Hi Ppl.
Once, couple weeks ago when I have updated * from CVS-HEAD something
happen and I could not send a fax anymore.
After that I have tried previous * CVS versions with different versions
of spandsp (0.0.1, 0.0.2pre4, 0.0.2pre10) but without any changes.
I have tried that on Fedora Core 2 with
Thx for your reply.
On Wed, 2005-03-16 at 17:35, Steve Underwood wrote:
Hi Vladyslav,
Use 0.0.2pre1, but add the line
fax.verbose = TRUE;
just after
fax_init(fax, calling_party, NULL);
That will turn on the detailed logging.
Added, recompiled and tested again
On Wed, 2005-03-16 at 18:25, Steve Underwood wrote:
Hi Vladyslav,
The log looks good so far. The far end has negotiated. The fast modem
has been tested. Transmission of the first page has been. What happens
next. I don't think the log really stopped at that point. Did you wait
long
On Wed, 2005-03-16 at 18:31, Kevin P. Fleming wrote:
[EMAIL PROTECTED] wrote:
If we want a box that can perform 60 calls. What would be apoproximate
budget
for that using AMD x86-64 ?
60 calls can easily be done on a 3.4GHz Pentium 4 box, no special
hardware is required.
Is that
On Sun, 2005-03-20 at 23:40, Joseph wrote:
How to compile additional module to asterisk?
I have app_nv_backgrounddetect.c file and followed instructions below,
but make did not generate app_nv_backgrounddetect.so or
app_nv_backgrounddetect.o
(1) Drop the code in your
Hello Ppl.
Please share info how have you set Agent priority in one queue.
Or there is no such kind of thing in current version ?
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There is option automon = *1 in features.conf
As I understand when *1 pressed during conversation = recording should
begin. But unfortunately it doesn't work for me.
I use CVS-HEAD-01/27/05
Does anyone has that feature working?
Thanks
--
Best Regards
VladK
Good day list,
I have a problem with ATA Handy Tone 286. It has been unsuccessfully
downgraded via HTTP. Seems like during downgrade there was a problem
with connection, because now it's not responding at all. There is no way
to get to it's voice menu via phone (by pressing button on it). The
Good day,
does anyone have pulse dialing working ?
http://voip-info.org/tiki-index.php?page=Asterisk+zaptel+pulse+dialing
At the link above there is a statement:
configuration for European telephone lines will look like:
make_time=63
break_time=37
pause_time=800
So where these pamameters
Good day!
Have a weird problem with HT-286 and Conference room. I use Asterisk
CVS-HEAD-06/04/04.
Here it is:
When HT-286 get into the conference room first and nobody in that room
everything seems ok (with any codec HT286 allowed), but when HT-286 get
into conference room when somebody already
On Fri, 2004-07-02 at 09:22, Dave Cotton wrote:
2. When HT use G729 codec = it gets busy signal and I could see such
output on asterisk console (
Jul 1 07:26:14 WARNING[737298]: chan_sip.c:1611 sip_write: Asked to
transmit frame type 4, while native formats is 256 (read/write = 4/256)
Hi All!
Have a problem with remote call pickup via sip.
When 1 sip phone is ringing and I'm trying to pickup a call from another
sip phone by dialing *8#
I'm getting:
-- Sent into invalid extension '*8#' in context 'from-sip-post' on
SIP/ciscok-8d39
such configs:
zapata.conf
--
ok. Thank U for a hint.
I have find out, the problem was with my ATA-186.
That box just use '#' not as sending key.
Does anyone know how to force ATA-186 to use '#'
as sending key.
Have tried *8 from softphone and that works fine.
On Mon, 2004-07-05 at 21:19, Brancaleoni Matteo wrote:
Hi
Il
The documentation is scarce, so could someone please share configs for
that.
On Fri, 2004-11-26 at 17:47, Brian West wrote:
I wonder why people are working on the MySQL specific version if ODBC
support is in and being developed.
Because people have this misconception that ODBC is slow.
In the call file U could Setvar (4 a channel) and after that use it in
dialplan.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out
On Fri, 2004-11-26 at 18:08, Steven Critchfield wrote:
On Fri, 2004-11-26 at 16:56 +0100, Ning Zhou wrote:
For example, I have a file under
so, Guys please anyone provide with an example (config files)
how to config sip peers and voicemail in the mysql DB.
On Fri, 2004-11-26 at 18:55, Brian West wrote:
Because people have this misconception that ODBC is slow.
It is not a misconception. It just depends on the application. For
Patch could not be applied to the latest cvs version
and also
http://www.voip-info.org/wiki-Asterisk+Broadvoice+patch?page=Asterisk%20Broadvoice%20patchcomments_threshold=0comments_offset=0comments_sort_mode=commentDate_desccomments_maxComments=10comments_parentId=1209#threadId1210
--
Best
Hi, ALL.
Have a problem with tiff image when receive fax in fine mode via Zap
(FXO card). The same via SIP is fine.
Could receive faxes in standard resolution without a problem, but fine
or super fine mode got tiff images corrupted.
With fine resolution, simply have twice the lines and it looks
Good day.
I have such problem with rxfax:
When I send a fax in fine mode resolution i receive only 50-60%
(sometimes 25%) of the page and the rest is compressed lines.
http://robik.azhelp.net/1084284449.0.tif
http://robik.azhelp.net/1084289786.1.tif
Please advise
On Mon, 2004-05-24 at 03:03,
Good day All.
Is there a way to pass DTMF signals to AGI script during conversation ?
Actually here what I want to make:
Users are usually dial using dialplan and when someone press *4 (during
conversation) I want to have agi script to deal with that, but those
users should keep talking and even
HI ALL.
Please help.
Problem: video calls drop after 15-20 seconds all the time.
Use * latest cvs.
from sip.conf
[1102]
type=friend
username=1102
host=dynamic
callerid=Veo webcam1102
canreinvite=no
disallow=all
allow=gsm
;allow=ulaw
allow=h261
allow=h263
from iax.conf
[peer2] ; 192.168.0.7
Windows messanger
On Thu, 2004-09-23 at 23:47, Florin Andrei wrote:
On Thu, 2004-09-23 at 01:59, Vladyslav wrote:
HI ALL.
Please help.
Problem: video calls drop after 15-20 seconds all the time.
Use * latest cvs.
What are the clients that you're using?
--
Best regards
Vlad
Have your even had success sending couple pages at once without loosing
a part of the page?
Had the same problem with X100P and it's still unsolved.
Just wondering how I could synchronize timing with PSTN on the FXO card.
On Thu, 2004-10-07 at 19:41, Snezhana Bekova wrote:
Hi!
I have
Hi All.
How to receive multiple pages with rxfax ?
Here is what I have:
exten = 10,1,Setvar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = 10,2,Setvar([EMAIL PROTECTED])
exten = 10,3,rxfax(${FAXFILE})
exten = 10,4,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM}
Hi.
Thank you all for your replies.
Now I do converting into pdf file and it's ok with multiple pages.
tiff2ps -a ${FAXFILE} | ps2pdf - ${FAXFILE//tiff/pdf}
On Wed, 2004-10-13 at 15:39, Steve Underwood wrote:
Vladyslav wrote:
Hi All.
How to receive multiple pages with rxfax ?
Here
Hi ALL!
I have such schema:
Asterisk #1 RxFAX -- Asterisk #2 TxFAX.
Both * - almost latest CVS version with spandsp-0.0.2.pre4
on the second * I put call file like this:
Channel: hidden with extension specified (in order with second
*)
Application: txfax
Data: /tmp/testfax.tif
so
HI!
I have a problem with Sjphone on ipaq.
It freeze when I receive a call on second line (seems like CPU is not
enough). It there a way to restrict call accepting when I'm already on
the phone via SIP in *?
because:
http://www.voip-info.org/wiki-PBX+Call+Waiting
For most POTS providers in the
Sorry, have already found that on wiki.
http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup
On Thu, 2004-10-28 at 13:31, Vladyslav wrote:
HI!
I have a problem with Sjphone on ipaq.
It freeze when I receive a call on second line (seems like CPU is not
enough). It there a way to restrict call
Try to mix them and you will get 1 file ...
On Thu, 2004-11-11 at 16:40, Matthew Boehm wrote:
What is the easiest way to record all parties of a meetme conference into 1
sound file?
I tried using Monitor just before the MeetMe call and it gave me files for
each person.
THanks,
Matthew
Hi.
On Asterisk for Xlite extension U need to set dtmf=inband
execute: sip reload
and that should be working
On Wed, 2006-02-01 at 17:02, Aisling wrote:
Hi,
Iâm wondering if anyone has experienced an issue with the XLite
softphone and asterisk accepting dtmf? I can listen to my voicemail
Just my couple notes on spa3000 and PSTN DTMFs.
Such schema:
PSTN - SPA3000 - Asterisk
Have problems with DTMF detection on incoming calls
when call comes from cell phone. Once per 4 times it
misdetect some ditigs (whether first digit will be
doubled or unrecognized at all).
Were tried
On Thu, 2006-03-02 at 10:59, [EMAIL PROTECTED] wrote:
On Thu, 2 Mar 2006, Vladyslav wrote:
Just my couple notes on spa3000 and PSTN DTMFs.
Such schema:
PSTN - SPA3000 - Asterisk
Have problems with DTMF detection on incoming calls
when call comes from cell phone. Once per 4 times
I believe you need to modify a little bit member.c file
in CVS version they use cid, but in stable version callerid.
Just replace properly cid with callerid.
It should help with that problem.
For example:
chan-cid.cid_num change to chan-callerid
On Mon, 2005-04-18 at 10:04, E rikje wrote:
I do.
On Tue, 2005-04-19 at 02:05, Wang Xiangzhou wrote:
Hi,
Does anyone run asterisk on MIPS architecture successfully?
Thanks,
Fox
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U better add line to your /etc/asterisk/modules.conf
noload = chan_phone.so
before [global] section
(of course in case U don't need that module...)
and it will start
On Tue, 2005-04-19 at 02:37, Paul A Brown wrote:
I have just installed 1.0.7 from the debian source using apt-get.
Now it
On Sun, 2005-04-17 at 17:14, Walt Reed wrote:
On Fri, Apr 15, 2005 at 10:39:44PM +0800, Ronald Wiplinger said:
Eric Wieling wrote:
I have put into each phone settings (sip.conf and zapata.conf) in my
office:
callgroup=1
pickupgroup=1
I cannot pickup any calls from another
In your incoming context add
exten = fax,1,Goto(fax,2202,1)
On Wed, 2005-04-20 at 12:26, Ronald Wiplinger wrote:
-- Starting simple switch on 'Zap/3-1'
-- Executing NoOp(Zap/3-1, 9229443944-) in new stack
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing
U don't need to have sound device for * sound service running
just make sure that you have in modules.conf
noload = chan_alsa.so
noload = chan_oss.so
On Wed, 2005-04-20 at 08:44, [EMAIL PROTECTED] wrote:
Hi,
I installed asterisk-1-0-7 and running it succesfully. But iam unable to use
the
You need to add that to context where you have BackGround application
running.
house-day and house-night I believe.
On Wed, 2005-04-20 at 13:16, Ronald Wiplinger wrote:
Vladyslav wrote:
In your incoming context add
exten = fax,1,Goto(fax,2202,1)
It did not work
Have U tried to use DUNDI for that purpose ?
It's the best solution U could find.
http://www.voip-info.org/wiki-Asterisk+DUNDi+Call+Routing
On Mon, 2005-05-09 at 20:48, Vikram Rangnekar wrote:
+++ Kanuri, Seshu (Company IT) [09/05/05 11:25 -0400]:
Vikram,
Instead of trying to be
Better take a look at Dial cmd. and on it's possibility to run Macros.
On Thu, 2005-05-19 at 00:19, Ben Johnson wrote:
I read an article in the wiki on a
(http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) that allows Asterisk
to forward a call to a cell phone if someone does not answer
Hi ALL.
I have a problem with TxFax application. (RxFax is working properly)
Txfax does not work when sending tiff files bigger than approx. 18Kb. If tiff
file smaller than 18Kb everything is OK. Tested in LAN with Panasonic UF-E1 fax
machine (tiff files was created by the rxfax from that
Hi All.
I have notice a problem with FM feature (screen macros) on Asterisk CVS
version.
When call goes via IAX and calling part accept the call on Dial
command with option M, in macros context it's setting
MACRO_RESULT=CONTINUE, but anyway it hangups both channels.
If anyone faced with such
What's wrong with
ftp://ftp.opencall.org/pub/
It says Can't open data connection
On Thu, 2004-07-29 at 17:44, Steve Underwood wrote:
[EMAIL PROTECTED] wrote:
What are your experiences with faxing through Asterisk to the PSTN?
We are using g.711u as a codec, and are
BTW, compilation of rxfax with latest CVS-2004-07-29 fails.
and Makefile.patch (which is on the site) should be modified as well.
gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c
In file included from app_rxfax.c:14:
../include/asterisk/lock.h: In function `ast_mutex_init':
HI All.
I'm using tiff-v.3.5.7 and spandsp-0.0.1k with latest * cvs on
Slackware-10.0.
Here is debug messages from * console.
Please advise.
Can receive fax
Selected data signalling rate: V.29, 9600bps
2D coding OK
Scan line length: 215mm
Recording length: A4 (297mm)
Minimum scan line time: 0ms
Try to comment out in your sip.conf
;qualify=yes
On Mon, 2004-08-09 at 06:52, Sathya Weerasooriya wrote:
Just wondering whether we have a resolution to iconnect incoming
problem, which started few days ago.
Cheers
SW
--
Best regards
Vlad
___
Good day ALL.
Could anyone tell me is there a way to get fax debug output into the
file when running safe_asterisk ?
--
V.8 capable
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter, V.29 and V.17
R8x7.7lines/mm
How to force * to write txfax console log into file ?
On Thu, 2004-08-26 at 13:24, Steve Underwood wrote:
Hi,
If that is what you are running, you should be getting audio log files.
The have names like /tmp/fax-rx-audio-date and
/tmp/fax-tx-audio-date. I need a matching pair and the
On Fri, 2004-09-03 at 10:56, Altus Snyman wrote:
Good day all
I'm interested in video on asterisk using SIP and windows clients
Now I did my research on http://www.voip-info.org/wiki-Asterisk+video
I have a few question:
*On the page they say you need the H.261 H.263? codecs,are these
Today morning cvs server checkout problem:
cvs server: Updating asterisk-addons/format_mp3
cvs server: failed to create lock directory for
`/usr/cvsroot/asterisk-addons/format_mp3'
(/usr/cvsroot/asterisk-addons/format_mp3/#cvs.lock): Permission denied
cvs server: failed to obtain dir lock in
HI!
Have a weird problem with X-lite Meetme.
When X-Lite user are join to conference room NOT first one, than
X-Lite user do not hear anything. This problem gone when X-Lite user get
into conference room first (when nobody there).
sip.conf
[104]
context=VoIP-only
type=friend
username=104
And the same problem with Grandstream HandyTone-286 as well
On Wed, 2004-09-08 at 11:43, Vladyslav wrote:
HI!
Have a weird problem with X-lite Meetme.
When X-Lite user are join to conference room NOT first one, than
X-Lite user do not hear anything. This problem gone when X-Lite user get
On Thu, 2004-09-09 at 09:47, Umar Sear wrote:
On Wed, 2004-09-08 at 09:43, Vladyslav wrote:
HI!
Have a weird problem with X-lite Meetme.
When X-Lite user are join to conference room NOT first one, than
X-Lite user do not hear anything. This problem gone when X-Lite user get
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