Well first of all if you're outside of US or callprogress-supported zones
then you can use only busydetect. And that will only work if after the
remote hangup your telco gives the fast-busy or any type of busy. You can
tweak the duration of tone/pause and increase the count and it *will*
work
all T1/E1 boards do
regards
Martin
On Wed, 30 Jun 2004, Gonzalo Mateos wrote:
Hi there,
I'm quite new to asterisk and digium hardware. I needed to know which of the digium
cards supports EM signaling?.
thnaks,
Gonzalo
---
Outgoing mail is certified Virus Free.
Checked by AVG
notransfer might be still a [global] only keyword for IAX2.
regards
Martin
On Tue, 20 Apr 2004, Hans-Henrik Andresen wrote:
Hi,
Another one.
I got notransfer=yes i iax.conf for both 2109 and dialout, but I still get
this in my logfile
Attempting native bridge of [EMAIL PROTECTED]/5 and
it looks like some other usb module tries to get loaded and that's what
causing it.
try to insmod the zaptel tor2 run ztcfg -vv instead.
or rmmod all the uhci modules...
regards
Martin
On Fri, 16 Apr 2004, Jim Gottlieb wrote:
We use dual Athlon machines with up to three T400P 4-span T1
try to do ps -auxm to list all the threads of the asterisk.
Then connect with gdb to the thread that takes 99% of CPU and find out
what it's doing.
Martin
On Mon, 22 Mar 2004, Bill Hamlin wrote:
Nope same problem. I just started it and did a couple of ps aux's and got
this output:
[EMAIL
You can also do R1 to do descending round-robin. Same with G1 and g1.
Martin
On Wed, 18 Feb 2004, Steve Creel wrote:
I've not seen it documented anywhere, but scrolled past it the other day
in chan_zap.c.
Apparently you can specify a zap group with an 'r' instead of a 'g' to use
the group
I wonder if that'll work only with Pingtel phones *smile*.
Martin
On Wed, 18 Feb 2004, Lenny Tropiano / asterisk.org Mailing list wrote:
I just read that Pingtel (www.pingtel.com) will be releasing it's IP PBX (which
runs under Linux) to open source (similar model to Redhat Linux, charging
you don't have libm (m for math) library ?
Martin
On Thu, 15 Jan 2004, [gb2312] Âí÷ë wrote:
erro cocent:cc -shared -Wl,-soname,libtonezone.so.1 -lm -o libtonezone.so.1.0
zonedata.lo tonezone.lo
/sbin/ldconfig -n .
ln -sf libtonezone.so.1 libtonezone.so
cc -o ztcfg ztcfg.o -lm -L.
are you running safe_asterisk ?
If so try to modify safe_asterisk ... CONSOLE=yes to CONSOLE=no
or if not
list all the asteirsk threads 'ps -axum | grep asterisk'
find the thread that takes the most CPU and connect with gdb
gdb /usr/sbin/asterisk pid
and do 'bt'
and post the last few lines
sure, use the 'n' option of the queue and put voicemail app as the next
priority
Martin
On Fri, 9 Jan 2004, Ken Alker wrote:
I am just studying Asterisk now and have a question. Is it possible to
force anyone who enters a queue into voice mail after they have been in the
queue for 30
busydetect should help you. Set busycount=10 busydetect=yes in zapata.conf
and measure the length of the tone .. should be equal the pause too.
Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like
this: your result - 100, your result + 100 [ms]
regards
Martin
On Fri, 2 Jan
try ztmonitor 1 -v
Martin
On Sat, 20 Dec 2003, Daniel Bichara wrote:
Hi,
I am trying to run ZTMonitor to get debug info from my E100P board but I
got the following message:
-bash-2.05b# ./ztmonitor 1
Unable to open /dev/dsp: No such file or directory
Cannot open audio ...
-bash-2.05b#
You need to have HDLC generic support compiled into your kernel ... I
think it's not good to have it compiled in modules ... just embedded in
kernel.
Martin
On Sat, 20 Dec 2003, Daniel Bichara wrote:
Hi All,
I wish to connect * to a Cisco using a E100P board.
When I load the driver I got
The registry expires after sime time. You can set the default expirey and
max in sip.conf. It's up to your phone/sip device to reregister after the
registration expires.
Martin
On Mon, 22 Dec 2003, Jonathan Tew wrote:
We have people connecting to an asterisk box over the internet. They're
It doesn't matter for the zaptel (since you can set dchan=any_channel) but
in chan_zap.c in asterisk dchannel for t1 cards is hardcoded to by on 24th
channel. You can change that though.
regards
Martin
On Thu, 18 Dec 2003, Michael Welter wrote:
We have contracted with Eschelon to provide
Did you ifdown the dynamic interfaces first ?
Martin
On Wed, 17 Dec 2003, Steven Critchfield wrote:
On Wed, 2003-12-17 at 10:36, john wrote:
Hi,
I have just begun working with TDMoE running between 2 fiber nics the
dynamic span works great. In my main asterisk box's startup file I just
/dev/zap/1
Martin
On Thu, 11 Dec 2003, Paulo Mannheimer wrote:
Sorry to bother again, but what is the syntax of a dchannel? I'm trying
1, zap/1, ... without success
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer
Sent: quarta-feira,
two d channels of two separate pris
Martin
On Wed, 10 Dec 2003, Paulo Mannheimer wrote:
Hi All,
Can anyone tell me what are the dev1 dev2 parameters that I should
use to run pridump? I took a look at the source code but couldn't figure
this one out.
Best,
PauloHM
most propably the globalnat is nat= defined in the [general] section.
Martin
On Wed, 10 Dec 2003, Andrew Thompson wrote:
Can someone tell me what this setting is supposed to be?
peer-nat = globalnat;
It looks like it's inheriting a parameter, but I'm curious, is globalnat an
option that
check the manager interface ... you can transfer the active call to some
other extension. (redirect). If these are zap channels there is
zaptransfer command and zapdialoffhook via the manager.
regards
Martin
On Wed, 3 Dec 2003, Alistair Cunningham wrote:
Does Asterisk have the capability to
Don't use dtmfmode=inband on GSM codec it'll only work on G711.
Martin
On Mon, 1 Dec 2003, Bartosz Jozwiak wrote:
What does it mean ??
WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2
frames
WARNING[265236]: File dsp.c, Line 1198 (ast_dsp_process):
You might need to edit the code of chan_zap.c You need two things to fix:
outgoing calls and incoming calls. Outgoing you should be able to find
pri_call call and do chan-1 for chans16. For incoming calls you need to
find the handling of PRI_EVENT_RING or something like that and do chan+1
for
Do you have up to date libpri and asterisk ?
Also it'd be good if you could send pri debug span 1 (or 2) trace.
regards
Martin
On Tue, 25 Nov 2003, Alastair Maw wrote:
I'm running an E400P. Every now and then Asterisk stops receiving
incoming calls.
This turns up in the messages log:
OK, that was obviously a 'typo' ... It's fixed.
Martin
On Tue, 25 Nov 2003, Detlef Wengorz wrote:
Daniel Chabrol wrote:
Hi List!
I get WARNING[14351]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 =
524300 is not codec1 = 524300, can't do reinvite at my asterisk console.
The
check 'show dialplan nonauthenticated'
regards
Martin
On Fri, 21 Nov 2003, James Sharp wrote:
I've got a couple of PRIs coming in from a SUMA 4 switch with some 800
numbers routed through it.
When the calls come in, I get the following message on the console and the
call never makes it
Did you place echocancel=yes before the definition of the channel with
channel keyword in zapata.conf ?
regards
Martin
On Wed, 19 Nov 2003, Elijah Chancey wrote:
I've got an X100P a cisco 7960. if i call from an analog line via the
x100p to the cisco, there is an overly audible echo on the
Try to use Background application at s,1
Martin
On Fri, 14 Nov 2003 [EMAIL PROTECTED] wrote:
Hi
I'm tring to use asterisk as IVR. But I have trouble when I
recieve fax.
When I recieve fax, asterisk show message to looks redirect
incoming fax to fax extension. But scripts in fax
extension
Try again ... with latest CVS.
Martin
On Fri, 14 Nov 2003, Jordi Haarman wrote:
Hi,
I would like to be able to switch dtmf mode of SIP calls of local
clients so the server can understand them and it can also be used when
connected to a remote location. I saw that there is an application
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Friday, November 14, 2003 6:26 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] dtmfmode SIPDtmfMode
Try again ... with latest CVS.
Martin
On Fri, 14 Nov 2003, Jordi Haarman
make sure the modules for your boards are loaded.
ztcfg -vv shouldn't return with any errors.
regards
Martin
On Fri, 14 Nov 2003, Matt Lawson wrote:
I've just finished updating my Asterisk to the CVS version.
Unfortunately, chan_zap won't load anymore.
The hardware has not changed and the
what does show dialplan incoming show ?
Also try using Dial(Zap/bla,10) instead
Maritn
On Thu, 6 Nov 2003, Lal, Deepak (Contractor) wrote:
When I get a SIP call, I get the following error:
*CLI NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is
Check if you configured the clocking from their circuit correctly. You
need to have span=1,1 ... in zaptel.conf
Martin
On Tue, 4 Nov 2003, Eduardo Goncalves wrote:
On Mon, 3 Nov 2003 17:15:21 -0600
Don Pobanz [EMAIL PROTECTED] wrote:
Sometimes I receive a Red Alarm in my E1 trunk
It should. YOu need to do port forwarding on the firewall and use externip
not externalip in general section of sip.conf. Refer to
asterisk/configs/sip.conf.sample
Martin
On Tue, 4 Nov 2003, Leif Madsen wrote:
I'm just curious if I was to place my * box behind a a FW/NAT box
running linux, if
If you use TE410P make sure you have a recent zaptel from CVS.
Martin
On Tue, 4 Nov 2003, Eduardo Goncalves wrote:
On Tue, 4 Nov 2003 09:42:36 -0600 (CST)
Martin Pycko [EMAIL PROTECTED] wrote:
Check if you configured the clocking from their circuit correctly. You
need to have span=1,1
You can port forward the 5060 SIP port and use externip keyword in
sip.conf to have it working behind a NAT.
Martin
On Mon, 3 Nov 2003, WipeOut wrote:
Robert Mann wrote:
Problem I have is this. outside firewall (extension 2003) can call me
inside firewall (extension 2000) and all is
Maybe you need the straight through cable.
Martin
On Mon, 3 Nov 2003 [EMAIL PROTECTED] wrote:
Hi,
At least I have one E1 to test my E100P.
My telco company in Spain has installed one LiteSpan 1540 NT (UTR 2M)
I make a crossover cable between E100P and UTR.
1 - 4
2 - 5
after loading
It's new. It prevents asterisk from putting the private IP in the messages
that asterisk sends with SIP.
Martin
On Mon, 3 Nov 2003, WipeOut wrote:
Martin Pycko wrote:
You can port forward the 5060 SIP port and use externip keyword in
sip.conf to have it working behind a NAT.
Martin
, WipeOut wrote:
Martin Pycko wrote:
You can port forward the 5060 SIP port and use externip keyword in
sip.conf to have it working behind a NAT.
Martin
Martin,
Is externip and new parameter??
Does it do a similar thing for the server as what nat=yes does
this variable is refreshed!
This was taken from the CVS Viewer at: http://asterisk.espia-net.net/
chan_sip.c:
http://asterisk.espia-net.net/horde/chora/co.php/asterisk/channels/chan_sip.c?login=2r=1.204
-
Andrew Thompson
- Original Message -
From: Martin Pycko [EMAIL PROTECTED
Are you guys using voicemail2 ?
Martin
On Mon, 3 Nov 2003, Philipp von Klitzing wrote:
Hi!
The voicemails servermail and fromstring variables should change
default
values when email voicemail notification gets received by user.
I change it, but received mail still shows Asterisk PBX
It's not for phones, it's for asterisk behind a NAT.
Martin
On Mon, 3 Nov 2003, Robert L Mathews wrote:
At 11/3/03 10:00 AM, Martin Pycko [EMAIL PROTECTED] wrote:
Is externip and new parameter??
It's new. It prevents asterisk from putting the private IP in the messages
that asterisk
I'd suggest your telco doing loopup and checking the circuit.
regards
Martin
On Mon, 3 Nov 2003, Eduardo Goncalves wrote:
Hi list,
Sometimes I receive a Red Alarm in my E1 trunk (EM immediate start
signaling), and just few seconds after this, all alarms are cleared.
This
Try starting asterisk from /usr/src/asterisk
with the console
asterisk -vvvcng
regards
Martin
On Fri, 31 Oct 2003, Bartosz Jozwiak wrote:
I just download a new one!
And now I have that, it is even worser
WARNING[16384]: File codec_g729b.c, Line 202 (lintog729_new): No available
g729
[extensions.conf]
exten = 123456,1,SetVar,SIP_CODEC=ulaw
exten = 123456,2,Dial(${TRUNK}/${EXTEN})
The problem is with the SetVar function, the debug shows that the
function is executed, but after that, * sends the media capability to
the phone with g729 as preferred codec.
SIP_CODEC
Calling Number (len=12) [ Ext: 0 TON: International Number (1) NPI: ISDN/Telephony
Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user number passed
network screening (1) '4330' ]
It might be that the number plan is international
check 'screen -d -m asterisk -vvvcng'
regards
Martin
On Thu, 16 Oct 2003, CW_ASN - Gus wrote:
Hi all:
I've just purchase some licences of G.729 codecs, and I like to bring up * using
/etc/rc.d/init.d script.
Does anyone knows how to start in the old way?
Thanks in advance,
Gus
This means that I need to run * in this way forever?
Gus
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 16, 2003 6:13 PM
Subject: Re: [Asterisk-Users] Starting * with G729 licences
check 'screen -d -m asterisk -vvvcng
It's an application and not a cli command, put it in extensions.conf
[default]
exten = s,1,System(ls /tmp/log)
regards
Martin
On Tue, 14 Oct 2003, Chee Foong wrote:
Hello mate,
I tried that, i get No such command 'System(ls)'. I can't even make it work
on CLI.
I am able to execute linux
=as787ccf10
Supported: timer
To: sip:[EMAIL PROTECTED];tag=02f8-f0f0f208
Server: ipDialog SipTone 1.2.0 rc V UAS
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,SUBSCRIBE,INFO,NOTIFY
Content-Length: 0
11 headers, 0 lines
localhost*CLI
- Original Message -
From: Martin Pycko [EMAIL
Use tabulator button for asterisk to help you guess the name.
regards
Martin
On Tue, 14 Oct 2003, Walker Haddock wrote:
I am trying to figure out the correct syntax for the CLI command SIP SHOW
CHANNELS. I have tried
SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is
If you do make config in the zaptel then it's going to be loaded during
bootup. Otherwise it's not being loaded unless you do 'modprobe wcusb'
regards
Martin
On 14 Oct 2003, tom wrote:
From -
Received: from rwcrmhc12.comcast.net ([216.148.227.85]) by
sccrmxc11.comcast.net (sccrmxc11)
With the musiconhold and SIP-SIP call it turnes out that you need to
disable silence supporesion on your phones/gateways since the timing is
taken from the coming stream (but only for musiconhold AFAIK)
regards
Martin
On Tue, 14 Oct 2003, Michael Ulitskiy wrote:
Hi,
I've found that neither
It means that this IE is not implemented in the libpri or is not very
standarized.
regards
Martin
On Mon, 13 Oct 2003, Marcel Prisi wrote:
Here is an example call (works) :
-- Executing Dial(SIP/25-e804, Zap/g1/0707038340) in new stack
-- Called g1/0707038340
-- Zap/1-1 is
13 - 12167 Berlin - Germany
fon: +49 30 79705392
fax: +49 30 79705391
iaxtel: 1-700-157-8753
email:[EMAIL PROTECTED]
http://www.junghanns.net/asterisk
Am Mon, 2003-10-13 um 17.24 schrieb Martin Pycko:
It means that this IE is not implemented in the libpri or is not very
I just tested fromstring and emailbody with voicemail2 and a farily new
code and it's working. I don't know what you're doing wrong ... but
something for sure.
regards
Martin
On Mon, 13 Oct 2003, John Todd wrote:
I would recommend then doing grep fromstring
However, the timezone is still not straight in the message body.
${VM_DATE} doesn't seem to use the timezone matching routines defined
by the user's tz= setting.
Well it's the task for those who add features to have a global-system
thinking. The emailbody was added way before the timezones ...
What protocol ? H323 ? Which channel driver ? chan_oh323 or chan_h323 ?
Martin
On Wed, 8 Oct 2003 [EMAIL PROTECTED] wrote:
Hi!
When I try to make a call with ohphone, that is the message I get:
Call to 06302 aborted, insufficient bandwidth
Can anybody tell me a solution or a reason why
Compile zaptel without PPP support or compile PPP support into your
kernel.
You can do the first in zaptel/Makefile
Martin
On Wed, 8 Oct 2003, Ron Arts wrote:
This is probably not a direct asterisk problem, but
I am quite at a loss here.
I am experiencing problems with zaptel drivers
Am
cd /usr/src/asterisk; make config; cd /usr/src/zaptel; make config
regards
Martin
On Tue, 7 Oct 2003, john lawler wrote:
Hi guys,
Thanks for your answers on my two questions yesterday. That's exactly
what I was looking for, sorry for not noticing it myself, but I'm still
getting
Yeah, I'd put usecallerid=no since I bet it's set by default as yes.
Martin
On Sat, 4 Oct 2003, Richard Scobie wrote:
Martin Pycko wrote:
take out usecallerid=yes in zapata.conf
Martin
Thanks Martin, but my zapata.conf is :
[channels]
echocancel=yes
echocancelwhenbridged=yes
What if you separate the fax machine channels to diffrent contexts that
don't call application Monitor ? It's for outgoing calls and for incoming
calls if you have certain extensions for faxes you can call StopMonitor
application.
regards
Martin
On Fri, 3 Oct 2003, Nicholas Romero wrote:
Is
take out usecallerid=yes in zapata.conf
Martin
On Sat, 4 Oct 2003, Richard Scobie wrote:
After some months of Make updates, I have just deleted my Zaptel and
Asterisk source directories and done cvs checkout 's of asterisk and
zaptel, in order to clean up the trees.
After re-installing, I
It's a WARNING, so if you want to know why your phone doesn't work you can
read it or ignore it.
regards
Martin
On Thu, 2 Oct 2003, Brian Capouch wrote:
Martin Pycko wrote:
We send SIP messages to that device up to 6-7 times and then we stop and
this message shows on the console
use quit or ctrl-D
Martin
On Thu, 2 Oct 2003, Andy Hester wrote:
This probably has an easy solution, but I found it yet. How can I get out
of a remote console after using ssh to get into the box, making changes,
reload etc. without stopping *?
Thanks in advance.
Sincerely,
Andy Hester
Because of the nature of busydetect algorithm busycount shouldn't be set
to less than 8. It's 10 by default.
Just imagine that you dial a number that is attached to some speed dial
key. It'll surely cause hangup if busydetect 8.
Martin
On Sat, 27 Sep 2003, Shaun Ewing wrote:
Hi All,
This
[incoming]
exten = _X.,1,DBGet(NEWCONTEXT=context/${CALLERIDNUM})
exten = _X.,2,Goto(${NEWCONTEXT},${EXTEN},1)
exten = _X.,102,Goto(allother,${EXTEN},1)
Martin
On Fri, 26 Sep 2003, Matt McIntyre wrote:
I was wondering if someone might be able to offer a suggestion to me
about how I might go
Asterisk does restart on idle channels every (I think) 20 minutes to
ensure that the remote switch treats the idle channels (on our side) as
idle. I don't get the channelid problem that you're reporting, maybe the
pri debug span span_no is a good idea to post.
regards
Martin
On Thu, 25 Sep 2003,
The call does not get compleated on the PRI so you should check the pri
debug span 1 on your 2nd box.
regards
Martin
On Tue, 23 Sep 2003, Thomas Haeger wrote:
I have tried it with a timeout and without...
here the * output for the first side:
-- Starting simple switch on 'Zap/3-1'
--
make sure the 'format=wav' in voicemail.conf
Martin
On Tue, 23 Sep 2003, Steve Totaro wrote:
I am getting the following error in Windows Media Player Version 9 when listening to
voice mails.
ClassFactory cannot supply requested class (Error=80040111)
Any ideas? I tried searching the
gdb /usr/src/asterisk core.6044
then 'bt'
Martin
On Tue, 23 Sep 2003, jerk face wrote:
I keep getting segmentation faults when I do a reload.
Here are the core file outputs from gdb:
(I have three of them and they produce the same
output)
(gdb) core core.6044
Core was generated by
actually
gdb /usr/sbin/asterisk core.6044, sorry
On Tue, 23 Sep 2003, jerk face wrote:
I keep getting segmentation faults when I do a reload.
Here are the core file outputs from gdb:
(I have three of them and they produce the same
output)
(gdb) core core.6044
Core was generated by
It does support it but you have to uncomment -DWANT-G729 in h323/Makefile
On Mon, 22 Sep 2003, Eric Wieling wrote:
I doubt that it's a codec problem. Maybe chan_h323 doesnt' support
G729. JerJer would know.
On Mon, 2003-09-22 at 04:55, Chee Foong wrote:
hello,
I have tried that but
The implementation of *72 is done for FXS port (the one that gives the
dialtone). However you could implement that with some extensions.conf
logic.
regards
Martin
On Sat, 20 Sep 2003, Rich Adamson wrote:
Martin,
That makes sense... but how would one actually use *72#, as an example,
when *
So 'zap show chanenl channel-no' shows that the echocan is turned on ?
Martin
On Sun, 21 Sep 2003, Asterisk PBX wrote:
Oh, I forgot to say, zaptel/wcfxo is compiled with:
KFLAGS+=-DECHO_CAN_MARK2
KFLAGS+=-DAGGRESSIVE_SUPPRESSOR
(and, Brian, my jack is wired correct..)
-Original
These functions are implemented only for chan_zap (zaptel hardware) and
work for FXS/FXO ports. Exception is *8 (remote call pickup) as far as I
know.
regards
Martin
On Fri, 19 Sep 2003, Rich Adamson wrote:
Someone recently posted the following list as functions built into *
*0# sends flash
Do you have silence in the channel when the remote user hangs up or busy
tone ?
If you have silence you can use maxsilence=x_seconds in voicemail.conf
with
Voicemail2 application and that will make sure the calls are hanged up
after x_seconds of silence in the channel.
If you have busy tone then
Can you send a pri debug span span_no trace ? Or do you have an analog
T1/E1 ?
regards
Martin
On Wed, 17 Sep 2003, denzel-infotechs wrote:
hi!
I've got a asterisk system running with around 50 per calls per minute. I've
connected * to internal pabx and outside telecom using E1 (ISDN
, since Asterisk obviously
detects SIP hangups correctly whether it's SIP to SIP or SIP to outside line.
The problem is really only when outside callers leave voicemail.
Thanks,
Chris
On Wednesday 17 September 2003 08:09, Martin Pycko wrote:
Do you have silence in the channel when the remote
The X100P together with asterisk does not support the distinctive ringing
detection on the line. Asterisk however can generate the distinctive ring
over FXS ports.
regards
Martin
On Tue, 16 Sep 2003, Robert Boardman wrote:
Hi
I've just signedup for Distinctive ringing on my PSTN line in the
Did you see /etc/asterisk/voicemail.conf ?
maxmessage=120 is 2 minutes
Martin
On Fri, 12 Sep 2003, Rich Adamson wrote:
Is there a way to limit the duration of any single voicemail recording?
I'd like to put a cap on that limit, say 2 minutes or whatever, for those
long winded
The Voicemail2 is better one, has more bug fixes, more functionality and
Voicemail (1) should stop existing soon.
regards
Martin
On Fri, 12 Sep 2003, Olle E. Johansson wrote:
While on the subject of Voicemail - what is the difference between
voicemail() and voicmail2() ?
The show
exten = _9X.,1,Dial(H323/[EMAIL PROTECTED])
If it's not working it's worth looking at the reson:
h.323 debug
h.323 trace 3
regards
Martin
On Fri, 12 Sep 2003, Cerrajetto wrote:
Hello:
I am testing Asterisk with oh323.
My question is: can Asterisk route some calls thru a second h323
So you don't receive any answer from the other side ?
Is the circuit in alarm ? Can they do remote loopup test ?
It might be that they don't have their D-channel turned on ...
Martin
On Fri, 12 Sep 2003, Alastair Maw wrote:
OK, so I've done this:
*CLI pri intense debug span 1
you can copy voicemail.conf.sample to be your voicemail.conf ...
Martin
On Fri, 12 Sep 2003, Olle E. Johansson wrote:
Steven Critchfield wrote:
On Fri, 2003-09-12 at 10:34, Olle E. Johansson wrote:
While on the subject of Voicemail - what is the difference between
voicemail() and
What does 'dmesg' says ?
Martin
On Fri, 12 Sep 2003, James Sharp wrote:
On Fri, 12 Sep 2003, Jim Paraschou wrote:
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
IAX2 uses 4569 UDP port.
You can see iax2 calls with iax2 show channels. Also you can send the
calls in IAX2 simply by Dial(IAX2/blahblah)
Also IAX2 is more recent, has more fixes and has the trunking mode to save
bandwidth if you're sending more than 10 calls to another destination.
regards
what does 'dmesg' says ?
Martin
On Fri, 12 Sep 2003, Jim Paraschou wrote:
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect
Because IAX2 in trunking mode adds the 10 bytes header ... So It might not
be a good idea if you're going to have only two calls.
Martin
On Fri, 12 Sep 2003 [EMAIL PROTECTED] wrote:
On Fri, 12 Sep 2003, Martin Pycko wrote:
Also IAX2 is more recent, has more fixes and has the trunking mode
Do you have an error about receiving the callerid ?
What happens when you pick up the Zap/2 phone ?
regards
Martin
On Thu, 11 Sep 2003, Alvaro Parres wrote:
Hi...
I have the next problem.. I have a FXO card with i can make calls but i cant
recive calls.
At the consol, i get the next
It should work but you need to do Goto(extensions,666${EXTEN},1)
Martin
On Wed, 10 Sep 2003, Ernest W. Lessenger wrote:
We are trying to implement area-code dialing in our asterisk PBX.
Basically: we will have a number of customers, who may be in different area
codes, that want to
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, September 08, 2003 10:32 PM
Subject: Re: [Asterisk-Users] Call Time out Problem-Very Urgent!
Do you have callprogress=yes in zapata.conf ? If yes, then comment it out.
Also you could
Do you have callprogress=yes in zapata.conf ? If yes, then comment it out.
Also you could send some trace from the console including pri debug span
span-no
Martin
On Mon, 8 Sep 2003, Surajee Ratnayake wrote:
Is it a problem with E1, bcos, when we dial a SIP extension from the same
asterisk
Hello Asterisk Community!
There have been some complaints made by those customers that purchased the
TDM400P board and it didn't work properly in their boxes. Digium promised
to swap such boards for the new - revised version and will keep the
promise. However since we were backordered we're
comment out register = user:[EMAIL PROTECTED]
from sip.conf
Martin
On Sat, 6 Sep 2003, fredrik chabot wrote:
Hello,
Is there any way to get rid of this message.
NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration
from 'sip:[EMAIL PROTECTED]' failed for
What does this step show on the CLI ?
exten = 1,1,SetVar(FOO=123**)
exten = 1,2,SetVar(CHECK=${FOO:-1:1})
? If you're going to see CHECK=* then there is a bug in = operator ...
Martin
On Fri, 5 Sep 2003, John Todd wrote:
I am having Yet Another Regular Expression problem, but this one
might
Examples I'd like to see:
1)
${FOO} contains 12345#
${HASH} contains #
something like this:
exten = 123,1,Gotoif($[${FOO} : 12345#]?2|102)
If ${FOO} contains the contents of ${HASH} anywhere, go to 2. If not, goto 102
exten= 123,1,GotoIf($[...???...]?2|102)
1.1)
If the
You could use ResponseTimeout together with Background instead of playing
silence files.
Martin
On Thu, 4 Sep 2003, John Todd wrote:
As has been noted before on this list, the Wait() application does
not listen for keystrokes from users. Many of you, like me, have
looping Background(),
It's defined in /etc/asterisk/parking.conf
and set by deafult as 700
Martin
On Fri, 5 Sep 2003, Dave Alan Caruana wrote:
what i'm asking is what is the key sequence
you have to dial for the transfer ..
it was something like *7# if I remember
well, I know I had it working, but the client
If you have 0.4 ms silence every 3 cycles then try to uncommnet
BUSYDETECT_TONEONLY in asterisk/Makefile and recompile.
regards
Martin
On Fri, 5 Sep 2003, Norberto Garcia Prieto wrote:
Martin Pycko wrote:
What's the Spain busy tone ? x ms tone, y ms of silence etc ...
If I
RTP ports are not applying to IAX/IAX2.
Martin
On Thu, 4 Sep 2003, WipeOut . wrote:
Yes, The RTP ports in * are configurable in rtp.conf..
The default is 1 - 2
Later
HI!
but when making iax2 calls, a packet monitor would only reveal this UDP port.
(Between two * servers)
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