Hello
I'm trying to install Asterisk in a Linux server without compiler, yum
or apt-get.
Is this possible? Where can I find the pre compiled binaries?
Thanks
Regards
Joao Pereira
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On Sun, Mar 14, 2010 at 08:43:09AM +, Joao Gomes Pereira wrote:
Hello
I'm trying to install Asterisk in a Linux server without compiler, yum
or apt-get.
Is this possible? Where can I find the pre compiled binaries?
Debian, Fedora, OpenSUSE and Ubuntu include binary packages of Asterisk.
I tried,
[agents]
exten = 1,hint,SIP/0317998975SIP/0317998985
exten = 1,1,Dial(SIP/0317998975SIP/0317998985)
and
queue add member Local/1...@agents to 0317998989
sip*CLI queue show 0317998989
0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s
holdtime, 109s talktime), W:0, C:1,
queue add member Local/1...@agents to 0317998989 penalty 1 as Magnus
Benngard state_interface hint:1...@agents
1,hint,SIP/0317998975SIP/0317998985
exten = 1,1,Dial(SIP/0317998975SIP/0317998985)
and
queue add member Local/1...@agents to 0317998989
sip*CLI queue show 0317998989
0317998989 has 0
Hi,
Did a test with Local, exten = 1234,1,Dial(Local/1...@agents)
[agents]
exten = 1,hint,SIP/0317998975SIP/0317998985
exten = 1,1,Dial(SIP/0317998975SIP/0317998985)
When calling 1234, both 0317998975 and 0317998985 rings
when answering in 0317998985, 0317998975 stops ringing, all fine but
Glad to see I was able to point you in the right direction.
On 03/14/10 23:56, Magnus Benngård wrote:
queue add member Local/1...@agents to 0317998989 penalty 1 as Magnus
Benngard state_interface hint:1...@agents - did the trick :)
On Sun, 14 Mar 2010 11:38:13 +0100, Magnus Benngård
Thx Rob!
On Mon, 15 Mar 2010 00:53:06 +1100, Rob Hillis wrote:
Glad to see I was able to point you in the right direction.
On 03/14/10 23:56, Magnus Benngård wrote:
queue add member Local/1...@agents to 0317998989 penalty 1 as Magnus
Benngard state_interface hint:1...@agents
On Sun,
I've got Asterisk 1.6 bridging to an Avaya using H323. The Avaya is
autoanswering calls to music (as expected) and audio seems fine, but I see
this error on bridging:
WARNING[8833]: chan_ooh323.c:1054 ooh323_indicate: Don't know how to
indicate condition 20 on ooh323c_o_2
Is this a warning I
I see the following: a stuck process
12651 ?S 0:00 gzip -9 /var/log/asterisk/messages.2
and then:
asterisk*CLI core show channels
Channel Location State Application(Data)
Logger/rotates...@default:1 Down(None)
On Sun, Mar 14, 2010 at 10:30:33AM -0400, Richard Kenner wrote:
I see the following: a stuck process
12651 ?S 0:00 gzip -9 /var/log/asterisk/messages.2
and then:
asterisk*CLI core show channels
Channel Location State Application(Data)
1. Any chance you're out of disk space?
Nope:
FilesystemSize Used Avail Use% Mounted on
/dev/mapper/VolGroup00-LogVol00
11G 5.2G 4.9G 52% /
/dev/sda1 99M 38M 56M 41% /boot
tmpfs1002M 0 1002M 0% /dev/shm
2. Why not
Hi,
we are using Asterisk 1.6.0.22 on Ubuntu 9.04 together with a Digium
Wildcard B410 and dahdi-linux-complete-2.2.0+2.2.0.
Internal calls between hard- and softphone are no problems.
When I call the asterisk with my cell phone and choose an extension
where PLAYBACK() plays tt-monkeys, I can
Hello All,
I'm trying to do a fresh installation on Ubuntu Server 9.10 (Karmic)
64-bit but I am getting error when make config is trying to install
the init script... Here is the output: - Can anyone help me please...
Thanking in advance...
Cheers,
Nitesh
On Sun, 2010-03-14 at 13:32 -0400, Nitesh Divecha wrote:
make[1]: Entering directory
...
/sbin/chkconfig --add dahdi
insserv: warning: script 'S20theserver' missing LSB tags and overrides
[snip other warnings]
insserv: There is a loop between service rsyslog and apache2 if stopped
insserv:
Thanks Tjoen for reply... Here is complete process when I execute make
config... Do i just comment out this line: -
CHKCONFIG := $(wildcard /sbin/chkconfig)
r...@bill:/usr/src/dahdi-linux-complete-2.2.1+2.2.1# make config
make -C linux all
make[1]: Entering directory
On Sun, Mar 14, 2010 at 01:32:09PM -0400, Nitesh Divecha wrote:
Hello All,
I'm trying to do a fresh installation on Ubuntu Server 9.10 (Karmic)
64-bit but I am getting error when make config is trying to install
the init script... Here is the output: - Can anyone help me please...
Thanks Tzafrir
Kinda solve the issue by commenting out following: -
CHKCONFIG := $(wildcard /sbin/chkconfig)
UPDATE_RCD := $(wildcard /usr/sbin/update-rc.d)
ifeq (,$(DESTDIR))
# ifneq (,$(CHKCONFIG))
#ADD_INITD := $(CHKCONFIG) --add dahdi
# else
ifneq (,$(UPDATE_RCD))
Hello all,
My folks would like to play a message to answering machines automatically after
hanging up the phone. So, when the caller dials the number of the callee,
hears an answering machine, they would like to enter a code on the phone and
hang up. After the hangup the message plays to the
I want callers to enter a queue and then hear music on hold.
does anyone have notes on how to integrate queuing to a dial plan that uses moh?
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New to
Hello All,
I have configured Asterisk to act like a Softswitch (routing calls in
and out) but I m facing one issue... I want to route advance all fail
calls with ISDN 34 (SIP 503) whenever calls are failed due to some
reasons...
Is there any way to flash back SIP 503 on all failed calls from
One possibility is to have the dialplan create another channel, bridge
the two channels together, then play the message, but I´m not sure of
the best method of accomplishing this.
Haven't touched 1.6.2 yet, so I'll speak in 1.4 terms:
No need for a conference, look at ChannelRedirect. Also
Hi!
after hanging up the phone. So, when the caller dials the number of the
callee, hears an answering machine, they would like to enter a code on the
phone and hang up. After the hangup the message plays to the callee and
disconnects.
There is an even simpler way: Just blind transfer the
On Thu, 2010-03-11 at 22:24 +0100, Christian Victor wrote:
2010/3/11 Eric Wheeler aster...@ew.ewheeler.org:
Hi Eric,
I have four spare TE411P but never used bonded T1 or T1 for data at
all. If you can supply me with instructions I could test if your plan
works the way you need.
Great!
Hi
I made a conclusion too soon, i couldnt count down while music was
playing back.
I'm running out of time, really need help.
Thanks
QuyPs
On Sat, 2010-03-13 at 08:42 +0700, Pham Quy wrote
Here again, the script should be described as
- Caller call to a
Hi there
I remember to ask this question in the past but now I have thought of
something little bit difference. While I understand that asterisk dialplan
accept the call to be answered[ Answer() ] in the dialplan, I wanna know if
this is possible;
i. A call on legacy PBX, extension to extension
Hi
I have the following scenario
A. A PBX on location A with network 192.168.1.1 with extension range 1XXX
and connected to the PSTN Network via the E1
B. Another PBX on location B with network 172.30.18.1 with extension range
2XXX and connected to the PSTN Network via the E1
I need to configure
Would you like the advice in all caps?
On 03/15/2010 01:20 AM, RESEARCH wrote:
Hi there
I remember to ask this question in the past but now I have thought of
something little bit difference. While I understand that asterisk dialplan
accept the call to be answered[ Answer() ] in the
Dns srv might be the solution for you.
Jai
www.didforsale.com
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From: RESEARCH
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To: asterisk-users@lists.digium.com
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] High
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