x27;: Invalid UTF-8 string.
> [Mar 5 11:13:29] ERROR[3526]: json.c:704 ast_json_vpack: Error building
> JSON from '{s: s, s: s}': Invalid UTF-8 string.
>
>
> This is call from H323, as I know avaya , chan_ooh323 from my side to
> another asterisk SIP chan_sip on both sides
an Asterisk 1.8 server
> for webrtc capabilities (but not any other sip). It uses the dispatcher
> module to dispatch to the underlying asterisk so you will still need to add
> the Asterisk to the dispatcher config.
>
+1 to everything here. We also do this and it works q
>
> [1]
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation
> [2]
> https://wiki.asterisk.org/wiki/display/AST/Installing+the+Asterisk+Test+Suite
> [3]
> https://wiki.asterisk.org/wiki/display/AST/Running+the+Asterisk+Test+Suite
>
It shou
://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> Hi
>
> All of that is possible and is exactly what we do, both for customer sounds
> and for call recordings. Just make sure you have resilience in your shared
> storage device.
>
> Alternatively, you could use somet
On Wed, Jan 28, 2015 at 1:37 PM, Paul Belanger
wrote:
> On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik wrote:
>> Hi
>>
>> We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for
>> queues.
>>
>> For a particular customer, when I ru
the queue show figure wrong due to a bug or am I making an incorrect
> assumption as to what it means?
>
> Thanks in advance
>
Welcome to business logic embedded into app_queue. The issue with the
queue show command rendering stats, is what timeframe are the stats
aggreg
looking,
likely, but you should be able to see anything you missed in your
testing phase.
You should be able to google Asterisk dialers to see some example that
people have done.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Githu
ut does show RTP flows to chrome, but there's no sound
> from chrome.
>
> I hope someone can intersperse the output with comments?
>
Pastebin the fill debug, you've delete an important piece of information.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeac
ecuting [h@pbx-routing:5]
> NoOp("SIP/SipAT01-0015", "sip cause = ") in new stack
> [Oct 30 14:48:03] -- Executing [h@pbx-routing:6]
> NoOp("SIP/SipAT01-0015", "sip cause = ") in new stack
>
>
> Can anyone tell me how this should
g anything like:
# asterisk -rx 'core show channels'
via an external process?
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--
__
t cannot use same username if that username is connected
> with the other user?
>
>
Since what you describe is a valid for SIP, you'll have to drop the
packets at the network level (firewall). Or use the ACL system in
asterisk to restrict it.
--
Paul Belanger | PolyBeacon, Inc
t think that
> I'm brute forcing.
>
> Just a question to check if there's any chance I could ask Asterisk not to
> register when I reset. Or is there any other possible solution for this?
>
No, only reload after your ITSP brute force timer has expired.
--
Paul Belanger |
e start, it is pretty stable. And this is the primary reason people
are using rtpengine with asterisk to start. So, in your setup listed
above, rtpengine is not needed, since newer versions of asterisk
support both. Adding it in will just complicate your setup.
--
Paul Belanger | PolyBeacon, Inc.
J
And, for us, we
keep RTP/SAVPF outside of asterisk since support for it has been
recently added. I also believe there are some open issue with dtls +
srtp too.
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Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: p
to do is pretty complicated, it took me about 2
weeks to get everything setup properly. There is good information[1]
on the web, you just need to google for it.
[1] http://www.slideshare.net/crocodilertc/webrtc-websockets
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybe
r words should I get the line bellow or something else ?
> libpjsua.so (libc6) => /usr/lib/libpjsua.so
>
You will likely need to pass the pjproject directory to configure.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github
> versions in rtp profile handling.
>
> It would be good to know how to handle this scenario in the new versions as
> well, I'll probably need to upgrade ahead anyway.
>
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Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pa
is needed for asterisk (in fact 1.8 has no
support for RTP/SAVPF) so we rewrite SDP on 488. Then setup a
kamailio peer and away you go.
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Paul Belanger | PolyBeacon, Inc.
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Github: https://github.com/pabelanger | Twitter: https:/
can reduce the window of opportunity for that by several seconds.
>
> It's only happened once in 2 years that I know of, so may not be worth
> worrying about.
>
AMI will raise the AgentCalled[1] event.
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_AgentCalled
On Thu, Apr 17, 2014 at 7:25 AM, binary dreamer
wrote:
> hi. I would not do that due to network issues.
> My approach is to record everything locally and every hour or so to move
> everything to a storage.
>
+1 save yourself the headache and do this.
--
Paul Belanger | PolyBeacon,
he
could leverage snapshots in VM ware for the purpose or migrating or
back ups. I don't think it is a waste per say, just different
requirements.
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Github: https://github.com/pabelanger | Twitt
27; before handing off to Asterisk -- easier to
> implement, easier to maintain, no legal BS to consider.
>
> Or can you express your creativity by fiddling with ASTERISK_PROMPT?
>
If you really want to do it:
1) create a wrapper to asterisk -r
2) pipe the welcome message to /dev/null
3) ?
ge you want to use. We used starpy for a
while, but ended up rewriting our own version. Currently we're
connecting AMI to a message bus and passing events across the bus.
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Paul Belanger | PolyBeacon, Inc.
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Github: https:
Madsen on this and he's updating the site now. Currently
only the 3rd edition is published online.
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en a user changes their password, secret.conf gets
updated not voicemail.conf.
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Github: https://github.com/pabelanger | Twitter: https://twitter.com/p
> bandwidth), but I'd lose some functionality and have to re-write parts of my
> application.
>
> Any clues of what limit I'm hitting and how to increase it?
>
DAHDI has a pseudo channel limit of 512, some
Generator: AppVoicemail
> ;! Creation Date: Thu Mar 20 06:48:16 2014
> ;!
>
>
> i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are not
> using realtime.
> anyway to prevent AppVoicemail ro auto generate files?
>
passwordlocation = spooldir
Read voicem
ss?
>
What sort of channel count are you looking for? We did some load
testing recently and found less people in a bridge is better then
more. Audio source location didn't really matter much.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
OpenStack. So you could offer your customers a self-managed, redundant
> Asterisk cloud or something like that. :)
>
> In theory, this combination should give you a 100% redundant, auto-healing,
> auto-scaling VoIP setup. :)
>
+1 to this post. A lot of good information here.
primary box goes live.
>
Correct, in this case para-virt is not the way to go. You'll want to
use a virtualization platform that does support multi-hardware with
live migration support.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenod
On Thu, Feb 27, 2014 at 10:55 PM, Darryl Moore wrote:
>
> On Feb 27, 2014 10:02 PM, "Paul Belanger"
> wrote:
>>
>> >
>> No such thing as 'free open source g729 license', if you actually read the
>> site:
>>
>
> There is regardin
s reluctant to bring this topic up yet again , and yes I did
> google around and read the different material on the subject however,
> I am still in need of some definitive answers.
>
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pab
;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285
>
> Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
>
> From: "Haley, Scott"
> ;tag=8066eb6f589ce3124b652973b4b00
>
> To: ;tag=as06e2e068
>
> Call-ID: 8066eb
will fail to establish because lack of
codecs. If you offer a both g729 and ulaw, then ulaw will be used.
--
Paul Belanger | PolyBeacon, Inc.
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Github: https://github.c
What you describe is more of a Linux support issue then specific to
Asterisk. Depending on your OS, will dictate how to change your
gateway.
check /etc/network/inferfaces if you are ubuntu / debian.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freeno
On Thu, Feb 13, 2014 at 1:04 AM, George Joseph
wrote:
> On Wed, Feb 12, 2014 at 6:26 PM, Paul Belanger
> wrote:
>>
>> On Wed, Feb 12, 2014 at 12:50 PM, Olivier wrote:
>> > Hello,
>> >
>> > How does extensions.lua compares to extensions.conf or exte
ing AGI if you want to leverage redis or memcached.
--
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Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
--
__
ty for Asterisk. If you are using
SIP, you want to REINVITE media away from your core Asterisk box. I
suggest picking up the book[1] and reading the chapter on connecting
multiple Asterisk boxes together.
[1] http://www.asteriskdocs.org/
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@
:5] Wait("SIP/abcde-0016", "5") in
> new stack
> -- Executing [12345678912@from-sip:6] Dial("SIP/abcde-0016",
> "SIP/123&SIP/456,30,oxX") in new stack
> == Using SIP RTP CoS mark 5
> == Using SIP RTP CoS mark 5
>
,
>
> When we have bindport = 172.x.x.14 then "netstat -udpln" shows the
> following. When bindport is 0.0.0.0 then netstat shows it listening on
> 0.0.0.0 as you'd expect.
>
> udp0 0 172.x.x.14:50600.0.0.0:*
> 18114/asterisk
>
>
> --
On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger
wrote:
> On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
> wrote:
>> Hi Paul,
>>
>> Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
>> and arriving at the Asterisk server. This is why i
On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
wrote:
> Hi Paul,
>
> Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
> and arriving at the Asterisk server. This is why it's a mystery that
> Asterisk doesn't see the call coming in. We tried removing the firewall (so
ch hop across your network. If are
Asterisk is not getting anything, either it is not receiving anything
(check transmit side) or the firewall is dropping it.
--
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Github: https://github.com/pabelange
erver's real address
> 192.z.z.z is the calling phone's LAN address
>
Sounds like a routing problem opposed to an application issue. You'll
have to fire up tcpdump on Kamailio and see what happens to the
packet. The look at the local routing tables to see where it is
getting
FRAMEWORK is an option selectable under the "Compiler Flags -
Development" menu in menuselect.
./configure --enable-dev-mode
--
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Github: https://github.com/pabelanger | Twitter:
https
your kind response!
Save yourself time / energy and insist using SIP. If your ITSP cannot
accommodate your request, thank them and look for another provider.
H323 is Asterisk is basically dead, sure there is a module, sure it
might compile, but you'll be going down the path of zero help.
hts ?
>
> Regards
>
I basically had the same issue as you for one of my sites. I tried
everything under the sun to figure it out, change cables, loop back
test, change out hardware, clocking, etc.
In the end I had to upgrade dahdi to 2.7+ and the issue
if you want anybody to call you, you need to leave it open to
the public. Meaning, you can't really secure it. Obviously, don't
have any outbound trunks configured on the box so that the only
location some could dial would be your extension.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: p
60
> negative_connection_cache => 600
>
>
> -- /etc/asterisk/cdr_adaptive_odbc.conf lists below:
> [cdr]
> connection=asterisk
> table=cdr
> alias start => calldate
> alias phoneno => phoneno
> alias userid => userid
> alias callerid => c
max,
and asterisk
disconnects the call. We have no idea why this is happening. SIP and RTP is
flowing as
expected.
Your help is greatly appreciated,
Nick.
Show us the problem, give us a SIP trace[1].
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Paul
der adding the OTHER management/control protocols to
this
list: ARI, and the ExternalIVR interface.
If not, it might be instructive to learn why!
Would also like to see this update to include ARI. We talked a little
about it at astridevcon, and I think it is likely an oversight.
--
Paul Belang
are saying?
Options 1 - log the agent out, they don't get the next call.
Option 2 - Set up weights for your agents, as answer a new call,
increment then up so they don't get the next.
Either way, I see issues with the setup. Best ways is to rethink your
queue strategy and stop using
footing on performance. I don´t mean another
slow cygwin port, I man a native Asterisk for windows. In fact, I
would invest on the project if somebody wants to do it.
Do you just sit around and think shit up to blame Digium all day?
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan
gt; _X,1,NoOp(Digit entered during prompt)
exten => _X,2,Goto(project,s,1)
Then you have a DTMF issue, Background will allow DTMF to interrupt the
prompts.
--
Paul Belanger | PolyBeacon, Inc.
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Github: https://github.com/pabelanger |
there is some trancoding when using voicemail...
How can I find out if there is trancoding ??
Maybe explain what your dialplan is doing. Are you making system calls
to a database or AGI?
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github
to(project,s,1)
exten => i,1,Playback(${sounds_path}error)
exten => i,n,goto(project,s,1)
my problem when the customor call the number 600 and press 1 in order to go
to the project menu he must wait all the speech music1 music2 and music 3
if there is any way to go to project menu during the spee
700 tps_processing_function started at [ 468]
taskprocessor.c ast_taskprocessor_get()
55 threads listed.
First thing, prune your Asterisk configuration and don't load any
modules you don't need to use. Are you really using chan_mgcp,
chan_skinny, res_calender, etc.
--
Paul Belanger
Then make an educated guess about what is
happening.
--
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Github: https://github.com/pabelanger | Twitter:
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On 13-11-13 10:20 AM, Jonas Kellens wrote:
Hello,
can I use include-statements in the calendar.conf configuration file ?
You _should_ be able to use it will every .conf file, otherwise it is a bug.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger
ver, solve the issue at the source. Spend the money for a UPS at
each desktop, convert your phones to PoE and install a UPS in your
server room.
--
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Github: https://github.com/pabelanger | Twitt
d and then breaking
all of those down by customer as we run a multi tenanted set up.
SNMP would give us totals but I don't think it would do the breakdown by
customer.
You should avoid using the CLI to access that information. You'd likely
getter better results using AMI or CEL.
--
Paul B
need to change to a different one.
Both flowroute.com and voip.ms work well for me (no affiliation). Or
maybe your Internet link sucks and you need to change your ISP.
^ this
Like others said, you really need to drill down and find out where your
audio issues are. Local is easy to do, since you c
nformation.
[1] https://wiki.asterisk.org/wiki/display/AST/AMI+1.4+Specification
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--
On 13-10-21 10:39 PM, bilal ghayyad wrote:
Hello;
I am looking for calls recording solution to do recording based on the network
traffic .. The solution to be competitive and appreciate if it is open source
.. Any suggested one?
http://www.orecx.com/
--
Paul Belanger | PolyBeacon, Inc
, you lost the connection. Open the
connection again and profit.
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sk.
Don't worry about it, I'll step up and pay for the patch. No need for
you to waste your profits on something this.
--
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https://tw
scripting languages?
Eliminating translation is difficult. How do you know you were
successful? Do 'module show like codec_' and 'module show like format_'
show anything unexpected?
Also drop Apache and Database from your PBX.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.b
OSUB X routine, you are likely blocking the
thread in Asterisk, which is causing your autoservice errors (and yes,
they are real errors) which increases the CPU on asterisk.
--
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Github: https://g
something like iotop, netstat and see what your system is doing.
I doubt this is a CPU issue.
--
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Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
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then record the other side.
Then, comparing the two files via Aqua, you get your MOS score.
If the score was less then x, you knew asterisk was hitting a
performance limit. Track that over time and concurrent calls, you have
your metrics.
[1] http://www.sevana.fi/aqua.php
--
Paul Bel
On 13-05-10 02:45 PM, James Mortensen wrote:
I was wondering if anyone knows if Asterisk 12 will be supporting the OPUS
codec, which is part of the WebRTC standard as the default codec.
Doubt it.
--
Paul Belanger | PolyBeacon, Inc.
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binar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Paul Belanger | PolyBeacon, Inc.
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Github:
ncrease whatever (i.e., allocated
memory, -p value)
before addressing hardware resources?
Your help is greatly appreciated,
Nick.
You failed to say what happens when 92 channels are created. Show us
your errors.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com |
pi.com,60,rS(1200))
do:
INVITE sip:lo...@skype.ippi.com@ippi.fr SIP/2.0
I studied the source code and found no ways to implement it :(
Dmitriy.
How about:
exten =>22,n,Dial(SIP/skype.ippi.com!lo...@skype.ippi.com,60,rS(1200))
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@pol
t asterisk as tts server.
Amit--
Asterisk is not a TTS server, it is a PBX. I'm sure you could hack
stuff together, but you'd be better off to use external services for TTS.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github
/Events from 50 agents?
You don't want to use PHP for your daemon, change to another scripting
language (EG: python).
--
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Github: https://github.com/pabelanger | Twitter:
https://twitte
time interface, and replace some of
the existing toolsets. Sadly, documentation is weak, and I don't
suspect it gets much love in production.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger |
Anybody using Apache to proxy HTTP traffic to Asterisk HTTP? I got a
request from a developer to add some CORS headers[1], for an
application we are writing, and wanted to see if anybody else has had
success.
[1] http://enable-cors.org/
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan
great, I was mostly trying to
understand why you choose to rewrite logrotate :)
--
Paul Belanger | PolyBeacon, Inc.
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Github: https://github.com/pabelanger | Twitter:
On 12-12-04 10:02 AM, Danny Nicholas wrote:
IIRC log rotate only rolls the files in /var/log/asterisk, not
/var/log/asterisk/cdr-csv
You need to configure logroate with the path and filename.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode
-cdr-rollover.
Why not use logroate?
$ man logrotate
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Paul Belanger | PolyBeacon, Inc.
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Github: https://github.com/pabelanger | Twitter:
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On 12-11-17 06:23 PM, Mitch Claborn wrote:
Is there an Asterisk repository for Ubuntu that has recent versions
(e.g. 11)? The standard Ubuntu repository for Ubuntu 12.04 is stick at
1.8.
None that I know of.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC
On 12-11-08 01:41 AM, martin f krafft wrote:
also sprach Paul Belanger [2012.11.07.2340
+0100]:
What is your point of pain? Right now we do most of the
configuration, provisioning, and system management outside of
asterisk.
My systems are already managed automatically, thankfully no longer
automated tool (chef / puppet). This will help you
get on the right path.
[1] https://github.com/kickstandproject/astricon-2012-presentation
[2] http://goo.gl/T8lJR
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelang
ject/asterisk/tree/master/debian/ast_config
[2]
https://github.com/kickstandproject/astricon-2012-presentation/tree/master/debian
[3]
https://github.com/kickstandproject/puppet-modules/tree/master/modules/asterisk/manifests
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com |
On 12-09-26 11:12 AM, motty.cruz wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: Wednesday, September 26, 2012 7:52 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk
r cannot
be reached because the interface to the destination is not functioning
correctly. The term "not functioning correctly" indicates that a signal
message was unable to be delivered to the remote party; e.g., a physical
layer or data link layer failure at the remote party or user e
ful and unsuccessful
write is the above line from the message log. The tcpdump looks the same.
Permissions issues?
If you switched to FTP or HTTP does it work?
--
Paul Belanger | PolyBeacon, Inc.
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Github: https://github.com/pa
repo, I'll share the link when finished
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Paul Belanger | PolyBeacon, Inc.
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On 12-08-28 10:14 PM, Chris Nighswonger wrote:
Are there deb packages available for Asterisk 10 or for 11 beta?
None.
Well, the Debian VoIP team has an experimental[1] package for Asterisk 10.
[1]
http://anonscm.debian.org/viewvc/pkg-voip/asterisk/branches/experimental/
--
Paul Belanger
value in the
old mantis data, could we not hand it over for somebody else to manage?
I'm sure we could find somebody to donate the bandwidth.
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Paul Belanger | PolyBeacon, Inc.
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Github: https://github.com/pabelanger | Twitt
w.
Are you saying 1.8 is EOL?
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Paul Belanger | PolyBeacon, Inc.
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Github: https://github.com/pabelanger | Twitter:
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--
_
-- Bandwidth a
On 12-08-22 02:04 PM, Giuseppe Longo wrote:
Just a little questions, what's the difference between asterisk 1.8
and asterisk 11?
Not a little answer[1].
[1] http://svnview.digium.com/svn/asterisk/branches/11/CHANGES?view=markup
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.
ot;core restart graceful" in an
automated way?
Monitor the events on the AMI, you should see the following:
Event: Shutdown
Privilege: system,all
SequenceNumber: 0
File: asterisk.c
Line: 1773
Func: really_quit
Shutdown: Cleanly
Restart: True
Then you can build out your monitoring too
tion:
http://svnview.digium.com/svn/asterisk/team/oej/sip-compliance/asterisk-sip.txt?view=markup
Interesting, never knew this existed. I think it would be worth the
time and effort to get this merged into trunk or into the wiki. A great
piece of documentation.
--
Paul Belanger | PolyBeacon, Inc.
Jabber:
t way, you can then breakout each mailbox into separate
config files with include statements.
[1] http://svnview.digium.com/svn/asterisk?revision=225406&view=revision
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://g
aster/debian/ast_config
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter:
https://twitter.com/pabelanger
--
_
-- Bandwidt
work is done by Tzafrir.
Is anyone officially working on this particular problem already? I was
tempted to have a closer look at it, but don't want to duplicate an
effort that is already underway elsewhere.
Best to check JIRA and see. Actually, does the issue even exist in JIRA?
--
Paul Belang
al changes should go into 1.8.13.2 for Debian?
We don't need to release a 1.8.13.2 release of Asterisk. Once the issue
has been fixed in the 1.8 release branch, it would just be back-ported
into a Debian patch for the package.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.c
might want to reach out to #asterisk-dev
or asterisk-dev mailing list.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter:
https://twitter.com/pabelanger
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