Adding the Free-RTC mailing list - anybody else want to join the dinner
that is being proposed for Saturday night at FOSDEM?
Have any details been confirmed? Is there a deadline or maximum number
of people?
On 20/01/17 10:38, Alexandr Dubovikov wrote:
> Hi Daniel,
>
> IMHO we can do like last
On 31/01/17 11:44, Daniel-Constantin Mierla wrote:
> Hello,
>
>
> On 31/01/2017 11:08, Daniel Pocock wrote:
>> Hi,
>>
>> Are there any public databases of dial plan information? Are there any
>> schemas (e.g. XML / XSD) useful for describing arbitrary dial pla
Hi,
Are there any public databases of dial plan information? Are there any
schemas (e.g. XML / XSD) useful for describing arbitrary dial plans from
different carriers around the world?
The type of things that I'm interested in:
- being able to translate any local number to E.164 (if possible)
Does anybody know if SEMS has HOMER / HEP support like Kamailio, or if
it is under development?
It would be useful to extract RTCP stats from SRTP streams, the
standalone HOMER captagent can't see inside them when they are encrypted
(SRTCP)
Regards,
Daniel
I recently set up a Kamailio instance using the default configuration
for HOMER with MySQL[1]
The database was not running and Kamailio refused to start.
Would it be better for Kamailio to start anyway and go into a loop
trying to connect to the database, just as if auto_reconnect was set?
Thi
servers that would be very welcome.
Forwarded Message
Subject: [Free-RTC] GSoC progress, more mentors?
Date: Mon, 13 Jun 2016 22:53:46 +0200
From: Daniel Pocock
Reply-To: Free real-time-communications discussion list
To: free-...@lists.fsfe.org
Hi all,
The GSoC students have
On 28/03/16 22:56, Daniel-Constantin Mierla wrote:
> Hello,
>
> no much spare time (which I guess everyone lacks of, anyhow), but
> ultimately I could assist a bit someone that is going to work with C
> (eventually C++) and/or WebRTC -- not doing much of the Java these days,
> I will let that fo
Hi all,
We had a few GSoC applications[1] from students who would like to do
work relating to SIP and WebRTC this summer.
The students have a range of skills. Some are interested in the server
side (C, C++) code. Some are interested in Android and Java projects
(e.g. the Lumicall and CSipSimpl
Forwarded Message
Subject:[Free-RTC] FOSDEM Real Time devoom: call for help
Date: Sun, 24 Jan 2016 19:17:02 +0100
From: Saúl Ibarra Corretgé
Reply-To: Free real-time-communications discussion list
To: Free real-time-communications discussion list
, sum..
On 25/01/16 21:59, Daniel-Constantin Mierla wrote:
> Hello,
>
> so it seems we may get to be like 20 people. Anyone here that can
> suggest a decent place in Brussels with food and drinks that can
> accommodate such group?
>
Could you handle three more people: Adrien and Guillaume from ring.cx
an
Reminder: speaker's deadline this Friday, 27 November at 23:59 UTC
We have already received several really exciting talk proposals
but there is still time for people to propose talks or encourage
friends or colleagues to speak.
Many other dev-rooms also have a deadline in the next few days and i
ite.
Contact
===
For discussion and queries, please join the free-rtc mailing list:
https://lists.fsfe.org/mailman/listinfo/free-rtc
The dev-room administration team:
Daniel Pocock
Ralph Meijer
Saúl Ibarra Corretgé
Iain R. Learmonth
Contributors to v1.10.0 include Alan Hawrylyshen, Alexander G.
Pronchenkov, Byron Campen, Daniel Pocock, Daniel Tacalau, Dan Petrie,
Dario Bozzali, Fedor Brunner, Gregor Jasny, Kevin Green, Reiner
Herrmann, Robert Sparks, Scott Godin, Thomas Thorne, Vasil Kolev and
Wolfgang Rosenauer. Please also
I'll be in Norfolk, VA for xTupleCon in October
On 15 October, there will be two events for WebRTC:
14:15 a talk about the xTuple WebRTC extension at xTupleCon
- must register for xTupleCon to attend this
17:30 a technical / developer workshop at xTuple's offices
- free, anybody welcome,
, Francis Joanis, Matthias Moetje, Catalin
Constantin Usurelu and Daniel Pocock. We are also very grateful to
all those who contributed feedback through the mailing list, testing
the beta packages in Debian and all past contributors to previous
releases of the reSIProcate project.
Any and all feedback is
On 24/01/14 17:04, Daniel-Constantin Mierla wrote:
> Hello,
>
> ok, since there were no other proposals so far, then let's join Jitsi at
> Beermania and if there is need for more food, we will figure out
> something on the fly.
I might be able to come down there too
It looks close enough to th
On 10/01/14 17:23, Henning Westerholt wrote:
> Am Mittwoch, 8. Januar 2014, 20:07:58 schrieb Daniel-Constantin Mierla:
>> not much about real time communications this year at Fosdem, but being a
>> convenient venue for me, the plan at this moment is to go .
>>
>> I saw some notes online from vari
On 10/01/14 10:35, Daniel-Constantin Mierla wrote:
>
> On 09/01/14 20:12, Daniel Pocock wrote:
>>
>> On 09/01/14 20:05, Daniel-Constantin Mierla wrote:
>>> Hello,
>>>
>>> I haven't used it, or better said, it's first time I hear about,
&g
On 09/01/14 20:05, Daniel-Constantin Mierla wrote:
> Hello,
>
> I haven't used it, or better said, it's first time I hear about, perhaps
> I missed some messages on various mailing lists. Anyhow, do you have any
> screenshots available? Will be easier to get a feeling about it.
Instead of a scr
Hi,
I'm just wondering if anybody has used JSCommunicator with Kamailio yet
and if anybody has any feedback?
http://jscommunicator.org/
If anybody publishes a dedicated blog about using them together I'll be
happy to link to it
I'm also looking for any feedback from anybody who may be able t
On 10/06/13 13:05, Klaus Darilion wrote:
>
>
> On 06.06.2013 16:35, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> On 6/6/13 11:05 AM, Daniel Pocock wrote:
>>> I was just looking over:
>>>
>>> http://kb.asipto.com/asterisk:realtime:kamailio-3
On 10/06/13 09:32, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 6/7/13 11:33 AM, Daniel Pocock wrote:
>> On 06/06/13 16:35, Daniel-Constantin Mierla wrote:
>>> Hello,
>>>
>>> On 6/6/13 11:05 AM, Daniel Pocock wrote:
>>>> I
On 06/06/13 16:35, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 6/6/13 11:05 AM, Daniel Pocock wrote:
>> I was just looking over:
>>
>> http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
>>
>>
>> A couple of things I noticed:
I was just looking over:
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
A couple of things I noticed:
- Kamailio is using a column sippasswd which is not hashed. Asterisk
doesn't use that column at all. Is there any reason this can't be done
with the H(A1) and H(A
On 13/05/13 16:58, Jesús Pérez Rubio wrote:
> Hi Daniel,
>
> We have something like you're asking for in the Github repository. First
> lines of this Quickstart guide show it (
> https://quobis.atlassian.net/wiki/display/QoffeeSIP/Quick+start+guide).
> Only two steps are needed:
> - Clone the re
On 13/05/13 12:24, James Cloos wrote:
>>>>>> "DP" == Daniel Pocock writes:
> DP> I'd like to write a brief blog about the status of WebRTC in Debian,
> DP> with a focus on SIP
>
> DP> I understand Kamailio 4.0.1 is already in unstable, i
I'd like to write a brief blog about the status of WebRTC in Debian,
with a focus on SIP
I understand Kamailio 4.0.1 is already in unstable, is that recommended
for potential WebSocket users? Has anybody else written any quickstart
blog about WebRTC with that particular version, possibly with ex
I've just released DruCall, the WebRTC module for Drupal.
http://danielpocock.com/announcing-drucall-webrtc-for-drupal
Making WebRTC available as a Drupal module hopefully means thousands of
blogs and small business web sites will start picking this up and
hopefully Kamailio will share in so
On 22/01/13 12:15, Benjamin Henrion wrote:
> On Tue, Jan 22, 2013 at 12:04 PM, Daniel-Constantin Mierla
> wrote:
>> Hello,
>>
>> I will probably be able to join, so I added myself on the list for the
>> moment. But it looks like 18 seats is not gonna be enough...
>
> I can rebook another restau
Just wondering if a Saturday night telephony dinner has been confirmed
by sr-dev, or if anybody is aware of similar group dinners on that night?
On 16/01/13 10:13, Carsten Bock wrote:
> Hi,
>
> count me in as well (and my wife, too; even though she mostly does
> non-VoIP-Java stuff, except f
ifferent, hence a new thread
> regards
> Klaus
>
> On 11.01.2013 18:45, Daniel Pocock wrote:
>>
>>
>>
>> I'm just wondering if anyone can comment on expected and actual behavior
>> if there is only a NAPTR record for TLS, e.g. I have:
>>
>&g
I'm just wondering if anyone can comment on expected and actual behavior
if there is only a NAPTR record for TLS, e.g. I have:
sip5060.net. INNAPTR10 0 "s" "SIPS+D2T" ""
_sips._tcp.sip5060.net.
and I don't have any entry for "SIP+D2U" or "SIP+D2T"
If some third party Kamaili
m.org/2013/schedule/event/free_open_secure_communications/
https://fosdem.org/2013/interviews/2013-daniel-pocock-and-peter-saint-andre-and-simon-tennant-and-evan-prodromou-and-daniel-constantin-mierla-and-emil-ivov/
> On 09 Jan 2013, at 19:39, Olle E. Johansson wrote:
>
>>
>>
I've recently released a dlz ENUM module for the bind9 nameserver:
http://www.opentelecoms.org/dlz-ldap-enum
Basically, it handles ENUM queries from Kamailio, Asterisk, FreeSWITCH,
repro, Lumicall, searches for the phone number in LDAP, and if found,
returns the email address as both a SIP a
Hi,
I'm just wondering if there has been any type of work to implement any
part of RFC 5626, particularly the registrar server code?
Lumicall now generates a UUID at install time, and sends that as
sip.instance in the Contact header. It always sends reg-id=1 (only
trying one proxy at the momen
On 17/02/12 20:34, Bruno Bresciani wrote:
> Thanks Daniel
>
> but I have some problems...
>
> a) my SIP gateway doesn't send the certificate when it isn't demanded...
>
> b) I was reading about TLS specific config file (tls.cfg), but my
> requirements doesn't allow to configure differents TLS
On 17/02/12 19:01, Bruno Bresciani wrote:
> Hi All,
>
> Does it possible on tls module require certificates only some hosts?
Yes, you have at least two options:
a) just set the require_certificate 0 option - make sure your client
still sends it's cert even when it is not demanded - and in your
On 17/02/12 00:40, David wrote:
> What SIP stack does this use?
It is based on the MJSIP SIP stack (same stack that is used in Sipdroid)
> Is it open source?
>
http://www.mjsip.org/
I've made some fixes to the stack, I've tried to contact the developers
about contributing the code back, but
>> Another strategy is to modularise the app: e.g. divide Lumicall into 3
>> apps, each with less permissions, and they collaborate using
>> inter-process communication (IPC)
> Not sure it will help much, from my point of view -- having too many
> apps to install for getting a proper user experien
> I installed it from the download page, worked fine on android 2.2 -- had
> no time to test it yet, but I noticed some "warning" messages during
> installation. Not sure if it is specific for each android phone type or
> for android in general, but I was alerted that I will allow the
> applicatio
On 09/02/12 21:49, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 2/9/12 5:21 PM, Daniel Pocock wrote:
>>
>> On 09/02/12 01:41, Daniel Pocock wrote:
>>>
>>> I've been contemplating Daniel's earlier question about using the CAcert
>>
On 09/02/12 01:41, Daniel Pocock wrote:
>
>
> I've been contemplating Daniel's earlier question about using the CAcert
> certificates with Lumicall
>
> sip5060.net should already accept mutual authentication from other
> Kamailio instances running with a CAcer
I've been contemplating Daniel's earlier question about using the CAcert
certificates with Lumicall
sip5060.net should already accept mutual authentication from other
Kamailio instances running with a CAcert certificate
However, the Lumicall dialer itself will only connect to servers that
are u
On 30/01/12 09:08, Klaus Darilion wrote:
>
>
> On 29.01.2012 12:21, Daniel Pocock wrote:
>>
>>
>>
>> There are now a number of TURN implementations available:
>>
>> http://www.resiprocate.org/ReTurn_Overview
>> http://turnserver.sourceforge.ne
>> I notice that Asterisk needs to be patched to do it the way Kamailio does:
>>
>> https://issues.asterisk.org/jira/browse/ASTERISK-17727
>
> The Asterisk TCP/TLS implementation is marked experimental for a reason. And
> it's been that way for many years.
All the more reason for people to use
On 29/01/12 21:47, Iñaki Baz Castillo wrote:
> 2012/1/29 Daniel Pocock :
>> It's a little bit different in Apache, where the user specifies a file
>> containing intermediate certs - many of the CAs give instructions for
>> adding that file in Apache, but they mak
>> Construct the PEM file in this exact order:
>>
>> cat server.example.com.pem > chain-server.example.com.pem
>> cat inter2.pem >> chain-server.example.com.pem
>> cat inter1.pem >> chain-server.example.com.pem
>>
>> and then, in tls.cfg:
>>
>> certificate=chain-server.example.com.pem
>>
>
> This
I found that my TLS client was not happy because my server cert is
signed by an intermediate root.
A quick search in Google found other people mentioning the same problem,
but no solution or documentation.
I've had a quick look in the Kamailio source and I notice it is using
the call:
There are now a number of TURN implementations available:
http://www.resiprocate.org/ReTurn_Overview
http://turnserver.sourceforge.net/
and for many purposes TURN relays and ICE are much better than relying
on NAT helper and rtpproxy
When clients use ICE:
- the client gets upset if Kamailio m
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