Turn off immediate=yes in zapata.conf
regards
Martin
On Thu, 4 Sep 2003, =?us-ascii?Q?Carlos Fern=E1ndez Puente?= wrote:
Hi everybody.
We have a problem with the configuration of the card, the cards work and we receive
incoming calls but asterisk don't receive dnid. We have 5 servers with
Someone had a patch to retrieve the oldest call from the parking queue...
maybe that could help
regards
Martin
On Thu, 4 Sep 2003, WipeOut . wrote:
Parking the call is a problem becasue you will not hear the parked call location
(because its a blind transfer into the parked call)..
The
Lets say that you have two phones: Zap/1 and Zap/2
and there comes a call over IAX to Zap/1
since channel 1 is in the callgroup 1
and channel 2 is in the pickupgroup 1 channel 2 can dial *8 and pick up
the call that comes to channel 1.
Martin
On Thu, 4 Sep 2003, Mickey Binder wrote:
I must be
I have to specify anything in
sip.conf or is it enough to specify it in zapata.conf?
-Original Message-
From: Martin Pycko [mailto:[EMAIL PROTECTED]
Sent: 4. september 2003 21:08
To: Asterisk maillist (E-mail)
Subject: Re: [Asterisk-Users] I don't think I understand Call pickup
: Martin Pycko [mailto:[EMAIL PROTECTED]
Sent: 4. september 2003 21:22
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] I don't think I understand Call pickup
Just have that zap channel in the pickupgroup = callgroup of the sip
phones
Martin
On Thu, 4 Sep 2003, Mickey Binder
Oh and with the recent CVS code call pickup is broken for sip phones ...
I just got that from bugtracker
Martin
On Thu, 4 Sep 2003, Martin Pycko wrote:
You have to do it reverse way ... pickupgroup = 1 for sip phone (since
you're picking it up on this one) and callgroup = 1 for zap channels
There were people that connected our hardware to Telstra in AU. However it
seems that only TE410P works well (there are some framer issues on
E[14]00P).
regards
Martin
On Wed, 3 Sep 2003, Phillip Britt wrote:
Hi,
Just wondering if there are any people out there using the Zaptel Wildcard
E1
But there were some ppl who did run EM on E1. It was on the list, search
archives.
regards
Martin
On Wed, 3 Sep 2003, Steve Underwood wrote:
Paulo Mannheimer wrote:
Hi,
I'm testing an E1 with EM signaling. Some of the problems I'm running
into are the following:
1) if I try to
If you're in US try to use callprogress=yes in zapata.conf. The problem
with analog channels is that we assume blindly that after the number has
been dialed the call is answered. Unless you use some progress detection
algorithm that works for you. So try callprogress=yes.
regards
Martin
On Tue,
This is a isdn4linux bug (it's in the kernel code) and you have to disable
it yourself or with some patches that were posted some time ago.
regards
Martin
On Wed, 3 Sep 2003, Jay Tyndall wrote:
Hi,
I am receiving calls via a Netjet-S card on asterisk, and I notice that
whenever I am
Zapata cards make 8000 irqs a second. There is no buffer so after each
I'd say 1000 IRQ per second per board (whether it's X100P or T100P or
T400P etc) We read 8 bytes per IRQ per interface
Martin
cycle, the computer must service the card. This is true for x100p and
s100U and the others
Not yet.
Asterisk always sends 20 ms of voice data per packet.
regards
Martin
On Wed, 3 Sep 2003, Paul Lambert wrote:
Noticed that I can adjust the number if frames/packet on the GrandStream
phone. Can * do the same?
___
Asterisk-Users mailing
Try the new_codec_binary/codec_g729b.so from our ftp site.
regards
Martin
On Wed, 3 Sep 2003, Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
After upgrading the kernel on an Asterisk box, asterisk segfaults on startup.
It seems like it's the g729 codec that
No, it doesn't need DSP to process audio. You can use your sound card as a
phone though.
regards
Martin
On Wed, 3 Sep 2003, Peter Pauly wrote:
adding nat=yes to the sip definition made no difference.
Does Asterisk use the DSP in your sound card to do the
audio processing?
or SS7. There is so little
standardisation. A place I used to work has a substantial team turning
out new signalling protocol state machines for each customer of its E1
muxes.
Regards,
Steve
Martin Pycko wrote:
Also try to see the debug on the console (edit /etc/asterisk/logger.conf
:
On Wed, 3 Sep 2003, Martin Pycko wrote:
If you're in US try to use callprogress=yes in zapata.conf. The problem
with analog channels is that we assume blindly that after the number has
been dialed the call is answered. Unless you use some progress detection
algorithm that works for you. So try
Install kernel-source-correct_kernel_version.rpm if you have RH.
regards
Martin
On Wed, 3 Sep 2003, Howard Tarlow wrote:
I got the following error when tying to make the zaptel files. Any help would be
greatly appreciated
Thanks,
HT
Error Modules shouild never use kernel-headers system
Well that can be done but you'd have to change some UDP socket functions
to TCP socket functions in channels/chan_sip.c
regards
Martin
On Thu, 4 Sep 2003, Master Abi wrote:
Hi
I read through the archives but could not find much reference to * using
SIP on TCP instead of UDP for signalling.
. They really do need state
machines. Write them in C, or do as we did (draw them graphically in
Autocad, and compile them directly down to ROM tables). Either way you
build a state machine.
Regards,
Steve
Martin Pycko wrote:
Maybe if they'd write the PRI stack in C instead of making a state
the limitation of most Internet
connections including a T1.
Martin Pycko wrote:
Not yet.
Asterisk always sends 20 ms of voice data per packet.
regards
Martin
On Wed, 3 Sep 2003, Paul Lambert wrote:
Noticed that I can adjust the number if frames/packet on the GrandStream
phone
Only in IAX2 trunking mode.
Martin
On Wed, 3 Sep 2003, Paul Lambert wrote:
When multiple calls are in session between two IAX servers do the voice
frames from the various calls get put into a single packet to conserve
on total packet rate?
___
Your configs look ok. All you need is BNC to RJ45 converter (I think the
standard is G.703)
regards
Martin
On Tue, 2 Sep 2003, Langley, Sean wrote:
Dear Telcotype Braniacs,
I have tried doing a google search to find out what this switch looks like, what the
physical interface is, but
This happens only on relaod. You can disable reload routine in chan_h323.c
...
Martin
On 1 Sep 2003, Michael wrote:
I'm running the CVS from last week and from day one (over 4 months now)
I've had this problem where asterisk core dumps when using chan_h323.
It appears to be a problem with
: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
Server: DTA SIP/0.11.8 NNOS/VR30
Content-Length: 0
8 headers, 0 lines
Message is NOTIFY
hm3*CLI
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Saturday, August 30, 2003
Try canreinvite=no
Martin
On Tue, 2 Sep 2003, Zac Sprackett wrote:
I'm having a problem getting outbound trunking to work using asterisk
and an external SIP FXO.
7 digit dialing produces the following output:
-- Executing Dial(SIP/mitel-fe17, SIP/[EMAIL PROTECTED]) in new stack
--
1. Are supplementary services like conferencing, call brokering or call
forwarding supported by * ?
Conferencing (check MeetMe application), cal brokering ??? call forwarding
you can do that by having a little script in extensions.conf (unless
you're using FXS ports, where you can use
Post the sip debug .. maybe someone will help you.
Martin
On Sat, 30 Aug 2003, Andrew Joakimsen wrote:
Has anyone been successful in using the DTA310 as provided by Packet8 to
work with asterisk? I have gotten it to register with Asterisk but
whenever I try to dial a call all I get is
Press flash on your phone (asterisk will intercept that) and then when you
have a dialtone press *0 then asterisk will send the flash to PSTN line.
regards
Martin
On Thu, 28 Aug 2003, Carlton J. O'Riley wrote:
I was wondering if anyone is able to use the three way calling features from
their
What's the Spain busy tone ? x ms tone, y ms of silence etc ...
Martin
On Thu, 28 Aug 2003, Stuart Hirst wrote:
Has anyone tried to use * and an X100P in Spain.
I have enabled busydetect in zapata.conf but still its not detecting
busy correctly. I guess that this is because the busy
Try to cvs update or edit /etc/asterisk/modems.conf and comment out
driver=aopen line.
regards
Martin
On Thu, 28 Aug 2003, John Congdon wrote:
I had the same problem, and yes I am using cdr_mysql.
What should be done?
John
On Thursday, August 28, 2003, at 12:44 PM, Martin Pycko wrote
check 'help'
include contexta in contextb
regards
Martin
On Wed, 27 Aug 2003, Rattana BIV wrote:
hi,
how can I add or remove this line include = context by the command CLI ?
regards
Rattana
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
zaptel.h:#defineZT_DEFAULT_FLASHTIME750 /* 750 ms default
flash
time */
Martin
On Wed, 27 Aug 2003, Andy Hester wrote:
Anyone know offhand what the default flash time is? Where to find and
adjust if necessary? Going to test out some analog sets with * and wanted
to know.
What version of chan_zap.c do you have ?
grep chan_zap /usr/src/asterisk/channels/CVS/Entries
regards
Martin
On Mon, 25 Aug 2003, John Congdon wrote:
Does anyone know what this means?
WARNING[229391]: File chan_zap.c, Line 5731 (pri_dchannel): Ring
requested on channel 3 already in use on
try adding
fromdomain=externalip
to your sip entries
Martin
On Fri, 22 Aug 2003, Yehiel Samson wrote:
I wanted to know if it is possible to change the contact info so it would be
[EMAIL PROTECTED] instead that [EMAIL PROTECTED]
If this is possible could you please give me the info.
tester (a Sunset T10) and I can also successfully connect to my Asterisk
box and place an inbound call. I just can't connect the Asterisk box
directly to my PBX. The PBX is a Mitel 3300, by the way.
Thanks for your help!
Barry
-Original Message-
From: Martin Pycko [mailto:[EMAIL
Why ?
Martin
On Tue, 19 Aug 2003, John Todd wrote:
If anyone out there has an ATA-186 that they purchased but cannot use
with Asterisk due to it's being locked by Vonage, please contact me
off-list.
JT
___
Asterisk-Users mailing list
[EMAIL
On Tue, 19 Aug 2003, Andrew Joakimsen wrote:
Maybe he figured something out
and ... :)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
zapata.conf
signalling=pri_cpe ;90% if not, then pri_net
switchtype=euroisdn
channel = 1-15,17-31
Martin
On Mon, 18 Aug 2003, Nicolas Cartron wrote:
Folks,
everything's in the subject, i've got a Linux Box with a Digium E100P
E1 Card, modules are loaded, but I don't know which signalling
First of all you should have callprogress=no and immediate=no for any kind
of a PRI. Also why is your d-channel going down ? Can you send a trace
pri intense debug span 1 ?
regards
Martin
On Mon, 18 Aug 2003, Barry Porch wrote:
I managed to get Asterisk working with my PBX using T1, now I am
ast_pthread_muxtex_* functions were changed to ast_mutex_* for the
possibility of debugging the mutexes with gdb.
regards
Martin
On Sun, 17 Aug 2003, Michiel Betel wrote:
Did something change in lock.h lately? I get all kind of ast_mutex errors
when trying to compile chan capi 0.24c with
fine.
Could it be a hardware problem?
Thanks in Advance
Eduardo
On Fri, 1 Aug 2003 16:11:12 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
Try to uncomment in zaptel/Makefile
KFLAGS+=-DNO_CALIBRATION
and make clean install
that should help
Martin
On Fri, 1 Aug
Only if you have another FXS port and a real modem connected to that and
you bridge the call between FXS and FXO in asterisk.
regards
Martin
On Thu, 14 Aug 2003, Dan wrote:
Hi,
There is any possibility to define a virtual extension on the asterisk box
to act as a local modem?
This is the
It could work if it would be coming over g711 and you'd have
dtmfmode=inband set for that call
regards
Martin
On Thu, 14 Aug 2003, James Golovich wrote:
On Thu, 14 Aug 2003, Eduardo Goncalves wrote:
I'm using G.711alaw.
My extensions.conf:
===
[globals]
)
exten = 1001,2,Voicemail2(u1001)
As soon as the Voicemail picks up the NOTICE line appears multiple times on
the console.
Thanks,
Ricardo
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 05, 2003 3:57 PM
Subject: Re: [Asterisk
This is some routine that comes with older versions of MySQL. You need to
find out what happened to it .. maybe they substituted it with somehting
else ...
regards
Martin
On Thu, 14 Aug 2003, Jerk Face wrote:
I updated asterisk this morning cvs update -dA
When I try to run Asterisk (asterisk
It's easier to check that up ...
On Wed, 6 Aug 2003, Jim Mercer wrote:
On Wed, Aug 06, 2003 at 09:59:18AM -0500, Martin Pycko wrote:
You're looking for libncurses-dev and in libpri you can remove -Werror
from libpri/Makefile or cvs update libpri (it should be fixed)
fixed
use rfc2833. G711 is only
needed if you are passing DTMF inband.
bkw
On Thu, 14 Aug 2003, Martin Pycko wrote:
It works only with G711 (ulaw/alaw)
regards
Martin
On Thu, 14 Aug 2003, Dan wrote:
Hi,
I cannot use '#' to initiate transfers
obvious.
Regards,
Ricardo Villa
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 05, 2003 4:25 PM
Subject: Re: [Asterisk-Users] Wierd Message
Can you send a trace from your screen after you turn of the debug in
/etc/asterisk
try in zapata.conf
usecallerid=no before the definition of channel = a
regards
Martin
On 13 Aug 2003, Mark Farver wrote:
On Tue, 2003-08-12 at 21:59, Steven Critchfield wrote:
To answer on the first ring turn off callerid support. If you need
callerid support and answering on the first
I recommend installing the same version of mysql and mysql-devel and then
recompiling the cdr_mysql.c
regards
Martin
On 7 Aug 2003, Johanna Kangas wrote:
Hey,
Have i done something wrong or is there something wrong with latest CVS
and cdr_mysql, cause after checking out latest CVS today, I
It works only with G711 (ulaw/alaw)
regards
Martin
On Thu, 14 Aug 2003, Dan wrote:
Hi,
I cannot use '#' to initiate transfers.
I have tried on different phones (7960, ATA, X-Lite).
When I press '#' during a call, nothing happen.
I have both T and t switches in Dial application.
The
Did you try BUSYDETECT_MARTIN in asterisk/Makefile ?
regards
Martin
On Thu, 14 Aug 2003, Andy Powell wrote:
Hi Dave,
I have a similar problem, I tried using busydetect and busycount but calls kept
being dropped
at random intervals. It didn't seem to matter what i set the busycount to. I
One could imagine that you could just use the ATA Vonage box creating some
fake routes to your system ... so that if the box tries to contact
gateway.com you could point that to asterisk and make it work using the
current vonage setup. I would also recommend adding a rule to firewall
that your ATA
On Wed, 13 Aug 2003, Chee Foong wrote:
Hello,
I have a E100P card from digium and I try to implement a conference
bridge in asterisk.
I wonder since I got the E100P card do I still need to load ztdummy
for caller from h323 endpoints to work with Meetme?
It's not necessary.
I load the
For PRI-*-fax over FXS
It's as simple as having the fax extension the the incoming context
associated with the PRI channels. With PRI channels we can hear the fax
before we even answer (in most cases)
regards
Martin
On Tue, 12 Aug 2003, Adams, Gavin wrote:
From: Martin Pycko [mailto:[EMAIL
Yes, all does.
regards
Martin
On Mon, 11 Aug 2003, Michiel Betel wrote:
Simple Q but I can't find the answer in the archives (and am too lazy to
look in the source, but then its 32 Celcius here...
Do all digium cards provide the zapata timing? e.g. also the analogs
(including the X100P) or
On analog ports you need to Answer
Ringing
Wait,2
and then do something .
That should detect faxes.
regards
Martin
On Mon, 11 Aug 2003, Tilghman Lesher wrote:
On Monday 11 August 2003 03:26 pm, Eduardo Goncalves wrote:
On Mon, 11 Aug 2003 15:15:08 -0500
Tilghman Lesher [EMAIL
Sometimes you need to start asterisk from
/usr/src/
or /usr/src/asterisk/
and also in console mode for g729 code to work.
But I've heard that safe_asterisk also works.
regards
Martin
On 8 Aug 2003, Eric Wieling wrote:
I'm getting the following message when I start Asterisk:
WARNING[1024]:
Try first to stop it :
asterisk -rx stop now
then killall -9 asterisk
On Fri, 8 Aug 2003, Jim Friedeck wrote:
How do I stop asterisk when it is in a bad mood? It keeps dialing
extensions and won't listen! I tried kill PID. No go. I don't want to
have to reboot again. Thanks.
Jim Friedeck
On Fri, 8 Aug 2003, Richard Scobie wrote:
I have been running busydetect=yes, using BUSYDETECT_MARTIN and am
having hangups during calls.
If you use also BUSYDETECT_TONEONLY then you can detect tones that are
irregular, eg: 200 ms of tone, 200 ms of silence, 200 ms of tone, 500 ms
of silence,
What if someone adds your number to that list ?
Someone would have to moderate it.
regards
Martin
On Tue, 5 Aug 2003, McAughan, Matt wrote:
Does anyone keep a known telemarketer caller id database? If not has anyone
proposed an Asterisk community project to share this information? Sort
You're looking for libncurses-dev and in libpri you can remove -Werror
from libpri/Makefile or cvs update libpri (it should be fixed)
regards
Martin
On Wed, 6 Aug 2003, Rhys Hopkins wrote:
Hi,
I am having trouble building and installing libpri and asterisk on my
system. Zaptel seemed to
exten = _9X.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
regards
Martin
On Thu, 7 Aug 2003, David Hindmarsh wrote:
Hi
Is it possible to use a sip gateway as a trunk.
If so, how would I do this
David Hindmarsh
- Original Message -
From: Jamie Carl [EMAIL PROTECTED]
To: [EMAIL
use extensions reload CLI command
Martin
On Wed, 6 Aug 2003, Steven J. Sobol wrote:
Is there a way to make * forget that SIP phone
[EMAIL PROTECTED] is registered? I ask because I have a few
different PSTN numbers that I use for various reasons, and I can reprogram
my Grandstream, but
well should be ok if you cvs update now.
Martin
On Wed, 6 Aug 2003, Rhys Hopkins wrote:
Martin Pycko wrote:
You're looking for libncurses-dev and in libpri you can remove -Werror
from libpri/Makefile or cvs update libpri (it should be fixed)
Thanks for that - I installed ncurses-devel
modprobe wcfxo
cd /usr/src/asterisk
/usr/bin/screen -d -m /usr/sbin/asterisk -vvvcn
if /usr/src/asterisk doesn't work try to start it from /usr/src
You may also try to download the new_codec_binary from our ftp site.
regards
Martin
On Wed, 6 Aug 2003 [EMAIL PROTECTED] wrote:
Hi,
What's
It means that some application scheduled an execution of some routine in
the past, eg: it will never be executed since it's way in the past ...
regards
Martin
On Tue, 5 Aug 2003, Ricardo Villa wrote:
Hi,
Whenever someone leaves a Voicemail in our system we get this message on the
console:
Don't use %'s with txgain/rxgain
for
txgain=5% is equal to txgain=5.0 and that might be too much
On Tue, 5 Aug 2003, WipeOut . wrote:
could you send me the exact syntax for rxgain / txgain?
I think that might help towards my problem
becuase i'm having to turn the handset volume all
Are you experiencing it over PRI ? Can you send the pri debug span
spanno trace ?Is your asterisk/libpri code very recent ?
regards
Martin
On Sun, 3 Aug 2003, Stefano Finetti wrote:
Mark,
I'm now able to send proper DTMF tones checking on the isdn driver and using
rfc2833 as dtmf mode for
Typically you use AbsoluteTimeout app.
Martin
On Sat, 2 Aug 2003 [EMAIL PROTECTED] wrote:
hi everybody,
can anybody pls tell me a way to hangup an answered Zap/SIP/IAX call, after a
specified time period
expires, like after 10, 15 minutes.
Surajee
--This mail sent
Well when you use your phone line and you hear the call waiting sound you
can press flash on your phone and then *0 and that will generate the flash
on your phone line. This switch to the incoming call.
regards
Martin
On Sat, 2 Aug 2003, lists wrote:
I have a x100p card that has call waiting
It doesn't look like a problem. It's that when you have so many calls ...
execution of some piece of code protected by mutex takes longer so it
happens that some calls wait for their time . I guess if you have too
many of those messages you should disable them.
regards
Martin
On Thu, 31 Jul
What does 'dmesg' say ?
On Fri, 1 Aug 2003, Eduardo Goncalves wrote:
Hi list,
I'm trying to set up a TDM40B, but modprobe returns the fowling errors:
asterisk:~# modprobe wcfxs
ZT_CHANCONFIG failed on channel 1: Invalid argument (22)
/lib/modules/2.4.18/misc/wcfxs.o: post-install
Try to uncomment in zaptel/Makefile
KFLAGS+=-DNO_CALIBRATION
and make clean install
that should help
Martin
On Fri, 1 Aug 2003, Eduardo Goncalves wrote:
On Fri, 1 Aug 2003 15:34:23 -0500 (CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
What does 'dmesg' say ?
CSLIP: code copyright 1989
Try to do the same in shell. Does it work ?
Martin
On Thu, 31 Jul 2003, Dan wrote:
Hi,
When I try to run the command wmix to mix two WAV files recorded by the
Monitor application I get the following warning in the console and the macro
exit at that point.
Running the command from a
One thing is sure: the system should return with 0 if it's successful.
Read man system
regards
Martin
On Thu, 31 Jul 2003, Dan wrote:
Something even more interesting.
I have tried to execute the command 'ls' in the following line:
...
exten = s,3,System(ls)
...
And this is the result
can prevent this type of
behaviour.
BR,
Dan
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Sent: Thursday, July 31, 2003 7:49 PM
Subject: Re: [Asterisk-Users] 'System' application exit with error even if
it performs the job
sip show registry is when asterisk registers with some gateway.
you want to look at sip show peers or sip show users.
regards
Martin
On Thu, 31 Jul 2003, Steve Woolley wrote:
I am trying to get SIP registrations to work within Asterisk. From my
snom 200 phone (and on my SJPhone soft client) I
It's fixed now
On Sun, 27 Jul 2003, Michael Bielicki wrote:
we have now perfect results with yesterdays cvs and the te410p
todays cvs allways thinks that immediate is set to yes in zapata.conf. weird
...
cheers
Michael
On Sunday 27 July 2003 7:12 pm, Mark Spencer wrote:
I put a TE410P
Do 'iax2 debug' to see more.
Martin
On Fri, 25 Jul 2003, Richard Scobie wrote:
A call is placed via IAX2 from one asterisk to another, to a TDM400
channel whose extensions.conf entry is
exten = 502,1,Dial(${COLIN})
exten = 502,2,Congestion
If this channel is already busy when called,
Try the new_codec_binary/codec_g729b.so from the digium ftp site.
regards
Martin
On Wed, 23 Jul 2003, Dan Fernandez wrote:
I am having some problems with g729 with SIP and ZAP channels.
1)
I have two g729 licences. Very frequetnly (I don´t know what triggers the error) I
get the
But you need to edit h323/Makefile and uncomment -DWANT_G729
Martin
On Mon, 21 Jul 2003, Jeremy McNamara wrote:
You should run chan_h323. It is distributed with Asterisk and works
with G.729 and any other codec asterisk supports TODAY. There is no
need to run a 3rd party driver.
Jeremy
Use application Background.
Martin
On 22 Jul 2003, Thilo Salmon wrote:
Hi,
is there a way to enable dtmf detection on zap channels? I am trying to
pickup, play a ringtone and the dial out. I.e.
exten = s,1,Wait,1
exten = s,1,Answer
exten = s,2,Playtones(dial)
exten = s,3,DigitTimeout,5
use BUSYDETECT_MARTIN in asteirsk/Makefile
Martin
On 22 Jul 2003, Brancaleoni Matteo wrote:
increase busycount in zapata.conf
busycount=6 is ok for me.
the default is 3 , I think, and sometimes
it hangsup on speaking (or some other moh ;) )
Matteo.
Il mar, 2003-07-22 alle 22:11, Paulo
It's fixed now. Aparently Mark forgot to compile before commiting.
Martin
On Tue, 22 Jul 2003, Ashley Jones wrote:
Hi all,
I'm trying to compile Asterisk (checked out of CVS at aprox 3pm PST 7.22.03)
on a 2U Compaq running Redhat 8 and 1 TCM400P(w/ 2 hot ports) and 2 X100P's.
The error I
I'm just curious if anyone has the X100P Callerid receiving working
outside US.
Replies are appreciated. Also if it's not working for you in a certain
coutry you can respond too.
regards
Martin
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
One can use the retrieve_extensions_from_mysql.pl script and then issue a
extensions reload command to asterisk. The pending calls are unaffected
and the final substitution of the new dialplan is done in a very short
time.
regards
Martin
On Tue, 22 Jul 2003, Jeremy McNamara wrote:
DynExtenDB
Try to install the new codec code that is available in
ftp://ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary/codec_g729b.so
place it in /usr/lib/asterisk/modules and restart asterisk (or try to
start it).
There is also a new command available g.729 show license usage and a few
fixes
.
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 21, 2003 9:03 PM
Subject: Re: [Asterisk-Users] anyone with X100P Callerid working outside
US ?
It's possible that your telco first transmits the DID (your number) and
then later
Yes, you can contact over the manager interface (you need to setup a
user/pass in /etc/asterisk/manager.conf). I've sent a short perl script
how to do that some time ago.
Now notice that extensions reload only renews extensions without
touching other modules.
regards
Martin
On Mon, 21 Jul 2003,
You need to have a pending call in the system (some extensions that is
ringing to test that). If you have 3 FXS ports try to place a call from
the first one to the 2nd and then instead of taking the 2nd off hook dial
*8 on the 3rd phone
Martin
On Thu, 17 Jul 2003, Jay Tyndall wrote:
Hi,
do you have in zapata.conf
busydetect=yes
or
callprogress=yes ?
Martin
On Thu, 17 Jul 2003, Paulo H. Mannheimer wrote:
Hi ,
I''m getting random hangups on zap channels with long calls. It seems that the
hungup happens after 10 minutes or so. AbsoluteTimeout is set to 0. Any other
thing I
of the connection
do not work either!!!
My incoming calls are coming from PSTN lines through an E1
so DTMF must be inline .. THe (thousands of) error messages
aren't really a problem, just annoying.
Dave
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED
You're trying to detect inband dtmfs from the codec stream.
Martin
On Tue, 15 Jul 2003, Dave Alan Caruana wrote:
hi ..
I have finally managed to get Chan_H323 G729 working
flawlessly, thanks to some help from Jerry McNamara.
For those out there who are stuck with the same problem
the
Your telco doesn't send you this IE
-- Processing IE 112 (Called Party Number)
Martin
On Tue, 15 Jul 2003, Cristi wrote:
I see the following line into debug (pri debug span 1):
1. Progress Description: Calling equipment is non-ISDN. (3) ]
2. Calling Number (len=14) [ Ext: 0 TON: Unknown
Check if the board is still getting interrupts.
grep wcfxs /proc/interrupts; sleep 10; grep wcfxs /proc/interrupts
should show two numbers that differ by ~1.
regards
Martin
On Tue, 15 Jul 2003, Jay Tyndall wrote:
Hi,
I have got my TDM400P working.(3 modules), asterisk dials Zap/1 and
Unfortunatelly if your telco doesn't send you any DID along with the SETUP
message you need to have immediate=yes in zapata.conf for those channels.
regards
Martin
On Fri, 11 Jul 2003 [EMAIL PROTECTED] wrote:
Very sorry about the previous mail,
heres the mail again,
hi Everyone,
We are
to Martin Pycko ...
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Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
--This mail sent through OmniBIS.com
Sure you just need to use Monitor and Changemonitor apps.
A little bit of scripting is a must though to get a unique id
eg a current date in seconds. I'm not sure if asterisk has it already.
regards
Martin
On Fri, 11 Jul 2003, Erik Kendall wrote:
Can Asterisk automatically record all
T100P handles the EM wink start signalling as well as D4AMI
framing/coding.
The config in /etc/zaptel.conf
span=1,0,0,d4,ami
em=1-24
in /etc/zapata.conf
[channels]
signalling=em_w
context=incoming
group = 1
channel = 1-24
But read more and have all the keywords/options that you need added to
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