Displaying 20 results from an estimated 5000 matches similar to: "How to configure the voicemail message playback sequence"
2005 Oct 14
5
sip accounts
hi, i facing a problem here. in my sip.conf, i specify a account like this,
[1234]
type=friend
context=from-sip
username=1234
secret=1234
nat=no
canreinvite=yes
dtmfmode=info
mailbox=1234@default
disallow=all
allow=ulaw
so i am able to login with username 1234 and password 1234
but ther weird part is, i can also register as any number (account)
without having to specify in sip.conf. thus
2005 Sep 05
9
Asterisk Follow ME
Hi All.
I have notice a problem with FM feature (screen macros) on Asterisk CVS
version.
When call goes via IAX and calling part "accept the call" on Dial
command with option M, in macros context it's setting
MACRO_RESULT=CONTINUE, but anyway it hangups both channels.
If anyone faced with such problem please let me know. I need to know
whether it's bug or just configuration
2005 Oct 07
3
Digium G.729 codec modules updated
This evening I posted a new set of Digium G.729 codec modules to our FTP
server and web site, for Linux x86 and x86-64 processors. They were
built using GCC 4.0.1, and they now report the processor they were
optimized for when they are loaded.
The previous x86-64 module required a non-standard Asterisk binary
configuration, so this was corrected. In addition, there was only a
generic version
2006 Mar 02
3
Native music on hold - Error
I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have
[native]
mode=files
directory=/var/lib/asterisk/moh-native
And in sip.conf I have
musicclass=native
When I put call on hold this is what I get at CLI.
-- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack
2006 Apr 07
2
Announcing Astmanproxy 1.20
Greetings everyone,
I'm pleased to announce the release of Astmanproxy 1.20, the fast,
flexible proxy server for Asterisk's Manager Interface. Astmanproxy
allows you to communicate with multiple Asterisk boxes from a single point
of contact using a variety of I/O formats, now including support for
XML, HTTP, HTTPS, SSL, CSV, and the Asterisk-native standard format.
Astmanproxy is
2006 Jan 20
5
iDEFISK (mac iax2 softphone) release
]
Hey ho,
A few days ago we released the linux version of the phone, today we are
very happy to have the mac version ready for a little field test.
Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php
At the same time, we also put a newer version of the windows and linux
versions online.
Let us know how you feel about it, a more mac look (brushed metal) is
coming.
2006 Feb 27
5
res_features pickupexten
is where anyone who knows what is needed to get the pickupexten (*8)
running ?
gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff
I've activated it in features.conf (default *8) and also tested other
extensions
res_features.so is loaded
show features says:
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind
2006 Mar 22
2
Pickupexten not working
Hi group. I have huge problem. My pickup exten #8 isn't working.
This is what I have configured.
pbx*CLI> show features
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 #8
In sip.conf I have
callgroup=2
pickupgroup=2
For called party and same for person that is trying to pick up the call.
The person that is trying
2006 Apr 05
3
queue issue
Hi,
I have several queues configured at my call center for different support levels.
Today, something weird happened:
- A client called queue 1 and was answered by an agent
- The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf
- The user transferred the client to another Queue, by using the second channel and the XFer key of her
2005 Oct 05
3
SIP Attended Transfer using REFER and Replaces: headers
hey all,
am wondering if anyone has successfuly done a SIP attended transfer using
the REFER method (after an INVITE obviously) and the Replaces: header.
we're writing our own SIP UAC and the asterisk code seems to support it,
but we're not really sure if this is so.
we plan on the following call flows:
1. incoming call from exten 1111 is sent to our UAC with Dial()
2. our UAC makes
2006 Jan 31
5
Queue() with timeout=0
Hello,
i've recently switched over from 1.0.9 to 1.2.3.
I've experienced some (to me) weird behaviour.
This is the config for an example queue.conf:
[654]
wrapuptime=30
timeout=20
strategy=ringall
retry=5
queue-youarenext=queue-youarenext
queue-thereare=queue-thereare
queue-thankyou=queue-thankyou
queue-callswaiting=queue-callswaiting
music=default
monitor-join=yes
monitor-format=
2020 Mar 25
1
Asterisk 17.3: No VoiceMailMain when enabling IMAP and ODBC
Hello,
On a Debian Buster instance, I compiled Asterisk 17.3.0 from source.
I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using
classical File module (in modules;conf and voicemail.conf):
cd asterisk-17.3.0
...
make menuselect.makeopts
menuselect/menuselect --enable app_voicemail_imap menuselect.makeopts; done
menuselect/menuselect --enable app_voicemail_odbc
2001 Oct 19
2
wine 20010824 and quake
i have quake v1.06 installed and running fine under windows. however,
running it in wine gives a bunch of errors. see below:
prophet% wine --winver win98 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath
fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD. Try --winver nt40 or win31
!
prophet% wine --winver nt40 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath
fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD.
2005 Oct 10
2
DTMF Question (misunderstood '*' button)
Hi all!
I'm experimenting a strange problem in my Asterisk PBX:
I've got an Asterisk pbx in the office: I dial an external number; the dialled
number answers me correctly, but as soon as I press the '*' button (i.e. to
navigate through the menus or to enter a password) my Asterisk box put me on
hold.
(CLI transcription follows:
-- Executing
2004 Oct 05
1
problems withX100P-Nochanneltyperegisteredfor'Zap'
For reference...
http://www.voip-info.org/wiki-Asterisk+zap+channels
Not sure it is relevant but go ahead and remove the spacing on the
channel line so it will read....
channel=>1
Here is the original incoming context you showed.
[incoming]
exten => s,1,Answer ; Answer the line
exten => s,2,Playback,demo-thanks ;for playing a file
The Playback looks malformed based upon the wiki
2004 Jun 26
2
Newbie needs help
I've been banging my head on a brick wall for about an hour now trying
to understand why the following doesn't work (which is even provided as
an example in the distribution!).
The goal is to create a voicemail-only extension not associated with a
phone. I'd rather not have an extension dedicated to VoicemailMain(),
so I would like the user to be able to hit '*' during
2005 Jul 01
3
pattern matching based on callerid, not working
according to the wiki, I should be able to do this:
exten => _9./3003,1,Set(CALLERID(number)=2814444443)
exten => _9./3004,n,Set(CALLERID(number)=2814444444)
exten => _9./3005,n,Set(CALLERID(number)=2814444445)
exten => _9./3006,n,Set(CALLERID(number)=2814444446)
exten => _9.,n,Dial(SIP/${EXTEN:1}@mycarrier,30,wt)
and have the correct calleridnum's set for each extension based
2004 Oct 05
1
problems with X100P -Nochanneltyperegisteredfor'Zap'
You should see something like this.... (I have 8 channels)
tuxpbx*CLI> zap show channels
Chan Extension Context Language MusicOnHold
pseudo incoming en
1 incoming en
2 incoming en
3 incoming en
4 incoming en
5 incoming en
6
2006 Mar 04
2
Asterisk 1.2.5 Released
Asterisk 1.2.5 is now available for download on the ftp. See the
ChangeLog for details about what has changed.
ftp://ftp.digium.com/pub/telephony/asterisk/
As mentioned in the release announcement for Zaptel 1.2.4, our releases
now contain some extra files. The Asterisk release is available as
asterisk-1.2.5.tar.gz. However, there is also a patch against the
previous release as an option for a
2006 Mar 04
2
Asterisk 1.2.5 Released
Asterisk 1.2.5 is now available for download on the ftp. See the
ChangeLog for details about what has changed.
ftp://ftp.digium.com/pub/telephony/asterisk/
As mentioned in the release announcement for Zaptel 1.2.4, our releases
now contain some extra files. The Asterisk release is available as
asterisk-1.2.5.tar.gz. However, there is also a patch against the
previous release as an option for a