On Sun, Feb 13, 2011 at 9:25 PM, Roi Stork wrote:
> Here's the messages log. There's a line that says ERROR: Unsupported DS E1
> CHIP (00:00)
>
>
That's pretty bad. Could you post the output of "wanrouter hwprobe verbose"
?
Moises Silva
Senior Software Eng
le to synchronize their clocks.
If you were using 2 Sangoma ports you can sync the ports with the
TE_REF_CLOCK parameter.
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
>
1.4.12 is just a newer version than 1.4.11 and any released version is as
production-ready as can be reasonably be expected AFAIK.
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128
i could not see any shared interrupts for Snagoma card.
>
> Anyhelp would be highly appreciated.
> --
>
Hello M Shokuie,
This kind of troubleshooting is better addressed by Sangoma technical
support staff. You can send an email to techd...@sangoma.com and you will be
taken care of.
Reg
laimer says in the
web page, you still need to pay royalty fees to the g729 patent holders
somehow. Unless you live in a country where patents do not matter.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 199
phone already expects the audio for the call in
the format negotiated during call setup which may or may not be g729. Not
sure if a re-invite could be issued to change the codec type in the middle
of the call, but I suppose it should be possible to implement.
--
Moises Silva
Software Developer
risk.org and read the
guidelines before submitting the bug report.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
___
-- Bandwidth and Co
* version is 1.6.1.4
>
>
You mean you cannot see AsyncAGI events? did you enable "agi" in the read=
parameter in manager.conf for your Java application user?
Can you send AGI commands to the channel through the manager? or through the
Asterisk CLI "agi exec" cmd??
--
Moi
added to the default manager.conf ?? ;)
>
I agree, will get that added to the manager.conf sample
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 12
solved in asterisk or it is impossible task?
> I remember having problems with DTMF ever since ver. 1.2 :-/
>
Your question is too general. I don't remember ever having a DTMF problem
since 1.0, so it depends on the use you give to asterisk and the equipment
you use.
--
Moises Silva
Softw
d Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Moises Silva
Software Developer
Sangoma
reproduce it with latest 1.4
version, if you can reproduce it the bug must be filled in
issues.asterisk.org.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
__
problem is not at R2 but in your local trunk settings (may be dialing in the
wrong group or something).
--
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
_
s still into the AGI loop, after having sent it the AGI EXE
>> MixMonitor action) the MixMonitor AGI action is stopped automatically and
>> the recording ends.
>>
>> Therefore, does anyone know how to manage that an AsyncAGI action to
>> remain running in background even
and we get stumped.
You meant that this does not happen with the OpenVox card, didn't you?
otherwise, you lost me.
If you can easily reproduce this, I'd be interested in look into it.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
gards
> Jose
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Moises Sil
to think beforehand on this matter.
So please confirm this. If you get an incoming call and send it to
Playback(demo-congrats) and then receive a second call and send it to
Playback(tt-monkeys), both callers will listen both demo-congrats and
tt-monkeys sounds?
--
Moises Silva
Software Developer
S
o you mean with "so what?", if you have not been involved in the
conversation you would not understand.
http://lists.digium.com/pipermail/asterisk-users/2009-June/232995.html
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
L3
of where the channel
is (regardless of whether the command was executed in Async AGI or
dial plan or whatever). However you are also using an old asterisk
version and is not likely you can report a bug unless you upgrade to
the latest Asterisk and reproduce without a patched Asterisk (for
example
21 WRTDM/0/20
> 22 WRTDM/0/21
> 23 WRTDM/0/22
> 24 WRTDM/0/23
>
> I does not matter if i pass --with-dahdi to ./configure script or not.
>
After running ./configure you will get an output file named
config.log, that file has the details about which te
/www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
L3R
xMonitor does is to launch a
background thread that hooks into the channel audio, then the channel
continues to execute other applications in the dial plan while this
background thread monitors its audio, on a redirect StopMixMonitor
thread should continue saving audio until StopMixMonitor is call
is enough in order to get the required timing. The only reason to
increase the kernel timer to 1khz is when you need dahdi_dummy module, which
uses this timer to fake interrupts that otherwise would be generated by real
hardware.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 Mc
e RTP. Are you
suggesting to change the protocol to support such transfers?
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
___
-- Bandwidth an
s he want to do it ? Share secret / illegal files LOL ?
>
> Martin
>
I would think IAX ack just the signaling frames, not every single audio
frame, does it?
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 4
eway along with Asterisk (
http://wiki.sangoma.com/sangoma-wanpipe-smg-asterisk-bri-installation)
That is known to work pretty well for lots of people.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 1
interfaces (BRI, SS7, PRI) in another box and communicates with
Asterisk through the Woomera protocol.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. m
On Thu, Aug 6, 2009 at 6:19 AM, Benny Amorsen
> wrote:
> Moises Silva writes:
>
> > Just for the record, Sangoma Media Gateway does exactly that, leave all
> > your PSTN interfaces (BRI, SS7, PRI) in another box and communicates with
> > Asterisk through the Woomer
s. FreeSWITCH needs to catch
up with documentation, but I would defy anyone to say they've come to hang
around on IRC and did not get their question answered.
Both Asterisk and FreeSWITCH share features, pro's, cont's and for some
people one is better than the other. I am looking
a LOT.
>
No offense taken ;-), hope is the same for you. Again, our perspective of
what "comprehensive" documentation is differs, needs improvement for sure
though.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
n't voilate the GPL either.
>
> It's nice to have competition. Keeps you on your toes.
>
> Gordon
>
Because Digium OWNS the Asterisk code, and they make an exception for their
binary code, is their right as owners (copyright holders) of the code.
--
Moises Silva
Software D
t;. For the casual
reader, the clarification means developers contributing to Asterisk still
own the code, but the disclaimer signed by them gives Digium enough rights
to make the exception.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
s.
In both modes the drivers notify on-hook, off-hook events depending on the
battery status.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
___
le enough that I just did the quick fix and this little feature should
be available in the next wanpipe release within this week.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 4
r in order to get this feature
into Asterisk soon :-)
Also don't hesitate in asking for help with the configuration.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 4
>
>
> Is your code vendor locked to Sangoma ???
>
>
Hello Martin, not at all. The code is intended to be part of chan_dahdi
Asterisk channel driver and as such any card capable of using the dahdi
interface can benefit from it.
--
Moises Silva
Software Developer
Sangoma Technol
igium boards in
high impedance mode. It seems the feature may not be exported via
configuration files yet, so changes to the driver may be needed?
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 |
If you want to open a bug report the proper place to do it is at
http://issues.asterisk.org/
Compile with DEBUG_THREADS and DETECT_DEADLOCKS (see make menuselect
compiler flags).
--
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
the legacy pbx will place a secondary call via ISDN ( did he mean
PRI? ) therefore Asterisk will just Record(), what is it that is not so
simple about that?
--
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990
http://downloads.asterisk.org/pub/telephony/libiax/
That package is outdated AFAIK but is a start. You should be able to use
chan_iax in Asterisk as a reference to fix libiax and use it for your own
purposes.
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive
blocked for your country ) or the line is configured only for
incoming calls ( not possible since chan_unicall.c hard-codes that parameter
to allow calls in both ways ).
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905
Try disabling SELinux if you have it enabled (unless of course you need it).
I seem to remember there is certain compilation flags required (position
independent code, -fPIC?) to run with SELinux enabled, may be the
codec_g729a.so is not compiled properly to run under such circumstances?
Moises
always
> the same problem
>
> Thanks!
>
I'd like to know which problem you had with the Sangoma card as there are no
shared interrupt issues we know of.
There used to be a problem with some Dell servers though, but that was
already fixed some weeks ago.
Moises Silva
Senior Softwa
rovide better
results, but in no way that means that we will not look at your issue with
the card that does not have HWEC.
A senior tech support engineer will be contacting you soon today to follow
up on your issue appropriately.
Regards,
Moises Silva
Senior Software Engineer
Sangoma Technologies
bridging, meaning the only used applications are Answer() and Dial() with
the DAHDI and SIP channel drivers, typically with latest 1.4.
Additionally we always compile DAHDI modifying the chunk size to reduce the
interrupt load.
As far as your question about PCIe 2.0, yes the A108 should work ju
Those modifications are done via regular Sangoma installation with a special
option to the Setup script.
http://wiki.sangoma.com/wanpipe-linux-asterisk-appendix#zaptel_adjustable_chunk_sz
http://www.sangoma.com/assets/docs/misc/2009_10_09_How_to_Reduce_Asterisk_System_Loads.pdf
Moises Silva
o ulaw/alaw or something that is supported natively by
Asterisk without extra licensing required.
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. m...@sangoma.com
--
ebug message and there is nothing to worry
about and you insist in believing this is a problem.
If you want to know what the message means and why you should not worry you
must understand what a lock is, what lock contention is and what a deadlock
is.
Moises Silva
Senior Software Engineer
San
this tool. Which unit is used to measure the signal level?
>
>
dahdi_monitor uses the sample values in L16 format.
They are in orders of magnitud of G.711. See tables 5 and 6 of the G.711
spec. In the end, the reference value is the dBm (google that).
Moises Silva
Senior Software Engineer
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