Displaying 20 results from an estimated 2000 matches similar to: "SRTP testers needed"
2007 Mar 23
3
SRTP testers needed
please look at
http://www.voip-info.org/wiki/view/Asterisk+SRTP
and try compile&run clients with srtp (linksys,gxp-2000, minisip, twikle,
...)
---------------------------------------
Marek Cervenka
Centrum Vypocetni Techniky
CVT - http://cvt.fpf.slu.cz
FPF SLU OPAVA - http://www.fpf.slu.cz
LCNA - http://lcna.slu.cz
=======================================
2007 May 01
0
Re: [asterisk-dev] SRTP implementation
> Olle E Johansson wrote:
>>
>> 23 apr 2007 kl. 19.55 skrev Russell Bryant:
>>
>>> John Todd wrote:
>>>> To morph this into a -dev thread: if this patch were to become (again)
>>>> useful and error-free, is there any objection or usefulness in adding it
>>>> to TRUNK? Personally, I think there is, if there is a method by which
2010 May 20
1
Asterisk T.38 Gateway code testing
hi,
i made page for Asterisk T.38 Gateway code testing
http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway
Final T.38 Gateway API for asterisk 1.8 will be posted by Kevin Fleming
later BUT Asterisk 1.8 is too far and we need t.38 gw now
if you would like help/test current code(last patch from
https://issues.asterisk.org/view.php?id=13405), please contact me
i have 2 public testing
2015 Aug 13
2
simultaneous use of chan_sip/chan_pjsip
Dne 13.8.2015 v 17:20 Rusty Newton napsal(a):
> On Thu, Aug 13, 2015 at 3:54 AM, Marek ?ervenka <cervajs at fpf.slu.cz
> <mailto:cervajs at fpf.slu.cz>> wrote:
>
> hello,
>
> is it possible simultaneously use chan_sip and chan_pjsip?
>
> if yes, can you recommend settings
>
> i'm thinking about
> - chan_sip - for sip
2016 Jan 29
2
asterisk 13 mixmonitor - random missing syllables
Dne 28.1.2016 v 13:37 Brian :: napsal(a):
> when you say load - how many concurrent calls? Is there transcoding
> happening? sip / PRIs ? what load?
>
12 concurrent calls
no transcoding
SIP
under 1.5 with 4x 1Ghz vcpus (its vmware VPS)
> On Thu, Jan 28, 2016 at 9:57 AM, Marek ?ervenka <cervajs at fpf.slu.cz
> <mailto:cervajs at fpf.slu.cz>> wrote:
>
>
2010 Oct 12
2
libsrtp package anywhere?
Hi list,
I'm trying to create an asterisk 1.8 rpm with SRTP.
I found mention of a libsrtp rpm,
<http://qutecom.ipex.cz/RPMS/srtp-1.4.4-1.i386.rpm >
in these instructions,
<http://www.voip-info.org/wiki/view/Asterisk+SRTP>
but it is unreachable (by me, anyway).
The libSRTP source is here,
<http://srtp.sourceforge.net/download.html>.
Has this already been packaged for
2016 Feb 29
2
Asterisk 13 and WebRTC. Is wiki page still valid ?
2016-02-19 12:01 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>:
> on my own server
>
Today, I'm back from holidays trip.
First of all, thanks for replying !
I'll try to use jssip as you suggested.
Anyway, I'm still failing to understand if wiki's page [1] is still valid
with Asterisk 13, and if it's not valid anymore, which is the main change
that prevent
2007 May 16
1
Asterisk SRTP certificates
Hello all,
I want to use Asterisk with the SRTP patch from
http://bugs.digium.com/view.php?id=5413 .
I'm confused to create the certificates for it.
Can anybody help in such question?
P. S. I've created the pem files and renamed it to
* ${astetcdir}/asterisk.crt
* ${astetcdir}/asterisk.key
* ${astetcdir}/ca-certificates.crt
but the asterisk got "segmentation fault" error at
2008 Oct 17
1
Asterisk SIP and SRTP
Hello,
are there any plans in including SRTP into Asterisk?
The patches in http://bugs.digium.com/view.php?id=5413 are pretty old
and do not work with asterisk 1.6.0.
Thanks, Artem
2016 Feb 18
2
Asterisk 13 and WebRTC. Is wiki page still valid ?
2016-02-18 15:42 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>:
> my experience with pjsip for webrtc
> http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html
>
>
> Yes I saw this post earlier today.
Having to fight 14 days scared me a bit !
Did you set sipml5 on your own server or did you use Live demo (
2015 Oct 09
2
Storing HANGUPCAUSE in CDR
This was always possible in the past, however does not work in the current release.
I believe this is a bug.
To: asterisk-users at lists.digium.com
From: cervajs at fpf.slu.cz
Date: Fri, 9 Oct 2015 10:04:47 +0200
Subject: Re: [asterisk-users] Storing HANGUPCAUSE in CDR
search in archives
save the records to another table like cdr_extended
Dne
2015 Jun 16
1
Req help regarding webRTC : Attempted Attempted to set an invalid DTLS-SRTP configuration on RTP instance
Hi List,
I am trying to setup a Asterisk setup in AWS instance Centos6.5 . I
have installed Asterisk 13.4 with srtp,pjproject. I have configured two
numbers for webRTC clients, when i try to call from a client (sipml5) to
another client (sipml5) it throws the following error:
"chan_sip.c:5851 dialog_initialize_dtls_srtp: Attempted to set an invalid
DTLS-SRTP configuration on RTP
2007 Mar 20
4
blktap howto
hi,
i''m trying move from file: based disk to tap:aio but things don''t work
i have centos4 dom0 with centos4 domU
xen 3.0.4-testing changeset: 13138:d401cb96d8a0 self compiled
[root@xen linux-2.6.16.38-xen]# grep XEN_BLKDEV_TAP .config
CONFIG_XEN_BLKDEV_TAP=m
config
disk = [ ''file:/var/lib/xen/test.img,hda1,w'',
2011 Oct 05
1
call pickup
hello,
is there some way to notify people in the same pickup group about call
from caller to callee?
i.e. i have call from 111 to 222
there are 222,333,444 in the same pickup group
333,444 see on the phone (aastra) that 111 calling to 222 and can pickup
the call with *8
siemens have this on their sip openstage phones. how they do this?
thanks
--
---------------------------------------
Marek
2011 Aug 03
2
snom and srtp
Hi,
I am running asterisk 1.8.5.0 and have compiled in the srtp module
All but Snom phones are working.
I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when trying to make a call (this is snom calling snom).
---------snip------------------
== Using SIP RTP CoS mark 5
-- Executing [10000 at
2015 Aug 13
2
simultaneous use of chan_sip/chan_pjsip
hello,
is it possible simultaneously use chan_sip and chan_pjsip?
if yes, can you recommend settings
i'm thinking about
- chan_sip - for sip hardphones/softphones (sip udp 5060)
- chan_pjsip - for webrtc
--
---------------------------------------
Marek Cervenka
=======================================
2019 Feb 23
2
configure SRTP port range?
On 2/22/19 7:56 PM, Joshua C. Colp wrote:
> On Fri, Feb 22, 2019, at 2:48 PM, hw wrote:
>>
>> Hi,
>>
>> when trying to use SRTP, I can see UDP traffic from phones to the
>> asterisk server being dropped be the firewall on arbitrary ports.
>
> There is no separate port range used for SRTP, and Asterisk does not control the port that the phone uses for sending
2014 Apr 05
1
Asterisk and SRTP
Hi experts,
I am trying Asterisk SRTP in my environment, and find that when Asterisk
is behind a NAT, the audi/video UDP ports opened for SRTP relay by Asterisk
are local ports on the Asterisk server, media from the two clients out of
the NAT (for example from Internet) can not reach the ports, and thus the
two client can not establish the secure call via Asterisk. I have set up a
STUN server
2019 Feb 23
3
configure SRTP port range?
On 2/23/19 1:15 PM, Joshua C. Colp wrote:
> On Sat, Feb 23, 2019, at 8:06 AM, hw wrote:
>> On 2/22/19 7:56 PM, Joshua C. Colp wrote:
>>> On Fri, Feb 22, 2019, at 2:48 PM, hw wrote:
>>>>
>>>> Hi,
>>>>
>>>> when trying to use SRTP, I can see UDP traffic from phones to the
>>>> asterisk server being dropped be the firewall
2019 Feb 22
2
configure SRTP port range?
Hi,
when trying to use SRTP, I can see UDP traffic from phones to the
asterisk server being dropped be the firewall on arbitrary ports.
Where do I configure the SRTP port range (like the rtp port range)?
Why aren't the clients talking to each other directly but apparenty try
to send the SRTP traffic to the server?
That the traffic is being blocked by the firewall is probably the reason