If so, please consider attending SIPit - the SIP interoperability test event
held this September in the UNH Interoperability
labs in Durham, New Hampshire, USA.
At SIPit we test SIP implementations - it’s not a certification, but
peer-to-peer tests and multiparty tests. We have a great
testbed
- and many other related classes - in-house for
companies that need a combined
workshop and training session. If you are interested in such an event, just
contact me and we'll set it up.
Looking forward to seeing you in Madrid!
Best regards,
/Olle E. Johansson
---
* Olle E Johansson - o...@edvina.net
Hi!
The reason why I am writing to you is that I am preparing for the next SIP
Masterclass and still have a few seats available for you or your team members.
The class is administered by Avanzada 7 and held in beautiful Torremolinos,
just outside Malaga, Spain. We start June 30 and end after
The next Edvina SIP Masterclass featuring Kamailio and SIP will take place the
first week of July in Malaga, Spain. One week of labs, lessons and interaction
with other people working with realtime communication will be a boost to your
career, and give you insights into how you build scalable
to contribute at least 150 USD or more please contact me
off list. The fee would be billed from my company.
Thanks for your support,
/Olle
---
* Olle E Johansson - o...@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
site at this address:
http://edvina.net/training/new-sip-masterclass/
Looking forward to meeting you in Florida!
SIP-greetings!
/Olle Johansson
--
* Olle E. Johansson - o...@edvina.net
* Kamailio SIP Masterclass Miami FL December 2012
* http://edvina.net/training
Friends,
The New SIP Masterclass will be held in Stockholm, Sweden Oct 15-19. As this is
the first run of this new and upgraded training in SIP and Kamailio, we're
offering great discounts. 20% for all students and 30% for previous students in
our classes.
This class is built for persons that
is a perfect day to negotiate pricing with me...)
Feel free to contact me to register today!
For more information, please read http://edvina.net/training/sipmasterclass/
See you in Barcelona!
Regards,
/Olle
---
* Olle E Johansson - o...@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20
25 maj 2012 kl. 16:41 skrev Kevin P. Fleming:
We'd like you all to help us welcome Rusty Newton to Digium's Asterisk
development and community support team! Rusty has been with Digium for
over five years, starting in the Technical Support department and then
moving to a sales position where
23 maj 2012 kl. 14:21 skrev Kevin P. Fleming:
While this behavior is technically not RFC3261 compliant (and I've had
discussions about it with at least one of the RFC's authors), it's quite
useful in making decisions about whether a peer has become unavailable more
quickly than would
Just for the archives:
Don't forget that you can use the SIPPEER() dialplan funciton to check the
status of the peer with qualify=on before you place the call in the dialplan.
/O
--
_
-- Bandwidth and Colocation Provided by
23 maj 2012 kl. 14:35 skrev Alex Balashov:
That comes down to whether userspace, SIP stack-level OPTIONS pings are a
good estimate of RTT. :-)
Absolutely. But it's at least an estimate better than 500 ms in most
situations. It does affect the quality of the call.
/O
--
Friends,
I've been running the Asterisk SIP Masterclass for many years now. It's time to
run the last show - partly with new material. Compared with the very first
Asterisk SIP Masterclass I would say that I've rewritten 90% of the material.
That's what happens during the class. Students ask
Hello friends,
For a few years, I've been working as an advisor and teacher to several
Asterisk platform developers. They work with AGI, AMI and the dialplan.
Now I wonder if there are people interested in such a training class? I've done
several in-house trainings like this.
I would guess
6 feb 2011 kl. 08.31 skrev Dovid Bender:
Hi,
$150.00 bounty to fix: https://issues.asterisk.org/view.php?id=14239 for
1.8.X. Would like patch for 1.6.X as well. $200.00 if accepted by mantis.
What does accepted by mantis mean?
/O
--
don't expect immediate answers
:-)
Best regards,
/Olle
---
* Olle E Johansson - o...@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
--
_
-- Bandwidth and Colocation Provided by http://www.api
23 jul 2010 kl. 10.24 skrev Olle E. Johansson:
Friends,
I need to expand my network of Asterisk developers that can help Edvina with
small and larger projects, both related to SIP and to other parts of
Asterisk. If you are interested, please send me e-mail with your current
experience
24 feb 2010 kl. 15.45 skrev Chris Bagnall:
Can someone please let me know if you have a such experience?
Also, do you have any other negative or positive comments on 1.6
Most of our clients are still on 1.4 installs - I must admit largely because
of the time taken to rewrite the numerous
While we continue discussing all possible solutions to this and build an
expanding knowledgebase, I would like to repeat myself and kindly ask everyone
that blogs, twitters, talks and teaches about Asterisk to please spread the
word and the links. Later today, there will be an official Asterisk
6 jan 2010 kl. 02.56 skrev Matt Riddell:
On 6/01/10 1:28 PM, Erik Lagerway wrote:
Do any of you know of a simple remote SIP monitor service or
application/software? Looking for something I can point towards my
various SIP servers to monitor registrations, ongoing call connectivity
and
6 jan 2010 kl. 09.31 skrev Matt Riddell:
On 6/01/10 9:14 PM, Olle E. Johansson wrote:
I would like feedback on what's missing in the AMI in order to monitor an
Asterisk platform. I've added a lot of events and some actions to AMI in
order to enhance monitoring, but new ideas are always
6 jan 2010 kl. 15.30 skrev Jared Smith:
On Wed, 2010-01-06 at 09:45 +0100, Olle E. Johansson wrote:
I'm adding manager events and storing data in a realtime database -
one record per call leg. What I'm wondering is how we should handle
call transfers and hold situations. A call that's
30 dec 2009 kl. 15.03 skrev Kevin P. Fleming:
Peter Beckman wrote:
On Wed, 30 Dec 2009, Alex Balashov wrote:
I'll start after you with this innovative new negative proposition
trend: I cannot modify the solar orbit of Earth, thanks.
Crap! I was just gonna post and ask that...
29 nov 2009 kl. 12.14 skrev Mueller, Alexander:
CS, that's my problem, exactly how you describe it:
but why can't you redirect the AMI Originate to a dialplan script that
uses the SipAddHeader?
After lots of experimenting, I now come from AMI into the dialplan by just
calling Action:
29 nov 2009 kl. 15.32 skrev Kevin P. Fleming:
Olle E. Johansson wrote:
Secondly, this mailing list is for Asterisk development - new code and
issues with the Asterisk code. Your questions would get a faster and better
answer on asterisk-users. Even if it's about development, it's about
29 nov 2009 kl. 17.47 skrev Mueller, Alexander:
29.11. 17:37:19,060 Action: Originate
Channel: SIP/2000
Context: originating
Exten: #*00123456798
Priority: 1
CallerID: 2000
Variable: Outbound_CALLERID=07615987654321
ActionID: ORIGINATE_452
Instead of
During the last year, I've had a number of inhouse training sessions covering
* Asterisk
* Asterisk and SIP integration
* Asterisk application integration - AGI and AMI
* SIP
* SIP Proxys - OpenSER/Kamalio SIP-router
I'm currently planning the training schedule for the spring of 2010 for our
focus on those
releases.
Please contact me OFF LIST if you are interested in funding this work. If a few
interested parties can fund one work-day each, so I can spend at least three
full days on this issue, I think a lot can be done to improve our RTCP support.
Thanks,
/Olle
---
* Olle E
as well as NGN-class of services for telephony and IP unified
communication, such as instant messaging, presence, integration with
social networking.
Teachers:
- Daniel-Constantin Mierla - founder and developer of Kamailio SIP
server
- Olle E. Johansson - Asterisk developer, consultant in large
4 sep 2009 kl. 19.21 skrev Andy day:
Rehan,
Asterisk is likely looking at the sip headers for IP authentication
and not
the actual IP headers. SIP headers can be spoofed, but I don't
believe they
can spoof the IP packets and still have it routed properly to this
customer
unless
5 aug 2009 kl. 12.11 skrev Faiz Rehman:
We've been having problem in finding a really reliable company to
provide us with SIP Termination. So far we've used VoiceTrading
(Premium routes)
and have problems with calls not being connected, messages not being
played, false answer, etc etc.
5 aug 2009 kl. 18.16 skrev Alex Balashov:
Olle E. Johansson wrote:
We can provide termination in remote parts of Törnskogen, Sollentuna,
Sweden... :-)
I heard the tin can and string access charges are stupendous! Would
you
like to enter into a profit-sharing agreement like no-good
11 mar 2009 kl. 20.31 skrev Trixter aka Bret McDanel:
On Wed, 2009-03-11 at 15:13 -0400, Andrew M. Lauppe wrote:
Despite of all the arguments on other things we could do, why not
increase
the level of security in Asterisk if there is a possibility to do
so?
Bottom line here, I think,
11 mar 2009 kl. 20.03 skrev Remco Barendse:
Despite of all the arguments on other things we could do, why not
increase
the level of security in Asterisk if there is a possibility to do so?
As always with Open Source, it's a matter of funding. We have worked
out two
rather detailed plans
28 maj 2007 kl. 11.36 skrev S. A. Kamran:
Hi,
I am going to visit VON Europe and would like to meet Asterisk
community there. Please email me off the list if you are visiting VON
Stockholm or you are located in or around Frankfurt, Germany.
There is an Asterisk meeting on the VON agenda.
http://www.digium.com
and of course Edvina.
Don't hesitate to contact me if you have any questions or want to
proceed with a sponsorship!
Thanks for your support,
/Olle
* Olle E Johansson - [EMAIL PROTECTED]
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
31 dec 2006 kl. 13.12 skrev Rehan Allah Wala:
i do not need an account, I need to restrict the call based on the
called party id.
Ie you own the did number 12126559343 and you buy from me 5
channels of incoming, I want to restrict the calls going from our
end to your end without any sip
31 dec 2006 kl. 07.14 skrev Rehan Allah Wala:
Rehan,
Quoting Rehan Allah Wala [EMAIL PROTECTED]:
Hello,
Has anyone implimented SIP and IAX out going channels qty
restrictions on per user bases?
You mean limiting usage on channels per user ? i.e. 5 channels
for user
x and
.
Best regards,
/Olle
---
* Olle E Johansson - [EMAIL PROTECTED]
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
___
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asterisk-biz mailing list
To UNSUBSCRIBE or update options visit
6 jul 2006 kl. 04.14 skrev Rehan AllahWala:
like oh323
http://www.voip-info.org/wiki/view/H323+Variables
Isnt there a variable for SIP
Like
${OH323_RADDR} - Contains the remote IP address/port of the
connection.
That's the very old way. We are moving towards dialplan functions
where
and
IAX_PEER functions.
/Olle
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Coming trainings: Asterisk SIP Masterclass, Chicago, Asterisk
Bootcamp at the Beach, Spain
and Asterisk Bootcamp, Boston
Just a short message from Edvina - the Asterisk Training company:
The Asterisk SIP masterclass is in Chicago, Illinois next week. It's
going to be a lot of
labs and theory focused on how to make Asterisk work in a scalable
SIP network.
This class continues on an advanced level from the
24 maj 2006 kl. 21.47 skrev Greg Boehnlein:
On Wed, 24 May 2006, Paul wrote:
So stir up some new products. I would love to see an iax2 ata that
takes
plugin fxo and fxs modules. It should also support iax2 trunking.
That
would allow more of us to support your efforts with our wallets.
:
Teacher: Olle E. Johansson, Asterisk developer and trainer.
Material: Training slides (over 300 pages), The Asterisk Quick
Reference Guide
Dates: June 12-16 (starting 10 AM Monday, ending noon friday
Options: dCAP exam friday afternoon, June 16th
Price: 2.500 Euro (ex VAT). 200 Euro (ex VAT
Just a quick note to say that we're quickly running out of seats for
the Asterisk Bootcamp, May 22-26
in Edison, New Jersey, USA.
For information about the bootcamp, please visit:
http://edvina.net/training/bootcamp.shtm
The trainers for this bootcamp will be Olle E. Johansson, Ed Guy
that needs to be fixed somehow as well. I don't know if other VoIP
channels have similar behaviour - or the Zap channel for that matter.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED] * Meet Asterisk Europe http://
www.meetasterisk.com
* Asterisk Training http://edvina.net/training
Friends,
At the end of this month, I will travel around Europe to teach
Asterisk - the one day Meet Asterisk training.
MeetAsterisk is organized by Edvina in cooperation with Digium and
Voop. In many places, local Asterisk
equipment resellers participate and show their equipment.
This is
PBX.
During the week, we will build a
business PBX configuration as well as more advanced configurations
using E1/PRI, SIP and IAX2 protocols.
You can bring your own hardware as well.
Facts about this training:
--
* Teacher: Olle E. Johansson, Asterisk
Last year, I spent a lot of time rewriting SIP transfers in chan_sip, in
order to enhance the support for attended transfers, especially in the
case where two servers where supported. This was paid for by a service
provider, who after they installed it in their production systems
decided not
trixter aka Bret McDanel wrote:
On Wed, 2006-02-01 at 11:28 +0100, Olle E Johansson wrote:
Even though it works in their environment, there is still some work to
do to finish this quite large change and make it more generic and
complete for standalone servers, as well as cleaning the source
trixter aka Bret McDanel wrote:
I think what you are saying, and please correct me if I am wrong, is
that most of what you are doing in terms of these features is already
done, and that its to work with the current chan_sip, basically just
adding features to it.
Is that correct?
Yes, the
trixter aka Bret McDanel wrote:
On Wed, 2006-02-01 at 12:38 +0100, Olle E Johansson wrote:
trixter aka Bret McDanel wrote:
I think what you are saying, and please correct me if I am wrong, is
that most of what you are doing in terms of these features is already
done, and that its to work
** Astricon Trainings in Europe - book now!
We have scheduled two trainings for Europe in the first quarter of 2006:
* The Asterisk Bootcamp - Feb 6-10 2006
The complete Asterisk one-week training including the dCAP examination.
This training includes a lot of labs and in-depth knowledge about
Matt Riddell wrote:
Will pay well for the delicate taste of Asterisk flavoured fish.
Now !! Anyone that knows help you will, or re-mail again.
/O :-)
___
Asterisk-Biz mailing list
Asterisk-Biz@lists.digium.com
** Certify your knowledge of Asterisk in Madrid!
There will be an oppurtunity to take the dCAP exam at Astricon Europe in
Madrid,
the Asterisk user's conference next week. We need registrations for dCAP
in advance,
so please sign up on the web site so we can plan the resources.
Read more about
Last chance to register for Meet Asterisk! In Stockholm on friday,
may 27th. Meet Asterisk is a one-day introduction to Asterisk for new
users and everyone interested in what Asterisk can do and can't do.
Register today on http://www.astricon.net/meetasterisk !
Regards,
/Olle
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