Hello,
I have a machine with two cards
installed, one digium that gives e1 connectivity and one quadBri for the ISDN
line.
I canusethem
independently.I have one zaptel.conf and one zapata.conf for each card. I
would like to work with them at the same time and I am not sure about how could
Hello,
I'm trying to install CAPI Driver for Suse
9.2 and I found the documentation for this pretty old since It refers
toSuse 8.2 ( http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install).This
is especially apparent when I look at the section of these instructions for
Hi,
Regarding capi debug, I don't know how
to translate reasons like 0x3302 or infos like 0.I didn't find any
'translator' googleing capi debugging. Do you know about any 'translator' for
this or should I be as clever as to know what a reason 0x3302 is?
What is this debug for if I can't
I'm going to answer myself. I don't know If
somebody already did it because I'm using digest mode.
CAPI specification is available at http://www.capi.org/, It explains all the
commands and associated identifiers. Now I know that reason0x3302 in
DISCONNECT_IND means Protocol error, Layer 2.
Hi,
Trying to configure a voicemail system on
FreeBSD 4.10 + asterisk 0.9.0, I found the following
problems:
1. When I try to launch VoicemailMain from
IAX Softphone (IAXComm), asterisk generates a Segmentation fault (core dumped)
and obviosly all the system went down. It doesn't happen
Hi all,
I think there is no asterisk-addons version
for freebsd. Am I right? I tried to compile the standard version but I couldn't
do iton FreeBSD, may be the idea is as crazy as try to install asterisk
for linux on freebsd!
___
Asterisk-Users
Hello,
I'm just wondering what is the situation
today, 24 Nov 2004, regarding asterisk timer for freebsd. I would like
to know ifthere isany way to run Meetme on Freebsdorif
there is anybody currently working on it.Cheers,
Victor.
___
Hello,
I would like to know how to add an iax user
to the system, as simple as that, but it's hard to find one
example.
Just the entry to iax.conf and
extensions.conf would be enough. I have already conect two asterisk servers,
there are plenty of examples about how to do this but in this
Hi all,
Despite of www.openh323.org and some other sites claim the cvs has an empty
password for anonymous, I am unable to download the code from it. Any clue?
Logging in to :pserver:[EMAIL PROTECTED]:2401/cvsroot/openh323
CVS password:
cvs [login aborted]: reading from server:
Hi,
I am 'playing' with asttapi which looks great on a first installation but I
must be missing something regarding the source code because I haven't been
able to work with it without problems.
If you have played with this, you already know that the code to talk to
Asterisk is placed in a file
Hello,
My question is.. How does cdr_manager work? Does it suppose to populate
cdr-csv/Master.csv? What about the cdr table on the database? What is the
event some people talk about?
I have changed (and reloaded) my configuration of cdr_manager.conf to
;
; Asterisk Call Management CDR
;
Hi,
I'm trying to use the spandsp fax-back facility and despite I can send and
receive faxes, this is not working fine. I would like to get a debug of what
is going on. I am using the flag debug, but I don't know if txfax is
generating any log information or where it is saving it. I just don't
Hello everyone,
I'm trying to find out a way to whisper the time remaining for a
prepaid application on a established channel. Unfortunately I think
there is a lack of PlayBack/Background commands which can be applied on
a working channel as well as a lack of spy/whispering commands available
the call.
Victor Alvarez wrote:
Hello everyone,
I'm trying to find out a way to whisper the time remaining for a
prepaid application on a established channel. Unfortunately I think
there is a lack of PlayBack/Background commands which can be applied on
a working channel as well
PM, Victor Alvarez wrote:
Hello everyone,
I'm trying to find out a way to whisper the time remaining for a
prepaid application on a established channel. Unfortunately I think
there is a lack of PlayBack/Background commands which can be
applied on
a working channel as well as a lack
I really think this matter deserves
attention. I have been asked many timesabout it.
Regards,
Victor.
Hello, I can understand why asterisk is
designed to not to allow two UAs with the same usr/pwd, http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html,
but I have to
Hello,
I can understand why asterisk is designed to
not to allow two UAs with the same usr/pwd, http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html,
but I have to find a solution for this.
My first option is use SERas anextension end of Asterisk, to allow more than
Hi,
I'm trying to transfer an incomingcall
from the PSTNto another PSTN number through a SER - Asterisk system. SER
doing only routing..
pstn call- SER - asterisk (call
forward) - SER - pstn
Logic for SER: If something comes from the
pstn, send it to asterisk. If something comes from
Hello,
I havemore feedback regarding the
question I posted yesterday (Call forward SER as SIP
router).
pstn-SER-asterisk (call
forward)-SER-Pstn fails when the far end picks the phone up.
Errorshowed inasterisk is Got SIP response 481 "Invalid CSeq Number"
back from XXX.XX.XX.XX (SER).
Hello,
This is an issue posted last January in the
list: http://lists.digium.com/pipermail/asterisk-users/2005-January/080878.html
I have the same problem with 1.0.9. It doesn't
matter if you configure indications.conf with default country=uk, you get an US
ringback. Command answer before
Hi,
Monitor and soxmix (m option) work fine in
CVS Head, not in Asterisk 1.0.9, as the Wiki says.http://www.voip-info.org/tiki-index.php?page=Monitor+setup+sample
Anyway I am wondering why asterisk
1.0.9console shows on Hang up: "monitor executing ( nice -n 19 soxmix
Hello,
I can't use IAX with my last
CVS-NHEAD-04/28/05-16:00:04 installation.Every time I try to use an iax
channel or register an iax user, I get a Segmentation fault.
Trace:
-- Executing Dial("SIP/25-0368",
"IAX2/25|20|Tt") Segmentation fault
[EMAIL PROTECTED] root]# Ouch ... error
attempt. I don't
know whether to blame this version of asterisk or what. I used this machine with
an older version and IAX run without problems.
Just to mention that I'm using different
versions of IAX Softphones with the samesad result.
Regards,
Victor.
--
Victor
Hello,
Paul P.
Pongco already reported this issue on Friday, April 08, 2005.I've
used the same title for the Subject.
It seems that Asterisk crashes generating a
core dumped everytime there is an attempt of register an IAX softphone if
realtime is activated.
I don't have a clue about
Hello,
I'm willing to implement call barring for
incoming and outgoing calls and I would like to discuss it with the
listfirst, since I think It can't be implemented in a 'natural way' and I
will need to use agi scripting - database.
Process would be:
1. incoming calls
priority
1, call
I was thinking in use realtime asterisk to decide
wether to call the agi or not. I mean, add the agi in the first position of the
dialplan or delete it for each user. So I can activate - desactivate it and call
the script only when necessary.
Thanks,
Victor.
It's pretty obvious from the wiki that realtime and
Nat don't befriend quite well. As It is obvious the necesity of both of them,
mainly have clients under nat talking to an asterisk server. The question I
would like to throw away is.. What would you do to have both of them? I have two
Hi,
I'm afraid I don't know how to use
thecommand Transfer. I have a couple of SIP users in the system and
although exten = 35,1,Dial(SIP/33) works fine, exten = 35,1,Transfer(33)
just don't work. All the description in the wiki is 'Transfer(exten)' without a
single example.
Hi,
How could I change the
defaultpermissions for voicemails?
When I try to installthe patch
mentionedat http://www.voip-info.org/tiki-index.php?page=Asterisk+gui+vmail.cgi,
I get the following response:
patch
voicemail.patch
patching file app_voicemail.cHunk #1 FAILED at
39.Hunk #2
Grazie Giorgio.
Then I should reformulate the question.
Is there any new version of this patch (http://www.voip-info.org/img/wiki_up/voicemail.patch)?
The one at the wiki is dated November 2004 and doesn't seem to be suitable for
current version of voicemail.c.
Is there any other way
Hi,
I have been playing around with the latest release of asttapi and I have
found the 'hangup' problem already reported to the list here
http://lists.digium.com/pipermail/asterisk-users/2006-May/151260.html
Apparently hangup should be done by making use of UserEvent commands. So I
have
Hi,
I founda problem when trying to install
the module chan_bluetooth from 'the crazy greek'. Most of installation seems
fine, chan_bluetooth.so is created and located in /usr/src/asterisk/channels/.
But when I try to start up asterisk, I get the following message:
[chan_bluetooth.so]Jan 8
Thanks Colin and Mark for your answers. I finally
manage to start up asterisk bycopying libbluetooth to
/usr/lib/.
Now the final step comes, make this bluetooth
thing works.
These are my configuration
files:
/etc/bluetooth/rfcomm.conf:
rfcomm0
{ bind yes;
device
00:0E:6D:34:BD:B1;
Hi,
Thanks Dave, gracias Jose
Luis ;-).
Once everything is
configured, the mobile phone connected via bluetooth.. I've got a segmentation
fault when trying to dial from sip to bluetooth:
CLI Nov 11 16:53:34 NOTICE[]:
/usr/src/chan_bluetooth/chan_bluetooth.c:2041 try_connect:
Hi,
I have been tryingchan_bluetooth module
for asterisk during last week. I found some difficulties configuring it, due
mainlyto my ignorance and secondly to the lack of documentation. Thanks to
the listI have managed to configurethe Audio Gateway modeand I
have a strong doubt about how
I have been trying chan_bluetooth module for asterisk during last
week. I found some difficulties configuring it, due mainly to my
ignorance and secondly to the lack of documentation. Thanks to the
list I have managed to configure the Audio Gateway mode and I have a
Can you try this again with a CLI open on * with a high verbose level.
This is what I get when asterisk drops out of the chain.
chan_bluetooth.c:701 sco_thread: SCO connection error: Connection
refused (errno 111)
This is the trace asterisk is giving me:
Helloall!
I am trying to load sip.conf from mysql database. I have followed the
instructions at http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers. Seems that the authentication (user psw) works fine but I would
like to get more information from mysql and I don't know how to retrieve
Hi,
First of all thank you Matthew, Nicolas and
Ryan for your response.
I would like to get information like context,
mailbox, callgroup, pickupgroup, codecs... also nat! If I make the substitution
of the text file i wouldn't like to miss information in the
process.
,
the author of the current version according tochan_sip.c.
Cheers,
Victor Alvarez.
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