Hello,
Can i identify the sound files that are played in the asterisk
console? I defined the verbose to 100 but i can not see the sound
files that are played in some situations... :(
For example, I need to know what files are played for the message:
Extension xxx is unavailable
The goal is
Hello,
I need to connect one diva server 4bri to a portuguese BRI interface.
The operator (PT) said that this bri is in point-to-multipoint mode
(S0). Previously one PBX has connected to that interface.
The asterisk and diva drivers are working ok but i cannot communicate
to outside via this
problem?
Thanks and best regards,
PS.
2006/10/29, Alberto Pastore [EMAIL PROTECTED]:
Pedro Silva ha scritto:
Hello,
I need to connect one diva server 4bri to a portuguese BRI interface.
The operator (PT) said that this bri is in point-to-multipoint mode
(S0). Previously one PBX has connected
that the problem is between diva card and BRI access.
So i need to solve first this problem and only after that im care with
asterisk... :)
Obrigado desde já pela disponibilidade de ajuda!
PS.
May be i can help.
Sou Português:)
On 10/29/06, Pedro Silva [EMAIL PROTECTED] wrote:
Thanks Alberto!
I
Hello again Alberto!
Anyway, to get more info, try to open a second shell
and run /usr/lib/eicon/divas/xlog
then on the first shell redo the telsampl test, then
post the output of xlog off the list to my address
(alberto at msoft-italia.com)
This is the xlog output:
4:1736:074 - CREATEID ok:
Finally this works!!! :)
Tanks to Alberto and Marco by your help!
The problems are:
- the cable was connected to the wong card port... :(
- the card config needs to be: ETSI; TE; Point-to-Point (I thought
that was point-to-multipoint).
Best regards,
PS.
2006/10/29, Pedro Silva [EMAIL PROTECTED
... but is
problem in asterisk or is before asterisk, on diva card...?
Tanks by any possible help!
Best regards,
PS.
2006/10/29, Pedro Silva [EMAIL PROTECTED]:
Finally this works!!! :)
Tanks to Alberto and Marco by your help!
The problems are:
- the cable was connected to the wong card port
Hello,
The problem was wrong contexts defined like Marco said, and is solved.
Now, i have another problem...of course :)
On incoming calls, i only can receive calls if i define a line like
the following, in extensions.conf:
exten = _.,n,Dial(SIP/500,30,tr) (all incoming calls are redirected
to
2006/11/1, Armin Schindler [EMAIL PROTECTED]:
On Wed, 1 Nov 2006, Pedro Silva wrote:
As you can see in the log below, the called number is just '0':
CalledPartyNumber = 810
It seems DDI 0 of your line was called. So just do
exten = 0,n,Dial...
Armin
Is that right! Thanks
Hello all,
To test some configs i forgot the trixbox web config (freepbx) and i
made changes directly in asterisk config files (sip.conf,
extensions.conf, etc). Result: asterisk is working ok but the the web
config is totaly confused and, if i made a change via freepbx this not
works ok. Only
Hello,
Anyone knows if chan_capi-0.7.1 includes the patch to support capiAnswerFax?
I tried to apply this patch (from http://www.mlkj.net/asterisk/) but
it give me errors...
Also i tried define one extension for fax receptions but this dont works:
exten = 1,1,Goto(handle_fax,s,1)
exten =
Excellent, Michiel! This works :)
You know what kind of file it is created (SFF)?
Can you send to me the example faxreceive.php?
Thanks and best regards!
PS.
2006/11/7, Michiel van Baak [EMAIL PROTECTED]:
On 15:03, Tue 07 Nov 06, Pedro Silva wrote:
Hello,
Anyone knows if chan_capi-0.7.1
Hello,
When i try to install the sfftobmp3.1, the tribbox box give me the
following error:
...
checking for TIFFOpen in -ltiff... yes
checking jpeglib.h usability... no
checking jpeglib.h presence... no
checking for jpeglib.h... no
configure: error: jpeglib.h not found
I try to find packages
Hello,
From some days ago, when i made changes in web interface to asterisk
that comes with trixbox (freepbx), this dont reflect the changes in
asterisk configuration.
I has reviewed the file permissions in /etc/asterisk and all files are
writable to asterisk user.
In freepbx all appears to be
extensions_addicional.conf and sip_addicional.con are supposed to be
updated and are not.
Best regards,
PS.
Alex
On 11/17/06, Pedro Silva [EMAIL PROTECTED] wrote:
Hello,
From some days ago, when i made changes in web interface to asterisk
that comes with trixbox (freepbx), this dont reflect the changes
Alex was right. The problem is that when i make changes in freepbx,
those changes are not written in the config files.
I only made modifications in files_custom.conf.
The version of freePbx that i use is 2.1.1 (not beta) and Asterisk 1.2.12.1.
Thanks by your help,
Ps.
2006/11/18, Alex Robar
. i dont know what causes this
error but i have noticed that restarting FreePBX or re-installing the
application stops this. Just restart the box
On 11/18/06, Pedro Silva [EMAIL PROTECTED] wrote:
Alex was right. The problem is that when i make changes in freepbx,
those changes are not written
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