On 06-06-12 11:41, Thorsten Göllner wrote:
Where can I find such ip-lists, please?
http://www.ipdeny.com/
Regards,
Patrick
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the description or does it not have those features (yet)?
Regards,
Patrick
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Hi Khalid,
On 18-05-12 20:50, khalid touati wrote:
Hi Patrick,
it seems like you have the magic ball, I think what you described is
exactly what happened:
After we tested the server+ link and we were able to have simultaneous
calls (as expected), and knowing that this server was not touched
and tried other configuration files too but I always get faxes at 9600.
Did you restart iaxmodem and hylafax after you made those changes?
Regards,
Patrick
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-channel all the
time at their expense. And with their ISDN team not contactable it may
be a bit of a challenge to get them to enable the port again (after you
promised not to mess with heir D-channel again...).
Special thanks to Patrick! thanks and good luck to all!
My pleasure.
Regards
On 11-05-12 05:44, khalid touati wrote:
Thank you for your reply Patrick!
for the first situation, I did try asterisk 1.6.2.6 and dahdi 2.3 but
with no success.
Can anyone suggest a combination that works till a patch is released?
Unfortunately not and I don't have a Digium BRI card to test
and did one test call and that call worked
fine. Use at own risk :)
Regards,
Patrick
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On 10-05-12 23:47, Kevin P. Fleming wrote:
On 05/10/2012 03:20 PM, khalid touati wrote:
Thank you Patrick for the detailed info, it does make perfect sense to
me, I never expected that Digium cards have such an problem!
There are patches in the works already (being tested by users in Europe
Hi Khalid,
Judging from that bug report I *think*:
On 11-05-12 03:39, khalid touati wrote:
Patrick,
I got confused though is this true:
Any Asterisk soft+digium hdw = it doesn't work
There seem to be combinations that do work. It is my understanding from
that bugreport that an older libpri
using, what type of ISDN line
(PTP?), what's the DAHDI or Sangoma (or...) and Asterisk configuration,
what do the log files say, etc?
Have you tried calling the vendor of the ISDN card for support?
Regards,
Patrick
for step or
call Digium support.
http://docs.digium.com/H8/hx8_series_manual.pdf
Regards,
Patrick
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look at it and they said
BT equipements is not responding to the card within a certain range that
the card is looking for (i'm not sure what range but I do believe too
it's a BT issue), But I have run all the couple command that Patrick
suggested (to double check), tested again and still same kind
website (SugarForge?) and the last one can be found here:
https://github.com/blak3r/yaai
If you find something that works please let us know.
Regards,
Patrick
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for G.729 licensing.
Anybody with experience or quantitative measurements of the voice
quality degradation in that scenario?
The term that may interest you is Mean Opinion Score and iLBC is quite
good. See http://en.wikipedia.org/wiki/Mean_opinion_score
Regards,
Patrick
(that's on Red Hat. I have no idea
about Ubuntu).
Regards,
Patrick
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).
Regards,
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On 04/12/2012 06:11 AM, upendra wrote:
Hi,
thanks for reply , i want to know the 2.4.0 or 2.6.0 means what , how
they naming it , by the kernel version or its just a official release
number of digum...??
It's a Digium created release number.
Regards,
Patrick
experience on the list.
Regards,
Patrick
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On 03/29/2012 03:47 PM, Markus wrote:
I'm pretty sure there are a bunch of people who would be happy to pay
money for a better a2billing. Including myself :)
Have you looked at jbilling.com ? It's F/OSS with commercial support.
Regards,
Patrick
understand pox on sqlite3!. Please
elaborate.
Regards,
Patrick
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out the rescue cd and run memtest86+, check hardware etc.
Regards,
Patrick
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a quick test and it compiles fine
against 1.8.10. I have not seen a patch for 10. Iirc the 1.8 patch was
created by PrivateWave. Perhaps you can ask (hire) them for a 10 version
of the patch?
Regards,
Patrick
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tune in via a browser or streaming
through media player.
Perhaps it's possible to stream the conference with Icecast? In 1.8.9.3
there is an ices module which allows you to stream audio from Asterisk
to an Icecast server. Have you looked at that?
Regards,
Patrick
,
Patrick
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On 06-03-12 23:07, Karl Fife wrote:
Yep. That's what's happening. I'll file a bug.
AFAICT it's not a bug but the way RPM works.
Regards,
Patrick
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On 06-03-12 23:36, Jason Parker wrote:
On 03/06/2012 04:24 PM, Patrick Lists wrote:
On 06-03-12 23:07, Karl Fife wrote:
Yep. That's what's happening. I'll file a bug.
AFAICT it's not a bug but the way RPM works.
Regards,
Patrick
He didn't suggest that he was talking about RPMs
talk to the upstream provider? A quick whois on that IP address
suggests mentions PlusServer. Send them an email at ab...@plusserver.de
or use the chat link on the left side on this page:
http://www.plusserver.de/produkte/ or call them at +49-2233-6124300.
Regards,
Patrick
.
One of the VoIP providers I use is voip.ms which is in Canada. They can
port your and your client's numbers. Afaik they have a good reputation.
Why don't you give them a call. http://voip.ms/contactus.php
Regards,
Patrick (no affiliation with voip.ms, just a customer
2.6.32-37-server ?
Hire a kernel developer or ask Atcom to update their drivers.
Regards,
Patrick
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On 25-02-12 19:47, Jason Parker wrote:
yum and rpm do not support downgrades.
Incorrect. There is yum downgrade. See man yum.
Regards,
Patrick
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New
nice addition. Perhaps someone with Asterisk
coding skills can backport the patch. Have you checked if it applies at
all to the latest 1.8 master? I wonder if that patch is already part of
10 master or the 10.2 branch as I could not see anything mentioned on
reviewboard.
Regards,
Patrick
still buy those. At least I have not seen those in .nl in a
long time. Maybe anything from eBay that works?
Regards,
Patrick
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and sound quality good.
Regards,
Patrick
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be
an idea to use the message in the kernel log wct4xxp :02:08.0:
Interrupts not detected. also in the error message from dahdi_cfg (at
least the interrupts not detected part)? That would give a clue what's
going on without having to dig through logfiles.
Just my 0.02.
Regards,
Patrick
in logfiles it is.
Regards,
Patrick
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asterisk
in thinking that?
Yes.
Regards,
Patrick
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- requires PCI-X or at least PCI 2.1).
If that is an Eicon Diva *Server* card then it should work fine with the
drivers and chan_capi from http://www.melware.de
Regards,
Patrick
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) lines on a SIP trunk which
support multiple simultaneous calls on that single trunk?
Regards,
Patrick
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and Diva drivers from Melware or
Dialogic.
http://www.dialogic.com/products/media/diva.aspx
http://www.melware.de/en/download.html
Regards,
Patrick
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New
it
to 3.3.4. Hopefully it will not happen again.
Regards,
Patrick
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ulaw to g729 for a normal mobile-to-mobile call?
Afaik mobile to mobile uses the gsm codec (in gsm country, no idea about
cdma etc.). IMHO gsm is not bad but certainly not as good as ulaw/alaw.
Read up on MOS:
https://en.wikipedia.org/wiki/Mean_opinion_score
Regards,
Patrick
have changed with recent versions I think you still need
DAHDI if you want to use MeetMe and maybe other modules that require
proper timing (which DAHDI provides).
Regards,
Patrick
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be able to get support so just call
Digium. Good luck. Look forward to hear how it goes so please update the
list.
Regards,
Patrick
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asterisk-users
:
- B2BUA (Asterisk, FreeSWITCH, Trixbox)
- SIP Proxy (SER, OpenSER).
I'll leave figuring out where SipX and YATE fit in to you.
Regards,
Patrick
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as
IAX only needs one port. The question is how are you going to SSH into
the box if you use the SSH port for Asterisk?
Regards,
Patrick
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for 'HOST' - ACL error
(permit/deny)
NOTICE.* .*: Failed to authenticate user .*@HOST.*
How about those (no idea for which Asterisk version they are)?
Regards,
Patrick
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on Rails server). Can anybody tell me
what CSTele might be?
No idea.
Good luck!
Regards,
Patrick
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:
http://wiki.openvox.cn/index.php/OpenVox_B400P_User_Manual_for_dahdi
Please note that in the instructions they use older versions. I would
use the latest DAHDI, libpri (don't forget this one) and asterisk 1.8
available here: https://www.asterisk.org/downloads
Regards,
Patrick
On 29-11-11 19:54, bilal ghayyad wrote:
Can u help me to determine this for Cisco IP Phones model 7942G with SIP image
and Polycom IP Phones?
Why don't you just download the admin manuals of these phones and look
it up?
Regards,
Patrick
On 11/22/2011 08:14 AM, Jai Rangi wrote:
[removed commercial offer]
You posted to the wrong list. The correct list for commercial business
related discussion is asterisk-biz. Please do not spam the
asterisk-users list again with your commercial offers.
Regards,
Patrick
On 11/15/2011 02:45 AM, jordan pan wrote:
Recently，I met a very strange phenomenon。I found that my asterisk bin
file had changed when running。I checked a lot of machines ， and the
result is almost all of the bin files have taken place。
[snip]
Maybe prelink? See man prelink.
Regards,
Patrick
Fedora 10. If you still use F10, you do know that Fedora 10 is End of
Line for a long time and has not received security updates in a long
time and by now has more security holes than Swiss cheese?
Regards,
Patrick
of asking Digium?
Patrick
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for Notify User 4701
-- Executing [912066604@sipphones:2] Dial(SIP/4773-0003e920,
SIP/att/xxx,80) in new stack
I still only see one Set and one Dial.
Regards,
Patrick
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On 10/21/2011 04:33 AM, Luke Hamburg wrote:
Patrick:
would you mind sharing how you're using the menuselect.makeopts exactly? I
first had this problem with 1.8.7.0 and again with 1.8.8.0-rc1. I've since
updated to 1.8.8.0-rc2 but for some reason I am still unable to get
menuselect to use
On 10/19/2011 04:52 AM, Luke Hamburg wrote:
I think this might actually be a bug.
https://issues.asterisk.org/jira/browse/ASTERISK-18137
Thank you very much for pointing that out Luke. Seems I bumped into the
same bug.
Regards,
Patrick
for 1.8.8.0-rc2 which will be
released this morning.
Thanks Jason. I'll update when it is available.
Regards,
Patrick
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released this morning.
My problem is indeed fixed in 1.8.8.0-rc2. Thanks!
Regards,
Patrick
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$ make
All requirements are met as building with make menuselect and selecting
the options from menuselect.makeopts works fine. Anyone have any ideas?
Thanks!
Patrick
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On 10/13/2011 06:21 PM, Mike Diehl wrote:
I'm thinking it's a firewall/NAT timeout issue. Has anyone seen this? Has
anyone fixed it? Any ideas, otherwise?
Did you try turning off the SIP ALG on the Sonicwall?
Regards,
Patrick
Hi,
Does anyone know which exact version of mISDN is required for chan_misdn
in 1.8 10?
TIA,
Patrick
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and use
it with chan_capi (must be a Server card, regular Diva cards don't
work). I use that in one Asterisk box and it's very reliable.
Groet,
Patrick
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nice to know if things may fall apart :)
Regards,
Patrick
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. Alternatively
perhaps you could try to power the phone through an NT1?
Would you mind sharing which version of DAHDI you used and where you got
the HFC-S patch?
Regards,
Patrick
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On 08/30/2011 08:37 PM, Tamer Higazi wrote:
Hi Patrick!
Now i got it.
I am using Gentoo Linux, Asterisk 1.8.5 and Dahdi 2.4.1.
The patches are automatically integrated at Gentoo. I didn't have to
patch anything. That did the community.
Thanks for the info.
Another question, I really
) kernel. Not sure if I understand you. Are you
saying that Digium does not test their software on RHEL6/CentOS6 or that
you tested it on RHEL6/CentOS6 but just not against the stock kernel?
Thanks,
Patrick
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On 08/15/2011 06:04 AM, Shaun Ruffell wrote:
On Mon, Aug 15, 2011 at 05:21:36AM +0200, Patrick Lists wrote:
On 08/14/2011 10:31 PM, Shaun Ruffell wrote:
[snip]
While I can't say I've run against that particular CENTOS kernel version,
I would be very surprised (and interested to know) if you
and it
has worked flawlessly. From a laptop in the field I can send a fax (via
a VPN) straight from OpenOffice and receive faxes via email.
Regards,
Patrick
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.
Regards,
Patrick
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,
Patrick
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Microsoft Embrace Extent? Like e.g. Sonus and Cisco do with their
interpretation of SIP.
Regards,
Patrick
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the announcement.
Frankly I could not care less about version numbers. Instead I prefer
stability, speedy bug fixes and new cool features. But 2.0.0 has my vote.
Regards,
Patrick
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,
Patrick
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. But 900 g.729 - g.711 sessions on
a single V-600 card? Or 200 T.38 sessions?
Regards,
Patrick
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easy to spot why the
patch failed (some code was added). Line 5399 is removed and the
addition comes before line 5417:
http://svnview.digium.com/svn/asterisk/branches/1.8/channels/sig_pri.c?view=markup
Why don't you try to apply it manually?
Regards,
Patrick
On 07/08/2011 08:46 PM, Michael L. Young wrote:
Patrick,
The patch was merged in to the 1.8 branch on 5/13/2011 as revision 318783
(http://svnview.digium.com/svn/asterisk?revision=318783view=revision).
1.8.5-rc1 was tagged on 06/29/2011
(http://svnview.digium.com/svn/asterisk/tags/1.8.5-rc1
On 07/09/2011 01:28 AM, Doug Lytle wrote:
Can you say a Virtualized Asterisk with a PRI card!
http://www.phoronix.com/scan.php?page=news_itempx=OTY0OQ
With virtualized environments prone to timing issues does this make
sense at all?
Regards,
Patrick
license. Seems you just need to ship the license with the binaries and
all should be well. Please also note that the IP Rights Grant seems to
note some limitations that you may want to study before shipping to your
customers.
Regards,
Patrick
on irc.freenode.net and ask Qwell (aka Jason Parker)
about this.
Regards,
Patrick
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stick) and install (from the USB stick) all the
updates that are available since the release of 5.6.
Regards,
Patrick
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or Audiocodes. Both solid solutions although
Audiocodes has a bit of a reputation when it comes to configuring it.
Regards,
Patrick
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On 05/24/2011 09:56 AM, Kristijan Vrban wrote:
http://code.google.com/p/zaphfc/
Thanks! For the archives I found another site where the zaphfc code from
that google site is integrated into the DAHDI source:
http://sourceforge.net/projects/dahdi-zaphfc/
Regards,
Patrick
, Junghanns duoBRI quadBRI octoBRI and old beroNet BxN cards,
are supposedly supported out of the box by the latest libpri dahdi.
Also check out this site: http://sourceforge.net/projects/dahdi-zaphfc/
Regards,
Patrick
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I look forward to seeing your repo updated with your latest changes.
Thanks,
Patrick
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On 05/24/2011 03:43 PM, Steve Davies wrote:
[snip]
Repo updated. I have tried to merge all of the other changes that are
out there also.
Thanks Steve.
Regards,
Patrick
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On 05/25/2011 01:20 AM, bilal ghayyad wrote:
Hi All;
To enable the compilation for the addon that is coming with Asterisk 1.8 when
doing compilation for the Asterisk, what should I do?
make menuselect and enable what you would like to be build.
Regards,
Patrick
module stopping while Asterisk is DoS'ed ?
Best regards,
Patrick
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: no
Capabilities: -1
Digit Begin: yes
Digit End: yes
Send HTML : no
Image Support: no
Text Support: no
Regards,
Patrick
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:-)
Regards,
Patrick
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On 02/17/2011 02:26 PM, ERIC HERRON wrote:
Yeah it’s the same thing; I think.
I think we have different config files…are you using the split?
Unfortunately I have no idea what the split means. Can you please explain?
Regards,
Patrick
,
Patrick
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like OpenSIPS or Kamailio.
Regards,
Patrick
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...
Have you tried an older or newer release?
Regards,
Patrick
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. However the scrolling caller id for incoming
calls make this minor annoyance worth the upgrade.
That's good to know. Thanks for the info.
Regards,
Patrick
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there's no way to change its behavior.
Regards,
Patrick
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(if configured).
Hope this helps.
Regards,
Patrick
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.
Patrick
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se.pat.misc.messageWaiting.inst.3.value=0
/se.pat.misc.messageWaiting.inst
/se.pat.misc.messageWaiting
/se.pat.misc
/se.pat
/se
Regards,
Patrick
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