o left-over mpg123 processes to clean-up.
This is legacy safe_asterisk code that is left to confuse newbies.
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rebuild asterisk. Get the .src.rpm package of Asterisk that you want to
modify, and make sure that a 'rpm --rebuild' of it works on your build
system.
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t appear to have the sources for the 2.6.18-8.el5 kernel installed.
That's not the kernel of Centos 5.3 . Certianly not one with "all
packages updated.
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On Mon, Oct 05, 2009 at 10:37:52AM +1100, Paul Hales wrote:
>
> Is inbound working?
Considering he has no channel of type 'ZAP' and he uses Zaptel: no.
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+972-50-7952406 m
t;
> Zaptel Echo Canceller: MG2
Have you actually loaded the module wctdm ?
>
>
>
> So that looks encouraging
>
>
>
> But still getting problem dialing out.
>
>
>
> Also quite worrying is that there are no light
do you mean by "doesn't work"?
Was app_jack.so built?
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5:06:34] == Parsing '/opt/etc/asterisk/sip_notify.conf': [Oct
> 3 15:06:34] Found
>
> NSLU2*CLI> sip show peers
> Name/username HostDyn Nat ACL Port
> Status
> NSLU/nslu (Unspecified)D
rly handle PRI TON and allow changing number (bug #3493, with
mods)
This refers to https://issues.asterisk.org/view.php?id=3493 .
So the default value was never "never".
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gt;
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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h
6] Found
> Segmentation fault (core dumped)
>
> What causes a segmentation fault ??
Not sure. But it's likely the module loaded after res_features.so .
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+972-50
is not really a valid dialplan app).
>
> But I don't see the changes in the file.
Is it writable to the user running Asterisk?
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38583 Jan 13 2009 sip.conf
> -rw-r--r--1 root root 317 Jan 13 2009 sip_notify.conf
> -rw-r--r-- 1 root root 6695 Jan 13 2009 sla.conf
Is /opt/etc/asterisk the compiled config directory (astetcdir)?
What is the contents of asterisk.conf?
--
problem reset by restarting asterisk? 'dahdi
resstart'?
Also, please look at the reply he gets:
0/1/0 (Busy/Congested/Not-Available) - the call was marked as
"congested". Is that the same for you?
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Tzafrir Cohen
icq#16849755 jabber:tzafri
/002 xpp_usb- e4e4:1161 Astribank-modular
> USB-firmware
The FPGA firmware is not loaded yet. What do you see in
/var/log/user.log ?
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_timing_pthread.
1.6.2-trunk: Unless you can use the timerfd interface (a relatively new
Linux distribution: kernel >= 2.6.27), same answer as for 1.6.1 .
1.6.2 also provides an initial rpelacement for app_meetme
(app_conceference)
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Tzafrir Cohen
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o
canceller on channel one. Enable the above echocanceller lines, or use a
single one for all of them.
But that's not your real issue.
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http://www
on of zaptel.h to build
ztcfg (or maybe way older modules and a newer ztcfg). A second option is
mixing 64bit kernel with 32bit userspace.
Also, what is the output of:
uname -a
which ztcfg
file `which ztcfg`
# let's hope this one exists:
cat /sys/module/zaptel/ve
If you changed anyything, use 'logger reload' in the CLI to
reload the settings.
Now try an incoming call.
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e debug logging and provide log of a call that disconnect
successfully and a log of a call that fails to disconnect?
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that is in place, either disconnect and reconnect the Astribank, or
use:
/usr/share/dahdi/xpp_fxloader usb
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d like the nicest interface to me. Choosing from a list of names
sounds more probable.
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__
On Thu, Sep 24, 2009 at 10:54:17AM +0200, Loic Didelot wrote:
> Not sure,
> how can I check, but older astribanks work pretty fine on that system.
ls /proc/bus/usb
What is the output of:
lsusb
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Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+
S}" = "NOANSWER"]?7)
> exten => _X., 7, Dial(SIP/9...@a-test2,20,ro)
> exten => _X., 8, GotoIf($["${DIALSTATUS}" = "ANSWER"]?9)
> exten => _X., 9, Hangup
Does this actually work? With the extra space after the ','?
--
Tz
rl/5.8.8/Zaptel/Hardware/USB.pm line 108.
Is usbfs mounted under /pruc/bus/usb ?
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in the bristuff
installation or to apply the relevant patches from bristuff to this
tarball.
Note that in latest DAHDI, wcb4xxp supports all the devices supported by
qozap.
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e how ethernet over USB
works and how TDM over Ethernet (MF) works, I would speculate that it is
far from flooding the USB bus.
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ttp://svn.asterisk.org/svn/zaptel/branches/1.4/build_tools/zaptel_svn_tarball )
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but then again, you'd want to use
libgsm from Debian anyway :-) (the fix is to tell the configure script
not to attempt any special platform-specific optimizations in the case
of armv5). The latest releases of a number of days ago should include a
number of minor fixes that should help build on
TMF to compensate (because as a digital signal, DTMF is
> reliable, but when sent as audio, it might not be). What it probably
> indicates is that the DTMF sent to your system is _incredibly_ short, and if a
> DTMF detector is employed, it's possible that the DTMF audio is simply
t a problem.
> make[2]: Leaving directory `/usr/src/kernels/2.6.18-128.1.6.el5-i686'
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On Thu, Sep 17, 2009 at 01:27:19PM +0100, Gordon Henderson wrote:
> On Thu, 17 Sep 2009, Steve Underwood wrote:
>
> > Where do you think he might live? Antarctica? :-)
>
> Beirut, Lybia.
Beirut in Lybia? I thought Lybia only managed to snach Tripoly from
Lebanon.
--
that line.
You want to avoid calling if someone else is calling on a fax machine
connected to the same line?
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he line when it actually happens.
I have no idea about the Sangoma drivers, but reecnt in-tree DAHDI
drivers report this by raising a RED channel alarm if there's nothing
connected. This means that Asterisk won't try dialing through it.
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Tzafrir Cohen
icq#16849755
ing centos 5
With a custom kernel? Such a version string looks like an Ubuntu kernel.
On Debian and Ubuntu: aptitude install linux-headers-`uname -r`
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What E1 line parameters need
> to be specified in Asterisk(configuration files).
>
> They vary from country to country. What is difference in one countries E1
> and others E1.
Actually E1 parameters don't differ that much by country.
Do you have any specific examples?
--
On Mon, Sep 07, 2009 at 01:47:57PM -0400, Steve Totaro wrote:
> On Mon, Sep 7, 2009 at 10:09 AM, Tzafrir Cohen
> wrote:
>
> > On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote:
> > > On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen > >wrote:
> > >
On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote:
> On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen wrote:
>
> > On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote:
> > > On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen > >wrote:
> > >
> &
On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote:
> On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen wrote:
>
> > On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote:
> > > On Sun, Sep 6, 2009 at 10:47 PM, Research
> > wrote:
> > >
> >
This allows recording non-VoIP links, VoIP links where tapping is not
convinient, or more selective recording of VoIP calls.
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On Thu, Sep 03, 2009 at 05:34:30PM +0100, Joao Gomes Pereira wrote:
>
>
> Tzafrir Cohen escreveu:
> > On Tue, Sep 01, 2009 at 08:53:04AM -0500, Danny Nicholas wrote:
> >> This may be dumb and/or obvious, but did you do these steps?
> >> 1. dahdi_genconf dahdi mod
hdi-dummy .
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actually a cable plugged? Connecting it to a live system?
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g an interrupt with a usb
> bus?
lsusb reads information through /dev/bus/usb . I don't see how this can
generate any interrupts.
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On Fri, Aug 28, 2009 at 12:09:36PM +0100, Julian Lyndon-Smith wrote:
> getting some errors on my test system. this is 1.4 (Asterisk
> SVN-branch-1.4-r214194) with a 4 port T412p card.
What version of libpri?
DAHDI? Zaptel? What version?
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Tzafrir Cohen
icq#16
For the benefit of our readers, I will convert this back to a more
readable in-line replies format:
On Thu, Aug 27, 2009 at 11:45:10AM -0500, Danny Nicholas wrote:
> On Thursday, August 27, 2009 11:36 AM, Tzafrir Cohen wrote:
> >
> > Has this actually ever worked?
>
> yes it
> difference.
Has this actually ever worked?
Do you use the same zapata.conf / chan_dahdi.conf ? Do you have
'answeronpolarityswitch = yes' ?
To avoid confusion, consider making one a symlink to the other for the
time of testing.
--
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icq#16849755
erisk which did not help.
That's odd.
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to Asterisk)?
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an to
clear the alarm. You should be able to see those events on e.g. the
"full" log of Asterisk. Do you see them?
> [ 5335.053981] wcte12xp0: Missed interrupt. Increasing latency to 9 ms
> in order to compensate.
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Tzafrir Cohen
icq#16849755 jabber:tzaf
pan is in alarm?
cat /proc/zaptel/1
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AHDI
> Transcoder Codec Translator)
>
> How to fix this ? Google was not my friend this time :)
I'm not sure how to avoid that specific message. Most people just don't
have a transcoder card and don't need codec_dahdi .
So basically it's harmless, unless you
gt;
> Try "sip show peer " to see which codecs are configured
> (allowed) for that peer.
>
> Some variant of "show channel " will show you what codecs
> are being used by an in-progress call.
sip show channels
iax2 show channels
Will also show you the codecs act
ader.
gsm is a compressed format, which means lower quality (with gsm the
quality is noticably lower than plain 8kHz audio). It also means that
unless your target cchannel is in gsm format, there'll be more
transcoding done along the way and hence more CPU time eaten.
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phony/sounds/
has some newer files.
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goes
> against my grain to not specify a canceller to mean use a harware one if
> it happens to exist.
Though this means you have to explicitly configure hardware echo
cancellers to work, which is not as before. This leaves even more room
for error.
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Tzafrir Cohen
icq#168497
ere is what zaptel.conf and zapata.conf used to be:
> /etc/zaptel.conf:
> loadzone = au
> defaultzone=au
> #TE110P
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-10
> dchan=16
Something is missing here...
http://docs.tzafrir.org.il/dahdi-tools/#_echo_canceller_modules
--
evel-* paackages) for that specific kernel.
Is it a self-compiled kernel?
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> Zaptel 1.4.12
> Add ons 1.4.9
Later versions of Asterisk 1.4 can also work with Zaptel.
Though if you're starting afresh, moving to DAHDI should be a good idea.
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ching for -lpq
> /usr/bin/ld: cannot find -lpq
> collect2: ld returned 1 exit status
> make[1]: *** [cdr_pgsql.so] Error 1
> make[1]: Leaving directory `/usr/src/asterisk-1.2.34/cdr'
> make: *** [subdirs] Error 1
uname -r
file /usr/lib/libpq.so
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i
manually.
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real warnings.
In most cases creating an empty file will do to hush that warning.
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__
Reminder: there are alternatives:
http://identi.ca/doc/faq
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___
.
3. No usage limitation. Freely usable. So I can point it out to someone
and ask him to try it.
I'm too lazy to start looking for such a file. And I bet somebody in
this list can point me to an existing one.
--
Tzafrir Cohen
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ten). The hierarchy goes from "beta"
> to "release candidate" to "released".
But there's a missing characterization here: "actively maintained".
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+972-50-795
t; No such command 'set verbose 1' (type 'help set verbose' for other
> possible commands)
>
>
> so how will i decrease verbosity level???
core set verbose 1
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+972-50-795240
s not picked up by that module:
that's a good question. Maybe you need to contact the support of the
vendor of the card (But what do I know about PCI cards, anyway? ;-) )
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icq#16849755 jabber:tza
dy bothered writing drivers for any of them.
>
> the digium single fxo cards and clones for about $10 ARE modems.
> you can get a sip gateway fxo + fxs in one box for about $50
At the time they were. Nowadays they cost more. Those specific modems
are now obsolete and not manufactured anymor
PSTN line, but generally more expensive. If you're in the US,
this is probably only meaningful if you need at least c. 8 "outgoing
lines" (maximal number of concurrent outgoing calls) but I figure this
generally varies wildly.
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icq#16849755
mp them in the directory it was run from.
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n an MS Windows client ?
> (e.g. "pbx-invalid.gsm")
> Regards
> H.Motamedi
sox path/to/file.gsm path/to/file.wav
Alternatively:
asterisk -rx 'file convert file.gsm file.wav'
Play the resulting file.wav on Windows.
--
Tzafrir Cohen
icq#16849755
> }
> Playback(beep);
> y=${x} + 1;
AEL is so easy that you managed to err with it :-p
> }
> NoOp(done);
> }
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+
[0] Well, not exactly, there's a run-time configurable option to change
it, but this is generally the default.
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y level 3 shows you the dialplan trace, changing it
to Verbose-6 would gain you nothing much beyou what you get from
Verbose-3.
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http:/
$_->type() eq 'FXO';
printf "%3d %s\n",$_->num, $_->battery;
}'
The "battery" method ATM only implemented on Astribanks in the perl
modules, however this should be soon fixed.
--
Tzafrir Cohen
icq#16849755
n the
> http://docs.tzafrir.org.il/man/pattest.8.html, there are no setting files
> and cablling for the two sides. please explan the settings and cabliing.
> thanks!
> Chris
Nothing special. E.g. you should be able to call with Asterisk from one
to the other.
--
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of them.
It is also used in the asterisk-gui.
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On Tue, Jul 21, 2009 at 12:01:18PM +0800, Chris YM wrote:
> hello:
> I wan to use the test tools-patgen and pattest for pri cards. according to
> Tzafrir Cohen from
> http://docs.tzafrir.org.il/man/pattest.8.html, i still does not know how to
> use that.
> do i need to connect
root directory of asterisk and open the
> file "makeopts"
> last line, insert *SPANDSP_LIB=-lspandsp
> - *after them, you can type make menuselct and expected entries could be
That's wrong. It indicates you passed the wrong values to the configure
script.
--
0/1)
>
>
> I again removed /usr/lib/libpri* , reinstalled libpri 1.4.1, recompiled
> asterisk
> and everything worked again.
>
> What might me the issue here I originally thought it was a dahdi
> issue but
> it appears to be a libpri issue.
--
On Sat, Jul 18, 2009 at 10:11:42AM +0100, Julian Lyndon-Smith wrote:
> What is interesting is that there is no mention of the software used -
> if it is asterisk, he would need to make the code available, no ?
Most likely.
--
Tzafrir Cohen
icq#16
wnloaded asterisk win 32
http://www.asteriskwin32.com/ ? Do yourself a favour and avoid that. It
is ancient (based on 1.2. IIRC). Do use some more decent
version of Asterisk. Either on a separate box or on a virtual machine.
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Tzafrir Cohen
icq#16849755 jabber:tzaf
"
What version of Asterisk is it? Any chance of using the built-in
chan_h323?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://
)
('/' instead of ',')
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
___
-- Bandwidth and Colocation
On Mon, Jul 06, 2009 at 10:00:12AM -0400, Jerry Geis wrote:
> After upgrading incoming calls seemed to work just fine.
> Outgoing calls gave me an error 99
What is the outout of:
cat /proc/dahdi/*
Can you provide a trace of such a failed call?
--
Tzafrir Cohen
icq#16
k/ ?
https://issues.asterisk.org/view.php?id=15119
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
___
--
mmy so that status reports OK
> instead of unconfigured.
There is no need to configure dahdi_dummy.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@loc
Portable .Net could be used), but
still requires some actual work. Python seems to be more popular among
the target audince and I still don't see such a pbx_python.
If you really want it, have someone write it :-)
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.
On Fri, Jul 03, 2009 at 12:30:07AM -0700, Trevor Hammonds wrote:
> From: Tzafrir Cohen
> Sent: Thursday, July 02, 2009 11:47 PM
>
> On Thu, Jul 02, 2009 at 09:02:05PM -0700, Trevor Hammonds wrote:
>
> >> "Anyhow, on the Blackberry, when you hold down the Alt
7;asterisk -r'
connections. Use either socat or OpenBSD netcat (both support writing to
a unix-domain socket). This should work reasonably well if you just want
to write some commands but are not interested in their output.
--
Tzafrir Cohen
icq#16849755 jabber:t
> don't understand why incoming calls work and outgoing calls don't.
>
> Dahdi is using a TDM400 with 4 FXO modules. The messages at boot time
> seem correct and "dahdi show channels" seems to indicate all three
> dahdi channels I use are present an
chance of answering the OR?
There is no DTMF tone associated with digits (over D). So they must be
using something else. Any idea what?
If not: could you please provide some recordings of sume such tones (in
a ulaw / alaw codec, if possible).
--
Tzafrir Cohen
icq#16849755
> just doesn't work).
>
> Anyhow, on the Blackberry, when you hold down the Alt key and press
> the alpha character, the device sends out the correct digit tone
> associated with that character, like on a regular phone keypad.
Is it a tone?
Or the letter itself in SIP /
wrong? Where are modules?
ls -l /usr/sbin/asterisk
Any change you enabled module embedding?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@
nceller?
>
> This is the contents of /etc/asterisk/chan_dahdi.conf
> http://pastebin.com/f7ef35e72
>
> This is /proc/interrupts
> http://pastebin.com/f61cd8398
>
> This is lsmod
> http://pastebin.com/m56105bf5
dahdi_echocan_mg2 is loaded .
However, what do you
rvice
> asterisk restart" solves the problem. But it again comes after few hours
> again.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xor
ve
a waltie-talkie and not a phone.
So while you can basically encode your data into an audio file, send it
to the other side and decode it there Asterisk (or just about any VoIP
software) will opt for timely delivery rather than a reliable delivery.
--
Tzafrir Cohen
icq#16849755
te.org/ - mixes editing config files and
storing configuration in a database. Much saner data model than
FreePBX.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com
On Thu, Jun 25, 2009 at 11:02:44AM +0200, jonas kellens wrote:
> Tzafrir Cohen,
>
> if mixing hand-written configs with GUI-configs is not 'good practise',
> then how to build a scalable Asterisk IP-PBX where the customer is not
> 100% dependent of the implementer ?
&g
d
> /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.
> o.cmd for
> /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o
>
>
> Anyone else seeing this?
My guess: a failed download? A partial download?
--
On Wed, Jun 24, 2009 at 05:41:34PM -0400, Steve Totaro wrote:
> On Wed, Jun 24, 2009 at 5:31 PM, Tzafrir Cohen
> wrote:
>
> > On Wed, Jun 24, 2009 at 05:01:04PM -0400, Steve Totaro wrote:
> >
> > > In FreePBX there are whatever_custom.conf files that are not touch
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