamsi
On 6/30/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote:
Dear All,
I am using Linux-High Availability between two Asterisk servers, everything
is fine but I do have one problem with
Hi Bharat. I think that does not exists such a thing like an Asterisk
Dev App howto :p , so for now the best way to learn i think is check
out the apps/ directory on Asterisk Sources. Also check the app.h file
in includes/
in case you were wondering, i havent done any Asterisk App, just
modified
Asterisk doesn't use the syslog daemon tho does it? I thought it
did internal logging to a file.
My mistake, you are correct (both of you actually!)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Set DTMF to info
that fixed my problem.
On 6/30/05, Erdem HAKİ [EMAIL PROTECTED] wrote:
Hi,
Perhaps I'm wrong but if you use g729
with no translation (pass-thru) you can't hear voice mail. Set your codec
to gsm or g711 and try again.
Erdem HAKI –
[EMAIL PROTECTED]
On Thu, 2005-06-30 at 16:47 +0200, Hilton Williams wrote:
Hi
[EMAIL PROTECTED] uses the following file:
snip
On Mandriva I've got a file called asterisk in /etc/logrotate.d which
contains:-
var/log/asterisk/event_log /var/log/asterisk/queue_log
/var/log/asterisk/messages
Alex Vishnev wrote:
Hello,
I am using asterisk CVS-head from 6/28. I am also using chan_oh323 that
comes with asterisk. I tried to place a call from h323 device into asterisk.
in extensions.conf, I routed the call to my sip phone. The sip phone was
already registered with asterisk. all the
On 6/30/05, Mike Dent [EMAIL PROTECTED] wrote:
Here is what mine looks like:-
exten = h,103,DeadAGI(fastsms|44797149|Uberetchs Voicemail.
Caller ${CALLERID} left a new voice mail for sales a
t ${DATETIME}|441524342XX)
Hope that helps,
Mike
I started out using the solution like you
Hello
Any body was tested LIBISUP. and price of LIBISUP packet ?. how much to purchased it from digium.
if posible, tell me where are LIBISUP beta release to test with asterisk and my postoffice of my country (vietnam).
Best regards.___
Asterisk-Users
Why not set the register interval to something like 30-60 seconds?
In my experience, even if I shutdown asterisk and load it up again, the
phones can place calls right away, just not receive them until they
re-register in 60 seconds.
--- Mohamed A. Gombolaty [EMAIL PROTECTED] wrote:
Dear All,
Matthew Boehm wrote:
Damon Estep wrote:
Dear All,
I am using Linux-High Availability between two Asterisk servers,
everything is fine but I do have one problem with
this, When a server fails and the other assumes the ip address and
start asterisk on server 2, the ip phone must
he must have just added it all I saw last I looked was the forums
thanks for this ifnormation
take care
hank
- Original Message -
From: Dean Collins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, June 30,
I think this is a weak point in asterisk.
It doesn't even have a means of email notification if IAX or SIP
registration fails.
This would need to be added to the list of priorities.
But I'm not sure who to address to.
Most phone are controlled by their own software interface and have the
On Thu, 2005-06-30 at 10:53 -0400, John Novack wrote:
Guillermo Salas M wrote:
On Thu, 2005-06-30 at 16:33 +1000, Terry H. Gilsenan wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jamie Carl
Sent: Thursday, 30 June 2005
(I am looking for a
cheat sheet - or good documentation for Audiocodes MP-1xx series and
*)
I am a newbie *
user.
I am using this as
FXO
(Bridge from analog
POTS lines to IP (I think I used the correct terminology) (basically they are
incoming phone lines from my phone company))
I
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