On Oct 5, 2016 4:19 PM, "Rodrigo Ramírez Norambuena"
wrote:
>
> On Tue, 2016-10-04 at 16:45 -0600, George Joseph wrote:
> > Several folks have requested the ability to use the same schema for
> > their cdr, config and voicemail tables. Alembic currently has an
> > issue
On Tue, 2016-10-04 at 16:45 -0600, George Joseph wrote:
> Several folks have requested the ability to use the same schema for
> their cdr, config and voicemail tables. Alembic currently has an
> issue with this because the default name of the table alembic uses to
> track which revision the
On Tue, 2016-10-04 at 17:09 -0500, Matt Fredrickson wrote:
> Hey all,
Hello!,
> Welcome back to all of you who attended AstriDevCon. Thanks so much
> for all of you that attended and gave so much of your time to be able
> to contribute.
>
I missed it :(, maybe can be there the next year.
>
Hi,
why we need to specify the public (external) IP address when asterisk is
behind NAT ?
With stun, asterisk should be able to get this info by himself.
In bash for example we can get the public IP contacting the stun server:
echo -en '\x00\x01\x00\x08\xc0\x0c\xee\x42\x7c\x20\x25\xa3\x3f\x0f\
Dan,
Sure. Please see this discussion:
http://lists.digium.com/pipermail/asterisk-dev/2015-October/075122.html
I know some of that has been addressed already, but it was a show-stopper for
me at the time.
Michael
On Wednesday, October 05, 2016 05:23:27 PM Dan Jenkins wrote:
> On Wed, Oct 5,
On Wed, Oct 5, 2016 at 4:04 PM, Michael Ulitskiy
wrote:
> I am in the same situation. All my systems are business-critical and I'm
>
> yet to see a convincing argument to spend a lot of man power to migrate
> the systems.
>
> Yes, pjsip supposed to be more stable, but
I am in the same situation. All my systems are business-critical and I'm
yet to see a convincing argument to spend a lot of man power to migrate the
systems.
Yes, pjsip supposed to be more stable, but chan_sip in asterisk 11 (with a few
custom
patches) has been very stable for me. Yes, pjsip may
I don't have a preference in this fight. I am like Switzerland. Full
disclosure for those unaware I am one of the FreePBX developers. We support
both stacks individually and together. The project made a decision (good,
bad or otherwise) to push PJSIP as the default stack. This only affects new
+1
please respect Digiums reasons or dont use binary codec
Dne 05/10/2016 v 11:05 Tzafrir Cohen napsal(a):
On Fri, Sep 30, 2016 at 08:57:47AM +0100, Lefteris Zafiris wrote:
On Thu, 29 Sep 2016, at 18:10, George Joseph wrote:
The lawyers made us. ;-)
In order to lessen the risk of future
Part of what is holding me back is all my systems are production critical
for businesses. I need a business case for real world improvements pjsip
has over chan_sip if I'm going to risk downtime and issues for hundreds of
users.
I'm currently running certified Asterisks 11 on all my installs. In
Le 2016-10-05 à 05:25, Dan Jenkins a écrit :
[...]
Three potential working groups:
PJSIP migration
Documentation
ARI
Dan
Hello,
Yes, good point.
Maybe a group like "scaling asterisk" will be interesting?
Sylvain
--
_
Having worked in IETF working groups, I have to say that this is a move in
the right direction.
The questions of how will groups be created, defined, and when will
conclusion be declared (if the group does not petter out) still need to be
defined. But there are cases that are good examples.
So
James,
You missed a few points:
1. There needs to be a move in the training materials, and DCAP exam away
from (the soon to be depriciated) version 11 and move into versions that
support PJSip - familiarity will breed use.
2. One suggestion to do this was to declare that Chan_Sip would be
I have done exactly the same way from chan_sip to res_pjsip. It was a
great stress!
I talked a lot with Joshua Colp. Big thanks to him for the help.
But even now in the res_pjsip much of the functionality is unavailable.
I wrote a mail to the mailing lists, created issues.
Much of the
Hi
I have spent over a year migrating from chan_sip (1.8) to chan_pjsip (13) and
it has been stressful.
However, there is light at the end of the tunnel. When first migrating Asterisk
would crash around 20 times a day or more. However, by investing time and money
into resolving the
Hi!
From my perspective I know that maintaining a SIP stack requires *A LOT* of
effort, so I understand that a project can’t maintain two of them.
I suggest that a working group is created for the transition and that the first
task is to compare the functionality.
Last time I checked the
On Tue, Oct 4, 2016 at 11:09 PM, Matt Fredrickson
wrote:
> Hey all,
>
> Welcome back to all of you who attended AstriDevCon. Thanks so much
> for all of you that attended and gave so much of your time to be able
> to contribute.
>
> One of the ideas proposed in AstriDevCon
If chan_sip really has to go, then perhaps the place to start would be a
marketing effort?
* Why is PJSIP better?
* What does PJSIP do that chan_sip does not?
* Is there anything in chan_sip that PJSIP does not do, and how do you
solve that?
...
This can then be followed with bribery
* What
On Wed, Oct 5, 2016 at 8:21 AM, Leandro Dardini wrote:
> Your analysis of the chan_sip/PJSIP is really great and I agree with you.
> Being a grey haired tech, I can check what drives similar changes in the
> latest 20 years. We moved from Netware networks to TCP/IP, we moved
On Fri, Sep 30, 2016 at 08:57:47AM +0100, Lefteris Zafiris wrote:
> On Thu, 29 Sep 2016, at 18:10, George Joseph wrote:
> > The lawyers made us. ;-)
> > In order to lessen the risk of future legal action, the codec reports
> > anonymous stats to Digium once per day that contain the maximum
> >
Your analysis of the chan_sip/PJSIP is really great and I agree with you.
Being a grey haired tech, I can check what drives similar changes in the
latest 20 years. We moved from Netware networks to TCP/IP, we moved from
Windows 3.11 to Windows 95, we moved from IE to Chrome... in all these past
21 matches
Mail list logo