ACK and then the 200ok for INVITE should have
been re-transmitted few times within the 3 seconds you wait before
sending the BYE.
Cheers,
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com
Kamailio Worl
On 3/2/11 12:30 PM, Evgeniy Khramtsov wrote:
> 02.03.2011 19:41, Daniel-Constantin Mierla wrote:
>>>> If that is still hard, then go for sip:provider CE - after you install
>>>> it, then you just play with the web interface for managing sip
>>>> accounts
On 3/2/11 11:12 AM, Evgeniy Khramtsov wrote:
> 02.03.2011 17:25, Daniel-Constantin Mierla wrote:
>> Have you played with kamailio 3.1? You barely touch the config file
>> for most of the features there, just need to add simple lines like:
>> !#define WITH_XYZ and that'
en go for sip:provider CE - after
you install it, then you just play with the web interface for managing
sip accounts for your phones.
Btw, if you don't like the config file language, try with Lua :-)
* http://asipto.com/u/h
Cheers,
Daniel
es or
image for VirtualBox or VMWare -- it is free as well, see more details
about features and installation guidelines at:
http://asipto.com/u/1d
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
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Sip-
le from the project
to connect and meet face to face in the area we have other events. By
its type and duration, it is not technical oriented, just discussions
about present and the future development.
Thanks,
Daniel
> Thank you,
>
> James
>
>
> -Original Message-
> Fr
IP side
> and/or handset-to-handset side.
>
> Thank you,
>
> James
> Mocana Corp
>
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Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Jan 24-26, 2011, Irvine, CA, USA
http://www.asipto.com
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S
st (i.e., when kamailio was named openser and even 2 releases after),
there was no 'auth-int' support, only 'auth' qop.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
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Sip-imp
able/modules/auth.html#www_challenge
Cheers,
Daniel
--
Daniel-Constantin Mierla
Kamailio Advanced Training
Nov 22-25, 2010, Berlin, Germany
Jan 24-26, 2010, Irvine, CA, USA
http://www.asipto.com
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On 5/28/10 4:49 PM, Iñaki Baz Castillo wrote:
> 2010/5/28 Daniel-Constantin Mierla:
>
>> Hello,
>>
>> I added an XCAP server (supports basic operations by now) to open source
>> SIP servers Kamailio (OpenSER) and SIP Express Router (SER).
>>
>> Being
ks,
Daniel
--
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Miami, Fl, USA - June 21-23, 2010
http://www.asipto.com/index.php/kamailio-advanced-training/
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> http://lmgtfy.com/?q=sip+test+tools
>
>
:-) nice page ...
Daniel
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Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Miami, Fl, USA - June 21-23, 2010
http://www.asipto.com/index.php/kamailio-advanced-training/
___
lp ? Else, how do i make it ?
>
if you want more than two participants in a call, then you need an audio
conferencing server (able to mix the audio streams), for open source
apps, check asterisk, freeswitch or sems. Also, many phones can host
small conferences for three users.
Cheers,
Da
On 02/26/2009 09:44 PM, Aymeric Moizard wrote:
>
>> 2009/2/26 Daniel-Constantin Mierla :
>
>> True, what I mean is that SIP providers don't offer a NAPTR record for
>> their service since most clients don't implement it.
>
> But NAPTR request on windows
On 02/26/2009 07:30 PM, Iñaki Baz Castillo wrote:
> 2009/2/26 Daniel-Constantin Mierla :
>
>>> Devices don't implement it, so service providers don't implement it,
>>> so devices don't implement it, so... XD
>>>
>>>
>&
On 02/26/2009 07:08 PM, Iñaki Baz Castillo wrote:
> 2009/2/26 Daniel-Constantin Mierla :
>
>> However, being out there so many phones without such support, it is
>> practically unusable since service providers won't deploy different server
>> solutions for each gr
ce service providers won't deploy different
server solutions for each group of devices, so they stick to one size
fits all and that is not DNS for now.
Proper DNS support should be enforced somehow (who knows how?!?) before
anything else. At the end, DNS drives the IP worl
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Main organizers:
- Philippe Sultan, INRIA - contributor to the XMPP/Jingle support in
Asterisk, member of XMPP Standards Foundation
- Olle E. Johansson, Edvina - main SIP developer of Asterisk
- Daniel-Constantin Mierla, Asipto - co-founder Openser, developer of
XMPP/Jabber gateway
Goal:
- kick up
; It is expected to be used largely for diagnostic purposes.
>
Apart of troubleshooting, I would love to see the References header
mandatory for some cases, the first coming in my mind is redirect
situation. The new call should have the References set to call-id of the
session returni
n this specific case you can check the source IP (and port) of the sip
message and if comes from the AS then don't go back to AS.
if(src_ip==__ip_as__)
{
}
Checking the Via header is an option as well.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
__
Hello,
apologizes if the email looks too off-topic...
Last minute arrangements allowed to host one day of OpenSER Admin
Training session within VoN Fall Boston, Nov 1, 2007, course that will
cover openser and asterisk integration for basic media services. I
believe the event could bring more v
Hello,
the first one doesn't add too much overhead and complexity in server
side implementations, for a multi-process architecture.
Daniel
On 10/08/07 21:57, Vivek Gupta wrote:
> Hi,
>
> RFC4168 says about using SCTP as one of the transport protocols for carrying
> the SIP signaling. SCTP ha
thing
difficult.
Daniel
>
> -Daniel
>
> On Aug 3, 2007, at 6:45 AM, Daniel-Constantin Mierla wrote:
>
>>
>>
>> On 08/03/07 12:38, Daniel Corbe wrote:
>>> See below.
>>>
>>> -Daniel
>>>
>>> On Aug 3, 2007, at 5:04 AM, Dan
On 08/03/07 12:38, Daniel Corbe wrote:
> See below.
>
> -Daniel
>
> On Aug 3, 2007, at 5:04 AM, Daniel-Constantin Mierla wrote:
>
>>
>> On 08/02/07 20:04, Daniel Corbe wrote:
>>> Out of curiosity are you affiliated with the OpenSER project?
>> yes
g., phones)?
Cheers,
Daniel
>
> -DAniel
>
> On Aug 2, 2007, at 12:30 PM, Daniel-Constantin Mierla wrote:
>
>> Hello,
>>
>> I couldn't find much about SIP phones with SCTP support. Are you aware
>> of any?
>>
>> OpenSER introduced support f
Hello,
I couldn't find much about SIP phones with SCTP support. Are you aware
of any?
OpenSER introduced support for SCTP and for testing it was used:
[udp] [udp] openser [sctp] [sctp] openser [udp]
[udp]
But using against other devices is really the challen
Hello,
On 06/13/07 22:31, sumanth achar wrote:
> Hi,
> i wanted to know what's the use of stateless proxy.. a practical usecase
> would be helpful.
>
we use it for traffic dispatching among boxes with different
functionalities.
Daniel
> -Sumanth
>
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