On 21.10.19 at 17:23 Michael Maier wrote:
New patchset for Asterisk 18.4. As I don't use other versions of Asterisk any
more, I don't have a patchset for those versions.
How should it all be used now?
If you want to use SIPS and SRTP with Deutsche Telekom AllIP, you have to be
sure to enable
On 21.10.19 at 17:23 Michael Maier wrote:
Attached are actual patches for Asterisk 16.16.0-rc1, 18.0.1 and 18.2.0-rc1 (one
patch for each version). Version 16.16.0-rc1 is only compile tested (16.14.x was
the last asterisk 16 version I used myself). 18.2.0-rc1 is used here (as 18.1 and
Hi!
Meanwhile, there is an improved version of the mediasec patch, which adds a
switch to enable mediasec headers for each endpoint individually. This patch is
thankfully
provided by André Valentin (avalen...@marcant.net /
https://www.marcant.net/en/). It's the patch
On 02.09.19 at 19:03 Michael Maier wrote:
> On 30.05.19 at 10:24 Michael Maier wrote:
>> Hello!
>>
>> I wrote some code, which adds basic media encryption support to be used with
>> Deutsche Telekom. The attached patch is based on Asterisk 16.3
>> and works for me :-) - not fully tested yet. If
Hello Michael,
i just tested your patch with my tcom setup. I noticed that it works in most
cases.
On case that leads to a fail is a reinvite because of codec or connect line
information change. Take a look:
Calls starts:
INVITE sip:0191...@tel.t-online.de SIP/2.0
Via: SIP/2.0/TLS
On 30.05.19 at 10:24 Michael Maier wrote:
> Hello!
>
> I wrote some code, which adds basic media encryption support to be used with
> Deutsche Telekom. The attached patch is based on Asterisk 16.3
> and works for me :-) - not fully tested yet. If you want to use it, you have
> to enable
Hello!
That's the *complete* patch for Asterisk 16.3 (can be applied to asterisk 16.4,
too). Logging has been changed to ast_debug(3,).
See previous posts for information about usage.
Regards,
Michael
diff -urN asterisk-16.3.0.orig/res/res_pjsip/pjsip_options.c
On 31.05.19 at 19:25 Joshua C. Colp wrote:
> On Fri, May 31, 2019, at 1:57 PM, Michael Maier wrote:
>
>
>
>>
>> Thanks Joshua!
>>
>> I added the attached changes. Afterwards, I could see one SIGSEGV so
>> far which I can't understand. I would be very happy, if you could take
>> a
>> look at
On Fri, May 31, 2019, at 1:57 PM, Michael Maier wrote:
>
> Thanks Joshua!
>
> I added the attached changes. Afterwards, I could see one SIGSEGV so
> far which I can't understand. I would be very happy, if you could take
> a
> look at it - maybe you have an idea? Most probably I'm doing
On 31.05.19 at 11:34 Joshua C. Colp wrote:
On Fri, May 31, 2019, at 3:38 AM, Michael Maier wrote:
On 30.05.19 at 10:24 Michael Maier wrote:
[...]
Another yet missing point is the qualify OPTIONS package. I'm not sure where to
add the mediasec headers exactly (which function?). At the
moment,
On Fri, May 31, 2019, at 3:38 AM, Michael Maier wrote:
> On 30.05.19 at 10:24 Michael Maier wrote:
> [...]
> > Another yet missing point is the qualify OPTIONS package. I'm not sure
> > where to add the mediasec headers exactly (which function?). At the
> > moment, the Response after OPTION
On 30.05.19 at 10:24 Michael Maier wrote:
[...]
Another yet missing point is the qualify OPTIONS package. I'm not sure where to
add the mediasec headers exactly (which function?). At the
moment, the Response after OPTION request is (if already registered):
SIP/2.0 494 Security Agreement
Hello!
I wrote some code, which adds basic media encryption support to be used with
Deutsche Telekom. The attached patch is based on Asterisk 16.3
and works for me :-) - not fully tested yet. If you want to use it, you have to
enable media_encryption=sdes for the extension (and
transport tls
On 28.05.19 at 15:12 Joshua C. Colp wrote:
> On Sun, May 26, 2019, at 1:51 PM, Michael Maier wrote:
>> On 24.05.19 at 12:45 Joshua C. Colp wrote:
>>> On Fri, May 24, 2019, at 3:35 AM, Michael Maier wrote:
>> - Is there any possibility in handle_client_registration to check if
>> encryption is
On Sun, May 26, 2019, at 1:51 PM, Michael Maier wrote:
> On 24.05.19 at 12:45 Joshua C. Colp wrote:
> > On Fri, May 24, 2019, at 3:35 AM, Michael Maier wrote:
> >> On 03.02.19 at 12:00 Joshua C. Colp wrote:
> >>> On Sun, Feb 3, 2019, at 6:03 AM, Michael Maier wrote:
> On 15.01.19 at 20:27
On Sun, May 26, 2019, at 1:51 PM, Michael Maier wrote:
> On 24.05.19 at 12:45 Joshua C. Colp wrote:
> > On Fri, May 24, 2019, at 3:35 AM, Michael Maier wrote:
> >> On 03.02.19 at 12:00 Joshua C. Colp wrote:
> >>> On Sun, Feb 3, 2019, at 6:03 AM, Michael Maier wrote:
> On 15.01.19 at 20:27
On 24.05.19 at 12:45 Joshua C. Colp wrote:
> On Fri, May 24, 2019, at 3:35 AM, Michael Maier wrote:
>> On 03.02.19 at 12:00 Joshua C. Colp wrote:
>>> On Sun, Feb 3, 2019, at 6:03 AM, Michael Maier wrote:
On 15.01.19 at 20:27 Joshua C. Colp wrote:
[...]
In which function should the
On Fri, May 24, 2019, at 3:35 AM, Michael Maier wrote:
> On 03.02.19 at 12:00 Joshua C. Colp wrote:
> > On Sun, Feb 3, 2019, at 6:03 AM, Michael Maier wrote:
> >> On 15.01.19 at 20:27 Joshua C. Colp wrote:
> >
> >
> >
> >>
> >>
> >> If I wanted to try it myself - what would be the correct
On 03.02.19 at 12:00 Joshua C. Colp wrote:
> On Sun, Feb 3, 2019, at 6:03 AM, Michael Maier wrote:
>> On 15.01.19 at 20:27 Joshua C. Colp wrote:
>
>
>
>>
>>
>> If I wanted to try it myself - what would be the correct places to
>> implement it?
>>
>> It shouldn't be that complicated, because it
On Sun, Feb 3, 2019, at 6:03 AM, Michael Maier wrote:
> On 15.01.19 at 20:27 Joshua C. Colp wrote:
>
>
> If I wanted to try it myself - what would be the correct places to
> implement it?
>
> It shouldn't be that complicated, because it seems mostly to be done by
> adding some additional
On 15.01.19 at 20:27 Joshua C. Colp wrote:
>
>
> On Tue, Jan 15, 2019, at 3:23 PM, Michael Maier wrote:
>> Hello!
>>
>> Deutsche Telekom introduced sips and srtp. I tested it and it works
>> partly. Partly means: sips is working - but not srtp. srtp doesn't
>> work, because of missing
On 1/15/19 1:49 PM, Michael Maier wrote:
Hello!
Deutsche Telekom introduced sips and srtp. I tested it and it works partly.
Partly means: sips is working - but not srtp. srtp doesn't work, because of
missing additional
headers in the REGISTER and INVITE packages (according an enhancement of
Howdy,
As Asterisk is an open source project, new capabilities are implemented by
many different individuals and organizations, each driven by their own
self-interests. We at Digium and Sangoma, who do represent a large share
of the development contributions to Asterisk, are happy to take any
On 15.01.19 at 20:27 Joshua C. Colp wrote:
>
>
> On Tue, Jan 15, 2019, at 3:23 PM, Michael Maier wrote:
>> Hello!
>>
>> Deutsche Telekom introduced sips and srtp. I tested it and it works
>> partly. Partly means: sips is working - but not srtp. srtp doesn't
>> work, because of missing
On Tue, Jan 15, 2019, at 3:23 PM, Michael Maier wrote:
> Hello!
>
> Deutsche Telekom introduced sips and srtp. I tested it and it works
> partly. Partly means: sips is working - but not srtp. srtp doesn't
> work, because of missing additional
> headers in the REGISTER and INVITE packages
Hello!
Deutsche Telekom introduced sips and srtp. I tested it and it works partly.
Partly means: sips is working - but not srtp. srtp doesn't work, because of
missing additional
headers in the REGISTER and INVITE packages (according an enhancement of RFC
3329).
Example:
UAC
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