Dear all,
as you know, we can use Originate Command to auto-dial a out-bound
call to a extention or app since 1.6.2.
but when i Originate a call, and hangup. the cdr of this call has no
CDR(clid) and CDR(src).
Could you tell me how to set the Callerid to cdr from an Originate
call? I use
Hello list,
how can I connect 2 channels together ?! As this is what is suggested by
Philipp...
Kind regards,
Jonas.
On 09/07/2010 09:04 PM, Jonas Kellens wrote:
On 09/07/2010 06:40 PM, Philipp von Klitzing wrote:
is their a way to keep having a dialtone for the calling party when
Hi gang,
I'm in the process of documenting an 1100 line IVR dialplan. In
this dialplan I have almost 300 Set commands. The numeric ones are simple;
Exten = s,1,Set(TYPE3=0)
My question is about the Alpha/Text values;
If I do
Exten = s,1,Set(beep=beep)
Or
Exten =
I'm testing some of the new features of Asterisk 1.8,
but seems an impossible mission to make the calendars run.
cli show empty:
**CLI calendar show
calendars
Calendar Type
Status
-- *
And i can't see anything into the log, calendar.conf is
On Tue, Sep 7, 2010 at 2:56 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
I would like to use Asterisk for a call center, but really does not know if
Asterisk support the following in a good way:
1) Ability to do an inteligent routing, so to route the call to the proper
skill group
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrià Vidal
Subject: [asterisk-users] asterisk 1.8 Calendar
I'm testing some of the new features of Asterisk 1.8,
but seems an impossible mission to make the calendars run.
cli show
On Wed, Sep 8, 2010 at 3:56 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Adrià Vidal
*Subject:* [asterisk-users] asterisk 1.8 Calendar
I'm testing some of the new features of
Hello,
in asterisk 1.4.30 all realtime configurations go well.
In asterisk 1.6.2.11 the following appears on CLI :
[Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)
[Sep 8 16:43:43]
Jonas Kellens wrote:
Hello,
in asterisk 1.4.30 all realtime configurations go well.
In asterisk 1.6.2.11 the following appears on CLI :
[Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)
On 09/08/2010 04:50 PM, Gareth Blades wrote:
Jonas Kellens wrote:
Hello,
in asterisk 1.4.30 all realtime configurations go well.
In asterisk 1.6.2.11 the following appears on CLI :
[Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
MySQL RealTime: Invalid
Hi,
I am using 1.6.2.11, and I need to be able to include the name of the
channel that answered a call in the call-recording filename.
At a guess we need to use the Queue(name,,macro) or
Dial(chan1chan2,,M(macro)) and use the macro to update the call
recording filename. But, the macro runs
Jonas Kellens wrote:
On 09/08/2010 04:50 PM, Gareth Blades wrote:
Jonas Kellens wrote:
Hello,
in asterisk 1.4.30 all realtime configurations go well.
In asterisk 1.6.2.11 the following appears on CLI :
[Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
MySQL
On 09/08/2010 05:18 PM, Gareth Blades wrote:
Jonas Kellens wrote:
On 09/08/2010 04:50 PM, Gareth Blades wrote:
Jonas Kellens wrote:
Hello,
in asterisk 1.4.30 all realtime configurations go well.
In asterisk 1.6.2.11 the following appears on CLI :
[Sep 8 16:43:43]
Hello
I purchased an AEX800 card to replace the ageing cheap channel bank/T1
card solution a few months ago, assuming that it would be a more robust
solution for my small scale phone system. However, it appears to be
anything but that.
Originally implemented as a XEN dom-u virtual machine on a
On 09/08/2010 10:38 AM, Christian Weeks wrote:
So I am asking the list, do you have any advice except perhaps to go
back to the broken channel bank? Is it really true that my modern server
class machine (quad core xeon) cannot handle the AEX800, whereas my
seven year old AMD desktop (previous
Hi,
I am trying to configure ipsec on asterisk. Have configured
/etc/racoon/racoon.conf and /etc/raccoon/psk.txt. Also have policy file in
same folder.
Have run racoon. Still I can't receive calls.
Can anyone please tell if any extra step is needed.
Thanks
--
Situatation is that operator mangosip.ru got on 1 ip many realms. Problem is
that asterisk automatically changes host to ip in To: field. So operator
send error back. How to force asterisk not to change host to ip?
Settings:
register = A**1:p...@mangosip.ru 1%3ap...@mangosip.ru - registry OK. But
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta5.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any
I'm trying to install both a Sangnoma A102 (with echo cancellation) card and
a Digium 8 port analog card with echo cancellation (Digium AEX800E) in the
same server. I know I probably shouldn't have mixed vendors - lesson
learned for next time.
That said, I have everything working fine...except
On Wed, Sep 8, 2010 at 12:10 PM, Deepika Nijhawan
deepika.nijha...@oxygen8.com wrote:
Can anyone please tell if any extra step is needed.
Not without you providing some debugging information of your call.
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
--
Hello list,
I've upgraded to 1.6.2.11 to be able to use the bridge()-command, but
there is no info on it :
asterisk*CLI core show application Bridge
-= Info about application 'Bridge' =-
[Synopsis]
Not available
[Description]
Not available
[Syntax]
Not available
[Arguments]
Not
On Fri, Sep 3, 2010 at 6:11 AM, mattias m...@mjw.se wrote:
Can i run asterisk on a openvz vps or do i need a kernel?
I dont use dadi
--
Works just fine for our voicemail server (~450 users).
CP.
--
_
-- Bandwidth and
Hi
Could someone point me to a provider of DID's in Thailand.
Thanks for any response.
--
- Eric Smith
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Smith
Sent: Wednesday, September 08, 2010 2:43 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Thailand DID
Hi
Could someone point me to a
Hello!
I'm trying to make fax work on Asterisk 1.6. I've installed DAHDI,
marked spandsp as app_fax, but faxes are not going trough,
although application itself installs successfully. I've been using rx_fax
tx_fax on 1.4 and everything worked fine.
Can you recommend any specific solution to this
On Wed, 2010-09-08 at 11:06 -0500, Shaun Ruffell wrote:
On 09/08/2010 10:38 AM, Christian Weeks wrote:
So I am asking the list, do you have any advice except perhaps to go
back to the broken channel bank? Is it really true that my modern server
class machine (quad core xeon) cannot handle
On Wed, Sep 8, 2010 at 4:18 PM, Stanislav Korsei kor...@rinogo.com wrote:
Can you recommend any specific solution to this problem or way to install
app_fax?
Not without specific debugging about what problems you're seeing. You
get a lot of information when faxes succeed or fail. Try a fax and
Hi Everyone,
Can you tell me how many concurrent TDM (Dahdi) calls
that a single asterisk box can handle.
Configuration is as follow :
Quad core Xeon 3 GHZ, 4Gb RAM, asterisk 1.6.2.9
Also do you know a good tool to stress out asterisk?
Kind regards
--
*Adolphe CHER-AIME
First off Digium technical support should be able to help you trouble shoot.
On 09/08/2010 03:27 PM, Christian Weeks wrote:
On Wed, 2010-09-08 at 11:06 -0500, Shaun Ruffell wrote:
On 09/08/2010 10:38 AM, Christian Weeks wrote:
So I am asking the list, do you have any advice except perhaps to
In a moment of inspiration, I recompiled both DAHDI and Wanpipe - and this
seemed to have resolved my issues, all is working great now.
On Wed, Sep 8, 2010 at 10:52 AM, Joel Maslak jmas...@antelope.net wrote:
I'm trying to install both a Sangnoma A102 (with echo cancellation) card
and a Digium
On Wed, Sep 8, 2010 at 5:57 PM, Adolphe Cher-Aime achera...@gmail.com wrote:
Quad core Xeon 3 GHZ, 4Gb RAM, asterisk 1.6.2.9
How many PCI / PCI-X slots do you have? Do you plan to use hardware
Echo Cancellation? What about transcoding?
Theses will be your basic concerns when dealing with
Thank you for your quick reply. I'll use Redfone foneBridge to terminate
E1s.
I won't use any transcoding. For now I opt for software based echo
cancellation.
Thanks.
On Wed, Sep 8, 2010 at 6:43 PM, Paul Belanger
paul.belan...@polybeacon.comwrote:
On Wed, Sep 8, 2010 at 5:57 PM, Adolphe
On Wed, 8 Sep 2010, Adolphe Cher-Aime wrote:
Can you tell me how many concurrent TDM (Dahdi) calls that a single
asterisk box can handle. Configuration is as follow :
Quad core Xeon 3 GHZ, 4Gb RAM, asterisk 1.6.2.9
Every situation is different, but...
This box should be able to handle far
Hi All,
I am running asterisk ver 1.2.4 and have faced this error:
chan_sip.c: Failed to grab lock, trying again..
astrisk freezes and doesn't accept calls.
after some time everything is normalized.
I am not using realtime. However I am using cdr_mysql
I have gone through bug id 5942 6181
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