Displaying 20 results from an estimated 700 matches similar to: "Asterisk does not persist callgroup and pickgroup configuration."
2013 Mar 05
2
Redirect incoming call to SIP trunk.
Greetings.
I got two asterisk servers, one is connected to another via sip trunk. The
incoming calls are routed to the time period an if matches is transfered to
the designed extension. If don't, is redirected to a second time period.
Then, if the call matches the second time period it need to be transfered
to the trunk that directs to the second server.
How do I do to configure it this way?
2013 Mar 07
7
Extension cant pickup calls but can transfer.
Greetings.
I got an extension on my Elastix who cannot pick calls on the other
extensions, but It can transfer his calls to the other extensions. When
this extension tries to pickup a call pressing *8 it simply does not pick
it up. Transfering calls works just fine so dtmf may be not the problem.
Where should I look?
Any further information needed just ask.
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Att.*
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2013 Feb 01
1
RJ11 x RJ45
Sauda??es.
Como que se faz um conector RJ45 em uma ponta e RJ11 e outra. Pretendo
conectar a linha de um ATA em uma placa Khomp KFXO IP. A ponta que tem o
conector RJ45 est? crimpada com a sequencia 568B e vai ser conectada na
placa Khomp, mas a ponta RJ11 eu n?o sei como deve ficar.
Li alguns manuais na internet mas n?o entendi ao certo como tem que ser
feito.
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Luis H. Forchesatto
2013 Jun 10
2
Samba + LDAP: Issue adding machine.
Greetings.
I've run into a trouble when trying to add a new Win7 machine on a domain.
The domain is controlled by a server running Samba + LDAP (samba compiled
with ldap support), on a Debian 5 OS at the local network.
I've added the machine name to the LDAP three through phpldapadmin using
the option "Samba3 Machine" on the related submenu and via terminal on
samba. Then I
2006 Apr 24
2
Question about Asterisk realtime
Hi All:
I used FreePBX to configure Asterisk, and tables are create in MySQL by
using FreePBX install script.
I created two x-lite softphone accounts by using FreePBX, they are
stored in table sip as friend.
I followed wiki doc to edit the extconfig.conf file.
I can not get those two softphone to talk since I got the error message
from Xlite as:
Call failed: 503 service Unavailable
I noticed
2013 Sep 05
1
Windows 7 samba 4 domain join problem
I stood up a samba 4 (4.0.9) Active Directory domain controller on a Red
Hat Enterprise Linux 6.3 server, configured in accordance with the Samba
AD DC HOWTO <https://wiki.samba.org/index.php/Samba_AD_DC_HOWTO> , and
tailored to the domain name I want. I'm trying to join a Windows 7
Enterprise Edition client to the domain. Windows responds with "Your
computer could not be joined
2006 Feb 13
1
Bug in AMP 1.10.010 in sip outbound callerid
If you define a sip peer, wheather or not you put an entry in the field
OUTBOUND CID, if you
dial an external extension (let's say an extension on another asterisk
server, connected via IAX2 connection) the callerid
received by the foreign asterisk is device <YOURNUMBER>: i.e device <567>
If you take a look at etc/asterisk/sip_additional.conf, you can see under
the SIP extension
2005 Mar 11
7
Sip show registry returning nothing
Hello all,
For some reason I am not showing registration in SIP.
Can anyone give me an idea what can cause this?
asterisk1*CLI> sip show registry
Host Username Refresh State
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2006 Mar 29
0
R: RE : Echo cancellation
Hi Francois,
I kwnow, but I have "DSP:on" also if i not have an hardware echocan module :/ and I always have "Echo Cancellation: 0 taps, currently OFF".
This is my zapata.conf
[channels]
language = it
usecallerid = yes
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
cancallforward = yes
callreturn = yes
switchtype = euroisdn
2013 Jun 18
0
Identify port on Khomp card.
Greetings.
I've plugged 3 analog lines on an ethernet cable in an Khomp card to
receive it's incoming calls. Without any configuration, when I call those
numbers the asterisk server automatically answer the call and play the
default music.
The problem is: I need to discern the lines and redirect each one to his
respective extension. Since they doesn't got any Caller ID Service the
2005 May 15
2
SIP Gerenal settings conufsion
I have a little confusion about the general settings (other than the
register values) in the SIP
General area. I understand that for examle in a SIP context like [FWD]
or [BROADVOICE]
the entries in those areas are ths settings that take effect in any
communication woth FWD and/or BROADVOICE. However, I'm confused as to
the purpose of the
"general" settings -- to what or which
2009 Dec 24
0
X100P clone card problem
Hi,
I have problem with X100P clone card.I can not force it to work
under Asterisk 1.4.27.1 and DAHDI Version: 2.2.0.2.
I looked over and over on configuration and could not see any mistakes.
Here are relevant configuration files.
/etc/dahdi/system.conf
> fxsks=1
> echocanceller=mg2,1
> loadzone = hu
> defaultzone = hu
>
dahdi_tool
> OK Wildcard X100P
2007 Nov 30
2
Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX
Hi there!
I am having problems registering my 7970 hardphone with Asterisk 1.4(with FreePBX interface). I had an earlier post about trying to get it to work first with a 7970 emulator (Cisco IP Communicator) on the Asterisk Forum : http://forums.digium.com/viewtopic.php?t=19160
Instead I decided to try the real phone instead, and was able to advance further. The firmware was able to install
2005 Jun 06
1
Issue with SIP inter-op
Hi All,
I'm trying to connect to a SIP carrier who never connected with Asterisk.
I managed to connect with a sipura phone or a grandstream, no problem.
When I configure asterisk, I'm able to send out calls to the carrier no
problems,
however, receiving calls doesn't work, and I keep getting the following
messages:
<-- SIP read from 69.xx.xx.xx:5060:
INVITE
2005 Aug 16
1
problems with eyebeam - video phone
I am trying to connect two Xten eyeBeam Video Phone
No problems in voice connecting.
I tryed to modify my sip.conf
[general]
language=it
videosupport=yes
; enable Asterisk video support
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=h263
allow=gsm
allow=ulaw
allow=alaw
; H.263 is our video codec
;
2007 Oct 20
1
asterisk.conf and it's impact on CLI
I was previous using Asterisk 1.2.9.1 and decided to get some real servers
outside of my house. It was time for Asterisk 1.4.4.
I figured since all the conf files were in /etc/asterisk form the old box,
i'd just copy tha directory over to the new server. My SIP DID AGI stuff
worked, except running 'asterisk -r' doesn't. It tells me
' Unable to connect to remote asterisk (does
2005 Mar 08
1
All Circuits are Busy Now
I have downloaded and installed Asterisk@home and I have installed X-Lite on my Windows machine and I am able to connect it to the Asterisk server. I went ahead an created an account on Broadvoice today and followed the directions on http://voip-info.org/wiki-Asterisk+settings+Broadvoice and http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but when ever I try and make a call from
2007 Jul 03
1
Configuring BLF or Asterisk presence/Hints feature
Hi all,
I am working on
asterisk 1.2.18
zaptel 1.2.17
Polycom 650
polycom 430
SIP version 2.0.3.0131 for IP 650
SIP version for IP430 2.0.3.0127
freepbx 2.2.1
I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me.
Regards
FArooq
**********************************************
1
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in my
2011 Jun 10
1
Incoming Call Recording
Longtime lurker, first time poster. :)
A client of mine is in need of having Asterisk record every call that comes
in from a specific incoming route. I've added the following lines to the
sip_additional.conf file, but no recordings are showing up in the
/var/spool/asterisk/monitor/ folder.
record_out=always
record_in=always
Another page I came across on Google (
2005 Mar 10
1
Asterisk@Home, AMP, and Broadvoice
Egad, not again with Broadvoice! Anyhow, I recently installed AAH and
configured my TDM11B and got that and some SIP phones working. I still
have some issues to work out, etc, but my current problem is Broadvoice.
I have checked out all of the online resources, including the recent
list exchange about the recent changes made by Broadvoice. However, the
one thing I have found to be consitent in