://reviewboard.asterisk.org/r/4379/#comment24908
Issue I see with this file, it is going to be very complicated for a new
uses to program 16 character HEX into their SIP phones.
I'd prefer we make this some what simple for people to actually connect
phones to and demo out.
- Paul
are rerun and then gerrit eventually merges the
code into master.
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.
If people want to work together, then there are various ways of doing that,
one of which github makes incredibly straight forward.
I agree with everything everybody has said about team branches. They
are no longer needed server side for collaboration.
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Paul Belanger | PolyBeacon, Inc.
Jabber
(SIP/f...@example.org)
is a lot easier then origination a channel over ARI, creating bridges
and playing any tones needed using ARI. Easier might not be the right
work, more steps required is.
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]
{
message: Application or extension must be specified
}
Am I missing something here? or is this a bug?
You need to also pass the app parameter in this case. So Asterisk
knows where to send events.
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the associated bridge configuration?
The floor is now open :-)
I am a little confused what you are asking. You want to add some sort
of lifetime support to bridge within asterisk, using ARI?
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Github
. And since it is not in stasis, ARI cannot modify it. I think
the general idea was to build a new app_dial atop of ARI, then your
application would provide that functionality to control the L
parameter.
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don’t think the
Asterisk community has that kind of luck.
I still think it is crazy we didn't all agree finally naming the
testsuite was a big deal.
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to VoIP
UC Appliance Developers.” Go to
http://www.patton.com/company/newsrelease.asp?id=2592
Can I get one for free?
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On Fri, Oct 17, 2014 at 9:44 AM, Matthew Jordan mjor...@digium.com wrote:
On Fri, Oct 17, 2014 at 4:04 AM, Lenz Emilitri lenz.lo...@gmail.com wrote:
Marco and I will be around at Astricon and Astridevcon so would be
glad to talk about it.
l.
2014-10-16 21:10 GMT+02:00 Paul Belanger
/kickstandproject/asterisk-testsuite-temporary
[2]
https://github.com/kickstandproject/kickstandproject-ci-puppet/commit/f96fdd825e905c48d9cdf109c06c06ca8f90d0a1
[3] http://review.kickstand-project.org/#/c/839/
[4] http://www.mediawiki.org/wiki/Gerrit/git-review
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Jabber
or even feature requests.
Either way, I'll be kicking around Las Vegas is anybody wants to chat.
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() and accept a position
argument?
- Paul Belanger
On Sept. 30, 2014, 8:04 a.m., Kristian Høgh wrote:
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On Tue, Sep 23, 2014 at 1:47 PM, George Joseph
george.jos...@fairview5.com wrote:
On Tue, Sep 23, 2014 at 11:13 AM, Matthew Jordan mjor...@digium.com wrote:
On Tue, Sep 23, 2014 at 11:29 AM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Tue, Sep 23, 2014 at 11:45 AM, George Joseph
Bryant
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. The, people could do test commits / see how the system
works.
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On Sep 7, 2014 2:28 PM, Joshua Colp jc...@digium.com wrote:
Johann Steinwendtner wrote:
On 2014-09-07 17:07, Joshua Colp wrote:
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3981/
Testing
Originated a call to a UnicastRTP channel and
everyone in Las Vegas!
Matt
Is there any prices associated with hackathon? Both sites seem to be
missing this information. I'd consider them a good motivator for
people to join.
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On Thu, Sep 4, 2014 at 4:14 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Thu, Sep 4, 2014 at 3:54 PM, Matthew Jordan mjor...@digium.com wrote:
Join a worldwide community of designers, developers, and communications
technologists to to create, code, and design apps built on Asterisk
but rtp (media)
on eth1.
Diffs
-
trunk/configs/samples/sip.conf.sample 422198
trunk/channels/chan_sip.c 422198
trunk/CHANGES 422198
Diff: https://reviewboard.asterisk.org/r/3952/diff/
Testing
---
kamailio proxy with rtpengine. Multihomed asterisk system.
Thanks,
Paul
Diff: https://reviewboard.asterisk.org/r/3952/diff/
Testing
---
kamailio proxy with rtpengine. Multihomed asterisk system.
Thanks,
Paul Belanger
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On Aug. 21, 2014, 7:04 a.m., wdoekes wrote:
This changeset modifies the process by getting the hostname and then
resolving that into an IP address. On my box, I could change the IP
address by modifying my /etc/hosts file to resolve my hostname to a
different IP address.
That's
by changing /etc/hosts to not be 127.0.1.1 :)
Mark Michelson wrote:
In all seriousness though, a better idea would probably be to set the SDP
origin information per session based on the transport in use, similar to how
the connection line is set.
Paul Belanger wrote:
+1 I'd
to be some bumps in the road
however I'd rather see development efforts focused on chan_pjsip then
split with chan_sip.
If you are a digium developer, find me at Astricon, there will be a
beer in it for you.
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anything on Jira /
wiki that explained the need. I assume it is because of performance issues?
- Paul Belanger
On July 17, 2014, 12:49 p.m., opticron wrote:
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in the template
seem unnecessary and potentially break systems that depend on the default
config files.
- Paul Belanger
On May 28, 2014, 8:14 p.m., rnewton wrote:
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-sn uac 127.0.0.1) to
confirm HTTP 500 is no longer returned on answer.
[1] https://github.com/kickstandproject/payload-voice
Thanks,
Paul Belanger
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/res_stasis_answer.c 414194
Diff: https://reviewboard.asterisk.org/r/3549/diff/
Testing
---
Local ARI application (payload-voice[1]) with SIPp (sipp -sn uac 127.0.0.1) to
confirm HTTP 500 is no longer returned on answer.
[1] https://github.com/kickstandproject/payload-voice
Thanks,
Paul Belanger
is certainly going to break some
peoples boxes. Why not a deprecated warning and then removal from
trunk to give people time to react?
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onto
the wiki, people can chime in on the design side. Once people are
on-board with the technology and such, we'll fund the development and
aim to get it into Asterisk 13.
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Github: https
is outside
asterisk.
EG:
[example.org]
server = 192.168.1.1
users = *
So, thoughts? Interested in helping? Want to also fund this venture?
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On Wed, Mar 19, 2014 at 9:56 AM, Olle E. Johansson o...@edvina.net wrote:
On 19 Mar 2014, at 14:50, Paul Belanger paul.belan...@polybeacon.com wrote:
Greetings,
I wanted to ask if there is any sort of design document / work started
on having Asterisk 13 be able to subscribe to external SIP
are 488ing? According to
http://tools.ietf.org/html/rfc6337#section-2.3 it's preferred, but I'm unsure
if pjproject exposes the ability.
- Paul Belanger
On March 16, 2014, 8:36 p.m., Matt Jordan wrote:
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Ship it!
Ship It!
- Paul Belanger
On March 15, 2014, 10:02
/3349/#comment20860
space between t and + since you are here.
/trunk/channels/sip/reqresp_parser.c
https://reviewboard.asterisk.org/r/3349/#comment20861
might be worth adding a unit test for this code path. Something like tel:911
- Paul Belanger
On March 15, 2014, 12:42 p.m., Geert
On March 15, 2014, 6:46 p.m., Russell Bryant wrote:
So, after talking to pabelanger about it, this could be a bit more generic
without a lot of effort. Right now it's hard coded to do a Playback(beep)
into the call. It could pretty easily changed to just be a periodic
dialplan
On Fri, Mar 14, 2014 at 7:32 AM, Tony Mountifield t...@softins.co.uk wrote:
In article
CALLKq0S4TvEdbnCez_9soe9kEVbGyyO6_ru-_SJEHaxS=m0...@mail.gmail.com,
Paul Belanger paul.belan...@polybeacon.com wrote:
Sounds like the ulimit is at the default 1024. You need to increase it
because
ago on the list.
Interesting, I missed that discussion, can you sum it up to a few
lines while I look for the thread?
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On Fri, Mar 14, 2014 at 9:51 AM, Sean Bright sean.bri...@gmail.com wrote:
On 3/14/2014 2:41 AM, Olle E. Johansson wrote:
On 13 Mar 2014, at 22:13, Sean Bright sean.bri...@gmail.com wrote:
On 3/13/2014 4:42 PM, Paul Belanger wrote:
+1 with Dan. Comments aside on DNS functionality (I have
On Fri, Mar 14, 2014 at 10:02 AM, Shaun Ruffell sruff...@digium.com wrote:
On Fri, Mar 14, 2014 at 02:40:22PM +0100, Olle E. Johansson wrote:
On 14 Mar 2014, at 14:13, Paul Belanger paul.belan...@polybeacon.com wrote:
On Fri, Mar 14, 2014 at 3:02 AM, Olle E. Johansson o...@edvina.net wrote
not sure if Digium
had to get some support agreement or not but at the end of the day, we
are dependent on teluu's workflow process.
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expose
DNS for other channel drivers, great, but that should be a discussion
point. Either way, I'm happy Sean and Josh figure out how not to
directly ready from /etc/resolv.conf for nameservers, that was some
ugly code :D
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to see that turned around to not apply to
the new kid on the block.
+1 with Dan. Comments aside on DNS functionality (I have opinions but
sitting this one out). Any functionality should be channel agnostic.
I too am a little concern'd that statement seems to have changed.
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of question is better asked on the asterisk-users list.
Yup, next limit you'll hit is dahdi pseudo channels, which is 512.
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On Thu, Mar 6, 2014 at 5:32 PM, Damien Wedhorn v...@facts.com.au wrote:
On 07/03/14 08:21, Matthew Jordan wrote:
On Thu, Mar 6, 2014 at 3:42 PM, Damien Wedhorn v...@facts.com.au wrote:
On 07/03/14 07:29, Matthew Jordan wrote:
On Thu, Mar 6, 2014 at 3:22 PM, Paul Belanger
paul.belan
have to apply it to all
channel drives that implement said codecs allow / disallow logic, so
sip.conf, chan_ooh323.conf, gtalk.conf, h323.conf, iax.conf,
jingle.conf.
That way all our documentation / functionality is consistent among
channel drivers.
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need to do some
version checks on the AMI for the API breakage. I'm not sure what will happen
when you send asterisk a parameter it doesn't know about. But we still want
starpy to support 1.8, 11 as long as possible.
- Paul Belanger
On Feb. 25, 2014, 9:42 p.m., Scott Griepentrog wrote
?
/branches/12/rest-api/api-docs/channels.json
https://reviewboard.asterisk.org/r/3191/#comment20680
tabbing seems out of place
- Paul Belanger
On March 3, 2014, 11:51 p.m., Scott Griepentrog wrote:
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On March 1, 2014, 11:41 p.m., Paul Belanger wrote:
/branches/12/rest-api/api-docs/bridges.json, lines 50-56
https://reviewboard.asterisk.org/r/3278/diff/1/?file=54950#file54950line50
After reading Matts comments, this still doesn't make sense.
We are going to do POST
? I'll be trying
to use the new config framework and see if I can get it going.
- Paul Belanger
On Feb. 28, 2014, 4:34 p.m., Paul Belanger wrote:
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---
On Feb. 28, 2014, 1:44 a.m., Paul Belanger wrote
Diff: https://reviewboard.asterisk.org/r/3279/diff/
Testing
---
local development. Setup
[general]
queue_log = no
queue_log = yes
Queue logfiles were created.
Thanks,
Paul Belanger
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Testing
---
local development. Setup
[general]
queue_log = no
queue_log = yes
Queue logfiles were created.
Thanks,
Paul Belanger
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, creating the
bridge if it doesn't exists.
- Paul Belanger
On Feb. 28, 2014, 5:39 p.m., Scott Griepentrog wrote:
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On Feb. 28, 2014, 9:09 p.m., Paul Belanger wrote:
Woah, nice. What about the ability to do PUT /bridges/a1b2c3, creating the
bridge if it doesn't exists.
Actually, the more I think about this. The more I can see having PUT and
dropping POST, like we do in deviceStates and mailboxes
]
queue_log = no
queue_log = yes
Queue logfiles were created.
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Paul Belanger
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it.
Diffs
-
trunk/res/res_ari.c 400880
Diff: https://reviewboard.asterisk.org/r/2904/diff/
Testing
---
Local development box, with python-ari!
Thanks,
Paul Belanger
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Please add the appropriate unit tests here too.
- Paul Belanger
On Feb. 23, 2014, 11:17 a.m., wdoekes wrote:
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On Feb. 3, 2014, 9:41 p.m., Paul Belanger wrote:
So, I have some comments / concerns about the 'refactor' of chan_sip.
Basically, why are we doing it? I mean, now that chan_pjsip.c exists, I
would very much like to see actually development of chan_sip.c stop and be
focused
chan_pjsip is not feature par with chan_sip, however we should avoid
reworking it when possible.
Thoughts?
- Paul Belanger
On Feb. 3, 2014, 7:48 p.m., Corey Farrell wrote:
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suggest because you
could then easily add support for mysql and postgres
- Paul Belanger
On Jan. 31, 2014, 3:50 p.m., Matt Jordan wrote:
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as expected.
- Paul Belanger
On Jan. 27, 2014, 5:30 p.m., Russell Bryant wrote:
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and they are broken.
So, with that in mind I think we need two changesets, one that fixes the
current alembic migration paths. And the seconds which adds your logic.
Lastly, this we need an automate testcase for this, as it is obvious this code
was checked in broken from the start.
- Paul
On Jan. 22, 2014, 1:14 a.m., Paul Belanger wrote:
branches/12/contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py,
line 1
https://reviewboard.asterisk.org/r/3148/diff/1/?file=53005#file53005line1
You should be creating new alembic scripts
On Jan. 22, 2014, 1:14 a.m., Paul Belanger wrote:
branches/12/contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py,
line 1
https://reviewboard.asterisk.org/r/3148/diff/1/?file=53005#file53005line1
You should be creating new alembic scripts
.
branches/12/contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py
https://reviewboard.asterisk.org/r/3148/#comment20144
same comment as above
- Paul Belanger
On Jan. 21, 2014, 11:01 p.m., Kevin Harwell wrote
://reviewboard.asterisk.org/r/3117/#comment20013
same
- Paul Belanger
On Jan. 9, 2014, 11:38 p.m., Jonathan Rose wrote:
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see us add something to better increase out code
coverage.
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Ship It!
- Paul Belanger
On Jan. 3, 2014, 7:14
/asterisk.conf.sample
branches/12/main/asterisk.c
Please also update UPGRADE-12.txt and how we are breaking backwards
compatibility for upgrades.
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://svnview.digium.com/svn/repotools/
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On Dec. 14, 2013, 7:39 p.m., Paul Belanger wrote:
/branches/12/rest-api/api-docs/bridges.json, lines 505-514
https://reviewboard.asterisk.org/r/3070/diff/3/?file=49607#file49607line505
Why are these required, but every place else appear to be optional?
Jonathan Rose wrote
/bridges.json
https://reviewboard.asterisk.org/r/3070/#comment19860
Why are these required, but every place else appear to be optional?
- Paul Belanger
On Dec. 13, 2013, 9:40 p.m., Jonathan Rose wrote:
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On 13-12-10 10:30 AM, Pablo Carranza wrote:
Is it possible to install Asterisk 11 via the Ubuntu 12.04 repos? Is a PPA
needed? If so, which one is recommended?
I plan to do some work on it over the Christmas break. I'll likely
publish them to a PPA on launchpad.net.
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bump the AMI version, so libraries also know
of the functionality change.
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native
functionality to play numbers however, feel like libraries on top of ARI could
handle this. /me shrugs.
- Paul Belanger
On Nov. 23, 2013, 2:35 p.m., Joshua Colp wrote:
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On Nov. 23, 2013, 5:38 p.m., Paul Belanger wrote:
Wouldn't this be the same path as /dial? I know asterisk has native
functionality to play numbers however, feel like libraries on top of ARI
could handle this. /me shrugs.
Joshua Colp wrote:
To play numbers in various languages
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