If you need something quick you could create a batch script with sed or awk
to remove the log lines you want and attach it to the prerotate script of
logrotate (in case you use any of these in your env). Certainly this is not
a final solution but it is already something that doesn't depend on an
In case anyone out there is working with CentOS, you might reconsider that
decision: https://blog.centos.org/2020/12/future-is-centos-stream/
On Mon, 11 May 2020 at 23:53, George Joseph wrote:
>
>
> On Sun, May 3, 2020 at 6:07 PM Patrick Wakano wrote:
>
>> Hello George,
>
inphone just press the hold once and then press and
hold the spacebar which will repeatedly do the hold and unhold several
times). After Wait is finished the "Exceptionally long queue length
queuing" will show up but gets resolved very fast just because I think
there weren't enough frames qu
ying but does not to cause
issues:
/usr/include/features.h:381:4: warning: #warning _FORTIFY_SOURCE requires
compiling with optimization (-O) [-Wcpp].
Anyway, these problems do not happen if you manually build with the simple
configure and make commands.
Cheers,
Patrick Wakano
On Fri, 18 Oct
That makes sense Kevin!
Thanks for the explanation, I will create a ticket for this then!
Kinds regards,
Patrick Wakano
On Wed, 26 Feb 2020 at 09:33, Kevin Harwell wrote:
> On Tue, Feb 25, 2020 at 4:02 PM Patrick Wakano wrote:
>
>> Hi Kevin!
>> Thanks very much fo
and possibly removed from the configure/makefile stuff for future
releases?
Kind regards,
Patrick Wakano
On Wed, 26 Feb 2020 at 06:33, Kevin Harwell wrote:
> On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano wrote:
>
>> Hello list,
>> Hope you are all doing well!
>>
>> I am f
sl is used?
I could not find a clear explanation for this problem and how to fix it
Any idea is much appreciated!
Thank you,
Kind regards,
Patrick Wakano
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if the issue persists? Alternatively create your own CentOS 7
VM from the official CentOS 7 repositories using kickstart and try that on
azure.
Best, Patrick
[1]
https://azuremarketplace.microsoft.com/en-us/marketplace/apps/RogueWave.
connected via AMI and I want it logging
all the actions (and the details of these actions) it was asked to do.
Any idea is much appreciated!
Thanks,
Kind regards,
Patrick Wakano
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I've never worked with fabric nor pysipp, but if you want to run sipp in
background so you can log out from your remote server and leave the test
running, I suggest you can use screen. It works well for me.
Patrick Wakano
On Sat, 20 Oct. 2018, 01:55 Olivier, wrote:
> Hello,
>
> I'm
itself should be able to timeout a long running script and return to the
dialplan. However looks like there is nothing of this sort.
Kind regards,
Patrick Wakano
On Sat, 15 Sep 2018 at 03:56, Eric Wieling wrote:
> I don't know AGIspeedy, but I have some PHP scripts where I set a
> c
the dialplan to get stuck due to some external script
problem that is actually outside of Asterisk control. Does Asterisk provide
some control of this sort? I searched around and could not find any.
Any idea is appreciated!
Kind regards
Patrick Wakano
behaviour? I could not find anything
related
In my opinion, Asterisk should at fail the Dial and proceed with whatever
was configured in the dialplan I tried some other 4XX SIP codes, but
the only one I found not behaving properly is the 400 one
Thanks,
Kind regards,
Patrick Wakano
By the way, bear in mind this is exactly what a blind transfer from B to C
would do, but with a lot of more work
On Mon, 9 Jul. 2018, 16:36 Patrick Wakano, wrote:
> Not sure how elegant this is, but I think you can try to elaborate some
> logic that when phone C dials something, it
Not sure how elegant this is, but I think you can try to elaborate some
logic that when phone C dials something, it would retrieve you the channel
phone A is connected and use the Bridge application to force the connection
of phone C to phone A. So you need first to save the channels you have
opened by some other MixMonitor thread. Or is there any
reason/situation in which this is not desired?
Kind regards,
Cheers,
Patrick Wakano
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Check
Thanks for the reply Joshua!
I did look the code, but it's too complicated for my old C knowledge :(
So I guess I am left with the Monitor app Also a ConfBridge would also
work instead of ChanSpy with whisper
Cheers,
Patrick Wakano
On 6 July 2018 at 08:51, Joshua Colp wrote:
> On
it myself) Anyway why MixMonitor can't?
Also does anyone have an idea on how to record everything in the same file?
Thanks,
Kind regards,
Patrick Wakano
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ld be done? If not automatically maybe with some new flag
Thank you,
Kind regards,
Patrick Wakano
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Check out the new Asterisk community forum at: https://community.asterisk
I agree! I have my SBC and asterisk servers all configured with rfc2833, so
it should be ok! No need for auto mode!
Thanks again!
Cheers
Patrick
On Tue, 1 May 2018, 20:07 Joshua Colp, <jc...@digium.com> wrote:
> On Tue, May 1, 2018, at 6:52 AM, Patrick Wakano wrote:
> > T
Thanks very much for the reply Joshua!
So I guess that setting dtmfmode=auto would be the safest choice in order
to strip out the DTMFs from the recording, right?
Cheers!
Patrick Wakano
On Tue, 1 May 2018, 19:36 Joshua Colp, <jc...@digium.com> wrote:
> On Mon, Apr 30, 2018, at 11:23 PM
he RFC2833
events never show up in the recording, but I just want to confirm that this
is always true.
Thanks,
Kind regards,
Patrick Wakano
<http://www.avg.com/email-signature?utm_medium=email_source=link_campaign=sig-email_content=webmail>
Virus-free.
www.avg.com
<http://www.avg.com/email-s
Just did Tzafrir suggestion and it worked like a charm!
Thanks very much!
Cheers,
Patrick Wakano
<http://www.avg.com/email-signature?utm_medium=email_source=link_campaign=sig-email_content=webmail>
Virus-free.
www.avg.com
<http://www.avg.com/email-signature?utm_medium=email_source=link
That's quite interesting Tzafrir!
I will give it a try!
Thank you very much for the idea!
Cheers,
Patrick Wakano
On 23 April 2018 at 23:42, Tzafrir Cohen <tzafrir.co...@xorcom.com> wrote:
> Also,
>
> On Mon, Apr 23, 2018 at 04:08:58PM +1000, Patrick Wakano wrote:
> > Hel
Hello Richard!
Thanks very much for your answer! It all makes sense.
I will just duplicated the tones config then!
Thanks again!
Cheers!
Patrick Wakano
On 23 April 2018 at 23:17, Richard Mudgett <rmudg...@digium.com> wrote:
> It looks like any support for "alias" as a tone
lplan execution causes this:
ERROR[20778][C-0006]: func_channel.c:661 func_channel_write_real:
Unknown country code '*gb*' for tonezone. Check indications.conf for
available country codes.
Any info is much appreciated!
Cheers,
Patrick Wakano
<http://www.avg.com/email-signature?utm_medium=e
That's really good info Tony!
Thanks very much for the response!
I will consider this to implement a better approach for the failed cases!
Cheers,
Patrick Wakano
On 14 March 2018 at 20:44, Tony Mountifield <t...@softins.co.uk> wrote:
> In article <CAPu3kNV8w+bYQT0W+QbnTSby0V5gfjLqZ
://wiki.asterisk.org/wiki/display/AST/Hangup+Cause) but I didn't find
someone actually stating it is a better alternative or replacement to the
DIALSTATUS or something similar.
Cheers,
Patrick Wakano
On 14 March 2018 at 13:30, Dovid Bender <do...@telecurve.com> wrote:
> I would think that is a bug since
knows exactly where is more suitable to use one over the
other?
Thanks,
Kind regards,
Patrick Wakano
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Thanks for you answer Marcus,
So maybe this means some bug was fixed? Anyone aware of something related?
>From the release notes, I couldn't find any direct change that could fix
this
Thanks,
Kind regards,
Patrick Wakano
On 21 February 2018 at 20:29, Marcus Kvarsell <marcus
times per month). Does anyone
have any idea of what may be happening/wrong?
I am running Asterisk version 13.6.0 on CentOS 6.6 kernel
2.6.32-504.el6.x86_64 and on CentOS 6.9 kernel 2.6.32-696.el6.x86_64.
Any idea is very appreciated!
Best regards,
Patrick Wakano
<http://www.avg.com/em
Apparently it is an
internal only problem, not actually outputting abnormal traffic in the
network
Best regards!
Patrick Wakano
On 16 November 2017 at 02:34, Matthew Jordan <mjor...@digium.com> wrote:
>
>
> On Mon, Nov 13, 2017 at 11:42 PM, Patrick Wakano <pwak...@gmail.com>
so
it seems that a bug exists somewhere leading to this problem.
So anyone has any idea of what could be happening or if it may be related
to some known bug?
We are running Asterisk 13.6.0 in CentOS 6.6.
Thanks very much
DROP
You can also look at xtables with geoip to drop countries (per
destination port) that should not connect to your Asterisk box. It's a
big hammer but it works really well.
Or put a proxy like Kamailio or OpenSIPS in front of the Asterisk box.
That's what the telco's/service provider
Thanks very much for the explanation Richard!!
Best Regards!
Patrick
<http://www.avg.com/email-signature?utm_medium=email_source=link_campaign=sig-email_content=webmail>
Virus-free.
www.avg.com
<http://www.avg.com/email-signature?utm_medium=email_source=link_campaign=sig-email_conten
e surrogate channels,
and what is expected of them when it comes to dialplan execution?
Cheers!
Patrick
<http://www.avg.com/email-signature?utm_medium=email_source=link_campaign=sig-email_content=webmail>
Virus-free.
www.avg.com
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Hello list,
We've got an Asterisk crash in one of our servers and the core dump showed
following call tree.
Is this anyhow helpful to someone? Seems like a regular RTP / RTCP handling
that lead to a malloc crash
Grateful for any help!
Cheers,
Patrick
Thread 1 (Thread 0x7f8d6b023700 (LWP
pjsip support:
https://github.com/fail2ban/fail2ban/commit/f85fb45b29768f687546ba25f805977cf00b6e43
HTH,
Patrick
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What an excellent response Richard!!! Thank you very much for that!!
Best regards!
Patrick
On Wed, Feb 15, 2017 at 5:23 AM, Richard Mudgett <rmudg...@digium.com>
wrote:
>
>
> On Tue, Feb 14, 2017 at 6:24 AM, Patrick Wakano <pwak...@gmail.com> wrote:
>
>> Hello Aste
e this?
Many thanks,
Cheers,
Patrick
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Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Sta
Keepalived + heartbeatd allows you to maintain a a floating IP between two
machines. If those two machines had configs, internal state synced, and the
IP is configured to float automatically between the two based on which is
actively up, would it be possible to not drop a call should the active
determined IP
address.
Thank you for your feedback Joshua. Does "right now" mean that this will
be fixed in the (near) future? Should I file a Jira ticket?
Thanks,
Patrick
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config_text_file_load:
Unterminated comment detected beginning on line 386
That needs fixing.
HTH,
Patrick
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New to Asterisk? Join us for a live
appened?
Thank in advanced.
Just a guess but try setting "pooling" to yes and "limit" to a higher value.
Best,
Patrick
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New to Asteri
al can be found at:
http://www.manualslib.com/manual/909341/Thomson-St2020.html?page=5
To download click the green Download button at the top.
In the right column there is also a link to the User Guide.
Cheers,
Patrick
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then be prepared for some fixing before it works with Asterisk
SIP on port 5060:
http://blog.laimbock.com/2014/03/27/how-to-make-asterisk-work-behind-fritz-box/
HTH,
Patrick
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with the GXP2140? Is it a reliable
phone? Does anyone have recommendations for other phones with gigabit pass
through?
Regards,
Patrick.
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New to Asterisk
I have seen a similar problem occasionally. We will be doing maintenance on a
customer's server and they will have one or two ghost channels on their
machine hundreds of hours old but with no call associated with them. So far we
have just been rebooting their server or issuing a hangup command
Thanks for the suggestions guys. I’ll try to have a play with Voipmonitor
in the near future.
So can I assume from the lack of discussion nobody is using the “sip show
channelstats” stuff?
Regards,
Patrick.
On 31/03/2015 08:23, Olivier oza.4...@gmail.com wrote:
Some SIP hardphones (Polycom
broadband
connection?
Regards,
Patrick.
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I encountered a bug in some Polycom models where it would refuse to register to
a domain that started with a number (e.g 3something.voip.com). Could that be
applicable here?
Regards,
Patrick.
From: Jordan Cook - Gyron Networks
jordan.c...@gyron.netmailto:jordan.c...@gyron.net
Reply
using a STUN server) and disable it.
Regards,
Patrick.
From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of Sonny Rajagopalan
sonny.rajagopa...@gmail.com
Sent: 09 January 2015 01:03
To: Asterisk Users Mailing List
I believe the Unmonitored status is linked to the qualify setting for each
user. If they aren't set to qualify=yes then it won't check their status.
Regards,
Patrick.
From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com
Have you enabled DTMF logging and seen the DTMF codes being recognised by
Asterisk? I had a bunch of soft phones that I had to change to using “sip
info” for the DTMF signalling as the RFC signalling was not always being
recognised. This would cause transfers to appear as if the user had not
On 16-12-14 14:00, Dario Estupinan wrote:
Como integrar asterisk con Ldap.??
https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver
Best,
Patrick
ps this mailing list uses the English language
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Thank you once again Richard. I think that covers all my confusion.
Regards,
Patrick.
From: Richard Mudgett rmudg...@digium.commailto:rmudg...@digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com
Date
that is causing softmix to not work
correctly.?
Regards,
Patrick.
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...@digium.com
Sent: 09 December 2014 20:49
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam
score:8%]
On Tue, Dec 9, 2014 at 1:35 PM, Patrick Beaumont
p.beaum...@hatsoffsoftware.co.ukmailto:p.beaum
On 05-12-14 08:25, Mayank Kumar Gour wrote:
Any help will be appreciated.
Help us help you by providing much much much more information then you
have right now. http://www.catb.org/esr/faqs/smart-questions.html
HTH,
Patrick
that. Or switch to CentOS7 which has an almost
current OpenLDAP. Both Symas and the LTB Project have current OpenLDAP
RPMs for EL6 ( EL7):
https://symas.com/products/symas-openldap-directory/
http://ltb-project.org/wiki/
HTH,
Patrick
version is subject to the POODLE vulnerability
for which a fix is available in 11.13.1.
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.13.1.tar.gz
HTH,
Patrick
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DAHDI. The one time I tried the
dahdi-hfcs stuff it seemed to work fine (very light usage only).
http://sourceforge.net/projects/dahdi-hfcs/
http://www.openvox.cn/pub/drivers/dahdi-linux-complete/
HTH,
Patrick
}*
Have you tried contacting the app_swift developer and/or filed a bug at
https://issues.asterisk.org/jira/secure/Dashboard.jspa ?
Should I look into using another TTS engine?
You could try UniMRCP which sits between Asterisk and Cepstral replacing
app_swift: http://unimrcp.org
HTH,
Patrick
of the Asterisk rules in
the next Fail2ban version (0.9.1).
https://github.com/fail2ban/fail2ban/pulls
HTH,
Patrick
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On 15-09-14 17:22, Rainer Piper wrote:
Hi Patrick,
github done ;-)
Thanks!
what is HTH ???
Hope this/that helps
http://www.internetslang.com/
http://www.urbandictionary.com/define.php?term=internet%20slang
HTH :)
Patrick
because extension not found in context 'default'.
Have a look at Fail2ban:
http://www.fail2ban.org/wiki/index.php/Main_Page
HTH,
Patrick
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New to Asterisk
://www.freepbx.org/support-and-professional-services
If you want to learn more about Asterisk in general then a good start is
to first read Asterisk: The Definitive Guide, 4th Edition and go
through the wiki at http://wiki.asterisk.org.
HTH,
Patrick
H264 or VP8.
HTH,
Patrick
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asterisk-users
clients you used, the Asterisk
version, the OS and the relevant Asterisk config.
Thanks,
Patrick
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Tuesday, September 02, 2014 6:39 PM
enabled.
Thanks Eric. The obvious difference is that your co-worker was using
H.263/H.263p while Bria uses H.264 or VP8. videosupport=yes was present
in my sip.conf so it might be the codec. Time for more tinkering.
Thanks,
Patrick
if (!strncasecmp(exten, tel:, 4)) {
exten += 4;
} else if (!strncasecmp(exten, sips:, 5)) {
exten += 5;
} else {
ast_log(LOG_WARNING, Huh? Not an RDNIS SIP header (%s)?\n, exten);
return -1;
}
HTH,
Patrick
but
I'm just not quite getting it.
How about the info on the Asterisk wiki:
https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels
On the left side there's a menu with examples and modifiers.
HTH,
Patrick
is between the 2 host machines host
kernels:
old: Linux 2.6.32-5-xen-686
new: Linux 3.13-0.bpo.1-amd64
With such an exotic setup and ancient versions of something that no
longer exists, I think you are SOL.
HTH,
Patrick
and make the required changes so it
works with the latest Asterisk 11. Or you could install the latest
FreePBX iso on the new machine if you prefer a GUI.
HTH,
Patrick
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and the other is for CAPI cards like
the Eicon Diva Server and the AVM C4.
Can't I just tell it somewhere to use the new card and I don't have to touch
the existing dialplans etc?
Nope.
HTH,
Patrick
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/wiki/display/AST/Development
Development related questions can best be asked on the asterisk-dev
mailing list or on irc.freenode.net in #asterisk-dev.
HTH,
Patrick
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at the CallManager as iirc it's EAL1 certified.
Re asterisk+architecture, Asterisk Security related best practices are
described here:
http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt
HTH,
Patrick
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cards for
your router from different carriers and test which carrier consistently
provides the best signal, least delay, packet loss and jitter.
HTH,
Patrick
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then that might not work because of the '@'.
You can easily test this by adding a peer [test_1234] (so with the
recommended syntax) and add it to your LDAP server with a password and
then check if it registers.
HTH,
Patrick
On 20-06-14 15:05, Linus Lüssing wrote:
[snip]
having [test_phone_120d] in my sip.conf works fine. Ah wait - do
I need to have a user both in LDAP and sip.conf and the only
thing LDAP can do for me is the authentication/password checking?
As far as I know, yes :)
Cheers,
Patrick
.
See http://heartbleed.com/
Unless you want to mess around with the Men in Black and leave your
system vulnerable to attack, you should install all security updates
ASAP and then restart the services that rely upon them.
HtH,
Patrick
your needs then maybe have a look at Jitsi
https://jitsi.org/ or Linphone https://www.linphone.org/
I prefer the client to have at least the following features:
Security:
- TLS
- SRTP
- ZRTP
Codecs:
- G722
- G729
Fight NAT (if IPv6 is not an option):
- STUN
- TURN
- ICE
Cheers,
Patrick
]: tcptls.c:274 handle_tcptls_connection:
FILE * open failed!
Anyone know what that error means? The source code does not tell me
much. FWIW the same setup works fine with an Android phone.
Thanks!
Patrick
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in Linphone.
Cheers,
Patrick
On Jun 8, 2014 6:50 PM, Patrick Laimbock patr...@laimbock.com
mailto:patr...@laimbock.com wrote:
Hi,
I'm trying to setup an iPhone 4S (iOS 7.1.1) with Linphone to
register with TLS to an Asterisk 11.10.0 box. The registration fails
and I see
. Is something like this
already implemented?
Have you looked at Call Completion Supplementary Services (CCSS)?
https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5243096
Cheers,
Patrick
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updating a RHEL5/CentOS5 kernel I always had to recompile
DAHDI for the new EL5 kernel.
HtH,
Patrick
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and
delete after email.
A copy of the message for each will be dropped into 2000 and deleted from the
original box.
Thanks John. I'll give that a try.
Cheers,
Patrick
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within the
dialplan.
If that's not possible, would adding an extension option to
app_voicemail.c solve this by decoupling the extension from the mailbox?
Thanks for any pointers.
Cheers,
Patrick
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On 30-04-14 12:50, [Digital^Dude] ® wrote:
make gives this:
IIRC Digium's policy is that there's no support on this list for
patented technologies like AMR which are possibly not officially
licensed. Obviously to prevent any legal liability.
HTH,
Patrick
it to you.
HTH,
Patrick
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asterisk-users
away.
HTH,
Patrick
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asterisk-users mailing
On 28-04-14 20:13, Haley,Scott A wrote:
That seemed to fix it. Thanks to everyone.
https://bugzilla.redhat.com/show_bug.cgi?id=1092150
HTH,
Patrick
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the sip.conf
example file to see what these options do and use what's best for your
situation.
canreinvite=no
directmedia=no
directrtpsetup=no
HTH,
Patrick
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Hi,
I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI
command to see if SRTP is active on a channel/call. I went through sip
show ... and core show channel... and did not see any mentioning of SRTP
while there is an SRTP call active.
Thanks,
Patrick
24 16:20 nexus.crt
The certs were created with ast_tls_cert as described in the tutorial. I
created a nexus.p12 for the phone and imported it before configuring
CSipSimple.
Does anyone know what's wrong? Pointers much appreciated.
Thanks,
Patrick
[0] https://wiki.asterisk.org/wiki/display
On 24-03-14 21:28, Patrick Laimbock wrote:
[snip]
== Problem setting up ssl connection: error:14094410:SSL
routines:SSL3_READ_BYTES:sslv3 alert handshake failure
[Mar 24 21:20:56] WARNING[28467]: tcptls.c:272 handle_tcptls_connection:
So others may find the fix: make sure the server
building installing the stuff from http://misdn.eu Works fine
with a USB TA with HFC chipset last time I tested it.
HTH,
Patrick
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On 03/15/2014 10:15 PM, binary wrote:
i have tried the misdn from git. my problem is that it needs LCR and it
fails to get installed
Then you need to fix that. AFAIK there is no other way to use a USB ISDN
TA than via mISDN/LCR.
HTH,
Patrick
still can't figure it out perhaps ask on the ISDN4Linux mailing
list: https://www.isdn4linux.de/mailman/listinfo/isdn4linux
Cheers,
Patrick
ps don't build as root, it's bad practice.
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Patrick
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