version and then re-installed
another Asterisk version without recompiling sngtc, you may get this issue.
*Moises Silva
**Manager, Software Engineering***
msi...@sangoma.com
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
t. +1 800 388 2475 (N. America)
t. +1
of wanpipemon -i interface -c astats -m
module
You should run that command when chan_dahdi is reporting the channel as
busy.
If you have not contacted techd...@sangoma.com I very much advice you to,
that way we can take care of you and get a clear history of your issues on
our tracker.
*Moises
though which uses chan_dahdi (patching needed at the moment).
*Moises Silva
**Manager, Software Engineering***
msi...@sangoma.com
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
t. +1 800 388 2475 (N. America)
t. +1 905 474 1990 x128
f. +1 905 474 9223
On Sat, May 26, 2012 at 2:55 AM, Mitul Limbani mi...@enterux.in wrote:
Dear Moises,
Does Sangoma manufacture 4 port gsm card? or is that an Openvox card?
As I said, Sangoma
http://www.sangoma.com/products/telecom_boards/wireless/w400.html
--
to specify
a different caller category.
*Moises Silva
**Software Engineer, Development Manager***
msi...@sangoma.com
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
t. +1 800 388 2475 (N. America)
t. +1 905 474 1990 x128
f. +1 905 474 9223
**http
asynchronous AGI application.
Despite being some shameless self-promotion, I want to point out this
post I wrote several years ago explaining the basics:
http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/
Moises Silva
Senior Software Engineer, Software Development Manager
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.
Regards,
Moises Silva
Senior Software Engineer, Software
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 | e. m...@sangoma.com
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 L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. m...@sangoma.com
On Sun, Feb 13, 2011 at 9:25 PM, Roi Stork roi.st...@gmail.com 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 Engineer
Sangoma Technologies
As the error suggest, try checking /var/log/messages for possible hints on
what went wrong.
Make sure you configured the device with wancfg_dahdi script first.
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474
2. ifconfig -a (from a working and non-working situation)
3. lspci -v and lsusb -v (from a working and non-working situation)
4. wanrouter hwprobe verbose (from a working and non-working situation)
5. /var/log/messages (near the date the problem happened)
Moises Silva
Senior Software Engineer
in 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
Sangoma
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
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100
.
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
--
_
-- Bandwidth and Colocation Provided by http://www.api
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 just fine
there.
Moises Silva
Senior
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
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 Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R
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 Software Engineer
Sangoma
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
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
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
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 x 128 | e. m
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
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
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
___
-- Bandwidth and Colocation Provided by http://www.api
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 9T3
Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
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 128 | e. m...@sangoma.com
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
Software Developer
Sangoma Technologies
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??
--
Moises Silva
Software Developer
Sangoma Technologies
.
--
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 Colocation Provided by http://www.api-digital.com --
AstriCon
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
Sangoma Technologies Inc. | 50 McIntosh Drive
, as the disclaimer 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
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 | e. m
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. 1 905 474 1990 x 128 | e. m...@sangoma.com
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 Technologies Inc. | 50 McIntosh
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. 1 905 474 1990 x 128 | e. m...@sangoma.com
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
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 Developer
Sangoma Technologies Inc. | 50 McIntosh
, 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 9T3
Canada
t. 1 905 474 1990 x 128
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 forward for people to make
informed opinions about their experience with both engines.
--
Moises Silva
Software Developer
Sangoma
some comprehensive FreeSWITCH docs. Heck, even SER has
more comprehensive documentation, and that's saying 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
On Thu, Aug 6, 2009 at 6:19 AM, Benny Amorsen
benny+use...@amorsen.dkbenny%2buse...@amorsen.dk
wrote:
Moises Silva moises.si...@gmail.com writes:
Just for the record, Sangoma Media Gateway does exactly that, leave all
your PSTN interfaces (BRI, SS7, PRI) in another box and communicates
://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 128 | e. m...@sangoma.com
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...@sangoma.com
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 and Colocation Provided by http://www.api
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. 1 905 474 1990 x 128 | e. m
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
this
background thread monitors its audio, on a redirect StopMixMonitor
thread should continue saving audio until StopMixMonitor is called.
--
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
-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 9T3 Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
a
link to it.
Also, what do you see when you do make menuselect - channel drivers
- chan_dahdi? if Asterisk see dahdi, then you should see it marked
for compilation.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
L3R 9T3 Canada
t. 1 905 474
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 executing EXEC MixMonitor inside a regular AGI script and then
redirect to MeetMe).
--
Moises Silva
Software
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
L3R 9T3 Canada
t. 1 905 474 1990 x 128 | e
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Moises Silva
Software Developer
Sangoma Technologies
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
Sangoma
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
L3R 9T3 Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
?
--
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 Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
to guess. So, first try this on the Asterisk
CLI:
stun set debug on
That should give you (and us) more information to troubleshoot why the
stun request failed (also enable debug and verbosity as usual).
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120
Hi, in MFC-R2 signaling there is a value Calling party category signal
(e.g., normal subscriber, high-priority subscriber, operator, coin-operated
telephone)
How can I get that information in my Asterisk??
That depends on which MFC-R2 solution are you using for Asterisk. The
2 most known are
2.2 is still a release candidate and NOT a final release,
the Wanpipe drivers were not tested with it.
Why don't you just use the latest DAHDI release (and not the release candidate)?
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
L3R 9T3
Most likely you screwed up and just compiled dahdi release but you
still have loaded the rc dahdi kernel modules.
cat /sys/module/dahdi/version
That will tell you for sure which dahdi module version is loaded.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive
Hi,
To further improve Asterisk documentation, would approve manager show
events and manager show event foo commands to be added to CLI ?
Today, it is possible to list available manager commands but not to list
available events, AFAIK.
Regards
The problem is that currently, manager events
plan like the 1.4.18 one and it worked fine.
I hope this can be useful.
Regards
Jose
-- Moises Silva wrote :
I really think you did not recompile and reinstall after applying the
new patch. I don't see any code path where the message
[Apr 13 12:03:57] DEBUG[2755]: res_agi.c:464
On Mon, Apr 13, 2009 at 6:59 AM, cyr2...@gmail.com wrote:
Hi Moy,
thanks a lot for your fix, but I'm afraid it doesn't work. I looked your
patch over and I realize the code never passes by neither of the two lines
you added with returnstatus = AGI_RESULT_HANGUP. Even, it seems the
Which Wanpipe version did you download?
2009/4/13 Giovanni Andrés Nopal Pascual giova...@voip.unam.mx:
Hi everybody!
I'm triying to install a Sangoma A200-R FXO card on a Debian Linux 5
(Lenny), 2.6.26 kernel.
To install wanpipe driver I type:
WANPIPE_FOLDER# ./Setup install
Duh, read the subject.
I suggest to try 3.3.16 beta, given that is probably a kernel version issue.
On Mon, Apr 13, 2009 at 7:15 PM, Moises Silva moises.si...@gmail.com wrote:
Which Wanpipe version did you download?
2009/4/13 Giovanni Andrés Nopal Pascual giova...@voip.unam.mx:
Hi everybody
Glad to hear it worked for you. I'd certainly like to take a look this
Monday and see why openr2 did not work for you.
Moy
On Fri, Apr 10, 2009 at 10:42 PM, Giovanny Magallanes
gmagalla...@gmail.com wrote:
Hi Moises and Steve,
I tried with all protocol variants for Openr2 (AR, BR, CN, CZ, CO,
there long ago). Which is the
protocol called CO in openr2?
Steve.
Moises Silva wrote:
Sounds like a protocol variant issue. Is the telco supposed to send you ANI?
You have 2 options, the first option is to try with the ITU variant,
if that does not work, set the option mfcr2_skip_category=yes
Sounds like a protocol variant issue. Is the telco supposed to send you ANI?
You have 2 options, the first option is to try with the ITU variant,
if that does not work, set the option mfcr2_skip_category=yes and see
if that helps.
Moy
On Wed, Apr 8, 2009 at 6:06 PM, Giovanny Magallanes
Released means no patching needed, it means it was tested and put
into Asterisk tree. So, I published a patch for 1.4 so it could be
used in 1.4 however the feature per se was just released for Asterisk
1.6.
Moy
On Tue, Apr 7, 2009 at 10:01 AM, cyr2...@gmail.com wrote:
Moy,
I apologize if you
It's a bug in the Async AGI feature. I have created a new patch
http://www.moythreads.com/asterisk-1.4.18-async-agi.patch
Please test it and let me know if it works for you,
Moy
On Tue, Apr 7, 2009 at 11:50 AM, Moises Silva moises.si...@gmail.com wrote:
Released means no patching needed
Moises Silva moises.si...@gmail.com
Async AGI was never released for Asterisk 1.4.X, so probably the patch
you used has a bug or something, do you still have the patch around?
Moy
On Thu, Apr 2, 2009 at 5:44 AM, cyr2...@gmail.com wrote:
Hi Henrik,
I would like to do the same thing you
Use dahdi_tool to see that.
On Fri, Apr 3, 2009 at 9:24 AM, criptos crip...@aullox.com wrote:
I'm using a Te201p card, with unichan, I want to know if my channels are
ready or in alarm... but uc show channel o uc show channels, doesn't show
me anything...
Any Ideas?
thanks.
Async AGI was never released for Asterisk 1.4.X, so probably the patch
you used has a bug or something, do you still have the patch around?
Moy
On Thu, Apr 2, 2009 at 5:44 AM, cyr2...@gmail.com wrote:
Hi Henrik,
I would like to do the same thing you are doing here. I want to implement an
Hello,
Which Asterisk version are you using? I was unable to reproduce your
problem with Asterisk 1.6.0.3, also please post details about your
dial plan extensions.
Moy
On Mon, Mar 30, 2009 at 7:13 AM, Jose Arias cyr2...@gmail.com wrote:
Hi,
I'm bringing this discussion here from
1) How can I use codec_dahdi? Would it be useful when passing a call from
one dahdi channel to another dahdi channel?
It is used whenever you need G729 or G723 transcoding (or any other
format supported by the Digium transcoding board). If you don't have a
Digium transcoding board then you don't
Is in the process of being merged.
http://bugs.digium.com/view.php?id=12509
http://reviewboard.digium.com/r/40/
http://www.libopenr2.org/
Moisés Silva
On Thu, Jan 15, 2009 at 9:44 AM, David fire ddf...@gmail.com wrote:
hi i am reading about new codecs and new stuff to be added to asterisk.
That's Digium's folks decision. It was said they wanted it for 1.6.3,
but, that's not for sure, as I said, they will decide.
On Thu, Jan 15, 2009 at 11:54 AM, David fire ddf...@gmail.com wrote:
thanks for the answer.
any idea in wich version it will be merged?
thanks
2009/1/15 Moises Silva
Hello Henrik,
I have not used Asterisk from a user perspective lately, but, when I
added the async agi functionality, I used to control this using a
manager redirect action to the same priority where the channel calls
async agi, that will work like a break that re-enters the async agi
loop .
There are probably other OpenR2 users that can help you in asterisk-r2
mailing list (http://lists.digium.com/pipermail/asterisk-r2/)
I have not tried Trixbox, but the first release of OpenR2 will be next
week, probably that will help to have Trixbox a proper support.
Moy
On Sat, Nov 29, 2008 at
Hello Peter,
You can ask this better in the asterisk-r2 mailing list.
I don't know of anyone that has used OpenR2 in Thailand, but I am
interested in adding support for that variant. Contact me at this same
e-mail address or via Google talk (my e-mail address works for MSN as
well ) to discuss
Hey Carlos,
What is the best method to debug DTMF issues? Do I have to sniff the
SIP packets?
The best method to debug DTMF issues depend on how you receive those
DTMF digits. Assuming you can use SIP INFO for the DTMF, that means
the DTMF digits are not really DTMF :-), that is, is
,
How i can do to join asterisk-r2 list ? My congratulations about your
article in digium blog http://blogs.digium.com/page/2/
I will collaborate in your project and give support from Venezuela.
Regards,
Luis Morales
On Sat, Sep 20, 2008 at 7:47 PM, Moises Silva [EMAIL PROTECTED] wrote:
Dae
] On Behalf Of Moises
Silva
Sent: Wednesday, September 17, 2008 10:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help with MFC/R2
It seems to me your lines are blocked.
Execute zttool and if you see 1101 in the rx bits, it means the telco
...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva
Sent: Wednesday, September 17, 2008 10:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help with MFC/R2
It seems to me your lines are blocked
The line is supposed be E1-MFC/R2 and works perfect with a Panasonic PBX,
actually
Exists any variant of MFC/R2? And how can I configure it to get working?
As I said, no matter which variant you try, the AB bits MUST be in 10
to be able to make calls with Unicall/libmfcr2. I have never seen
That means someone else has already open the zap device, most likely
Asterisk. Just one application at a given time can open a zap device.
You cannot run testcall and Asterisk at the same time unless you make
sure they don't try to open the same channels.
Moy
On Wed, Sep 17, 2008 at 1:27 AM, Dae
But after rerunning the test, I only get the first log (w/o Far end
replies.)
Any help will be really appreciated!
Thank you!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva
Sent: Wednesday, September 17, 2008 8:33 AM
To: Asterisk
You need to enable loglevel=255 in unicall.conf and enable all the
levels of logging in logger.conf, otherwise the logs you post don't
say much.
Moisés Silva
On Fri, Jun 6, 2008 at 2:58 PM, Mariano Borgognone
[EMAIL PROTECTED] wrote:
Dears,
I have problem ASTERISK with PSTN SIEMENS EWSD (MFC
The latest version of the driver included in
http://www.moythreads.com/astunicall/ comes with a change that will
set the variable UC_CATEGORY in your dialplan, Brasil has a special
category for those calls, don't remember the name that will show up,
but you can make a couple of tests and then drop
If you are an MFC/R2 user and want to help in the development of
chan_zap support for this signalling, please take a look at the
bugtracker at http://bugs.digium.com/view.php?id=12509 and/or contact
me. Currently just México support is built-in, if you want your
country variant supported, drop me
Hello Moisés, thanks for your effort on this! I would love to use Digium
cards for MFC/R2 signalling in the future.
Currently you can use Digium cards with Unicall :-) , tho, having
MFC/R2 on chan_zap is more handy.
I added some info you might like in the bugtracker, you might take a look
.
Moisés Silva
On Thu, Apr 24, 2008 at 8:26 AM, Ruben Zamora [EMAIL PROTECTED] wrote:
Moises
Thats means, that we arent going to use unicall?
If that true i can test these weekend with a E1-Axtel.
Thanks
Ruben
Moises Silva escribió:
If you are an MFC/R2 user and want to help
Way more handy and will be much more reliable too. Steve Underwood did a
great job implemeting it, but as far as I know the code isn't actively
maintained anymore. Of course your implementation of MFC/R2 will take a while
to become stable, but hey -- it's a start.
Agreed.
Russel pointed
Unicall MFC/R2 is activelly maintained. by Moy. Actually it's a backport of
the Steve driver (now coded for Callweaver derivative) to Asterisk (1.2, 1.4,
and 1.6 soon). It works pretty well. In fact, it works more stable in 1.4
than the original Steve driver in 1.2, and with better sound
The E1 use ALAW, if you want to avoid trans-coding use ALAW in your
phones as well. In any call you have 2 call legs, callee and caller,
try to isolate the problem and determine if the audio is really coming
that bad from the E1, you can use ztmonitor to hook into the E1 and
listen to the audio.
http://store.digium.com/productview.php?product_code=G729CODEC
http://www.digium.com/en/docs/G729/g729policy.php
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing
On Thu, Apr 17, 2008 at 11:14 AM, Carlos Chavez [EMAIL PROTECTED] wrote:
I need a refresher course on how many licenses
Hello Ivan,
I don't see nothing wrong in terms of signaling. When your side
(Asterisk/Unicall) request ANI, the other end answer with the signal
F, which means No More ANI, hence you receive an empty ANI string.
When your side request DNIS, the other end does not answer in several
seconds, which
On 3/6/08, Moises Silva [EMAIL PROTECTED] wrote:
What kind of problems are you talking about and what you want to modify?
On Thu, Mar 6, 2008 at 2:42 PM, Jessica Gonzalez Arriagada
[EMAIL PROTECTED] wrote:
Hi Asterisk-user, Steve;
I´m using libmfcr2-0.0.3.tar.gz
What kind of problems are you talking about and what you want to modify?
On Thu, Mar 6, 2008 at 2:42 PM, Jessica Gonzalez Arriagada
[EMAIL PROTECTED] wrote:
Hi Asterisk-user, Steve;
I´m using libmfcr2-0.0.3.tar.gz,
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