PJSIP is confusing at first but after a short time its flow is pretty easy...
the part i miss most about CHAN_SIP is the 'sip show peers' in a nice compact
easy to read format.. pjsip show endpoints is not so easy.. several lines per
station / trunk..
you can use the wizard format to have all
has PJ fixed the issues they had with BLF caller ID ? I had tried it a year or
so ago and the NOTIFY messages for BLF were not sending the calling party...
I really need to be making the move in our next production version to PJ since
chan_sip is fully on its way out now..
havent tried com
What is SE? I’m not familiar with your term. I only knew of regular and LTS
On Saturday, December 18, 2021, 09:36:27 PM EST, Michael Knill
wrote:
Ok thanks Lonnie. Already disabled pre SE __
Regards
Michael Knill
On 19/12/21, 12:41 pm, "Lonnie Abelbeck" wrote:
Hi Michael,
ive been running asterisk 16 in the lab and in small installations for awhile
now without any major issues.. there was some AMI changes I had to make t oa
couple of my applications that use CLI Command functions.. its running well
on my raspberry pi 4's and on one APU2. granted this is not a
I use MT routers for site routers all the time, really bug sites I go to a
2000 or 3000 series.
For remote workers each one gets a little MT hex poe for their phone .. it just
uses an L2TP connection back to our NOC for hosted customers or to their own
site if the customer has an on prem phon
for my remote workers I use a little Mikrotik POE router, establishes an L2TP
tunnel to a Mikrotik in my NOC which then talks to my Asterisk Server..
phones work perfectly in this manner and voice is encrypted, no SIP ports open
to the public side.. handles NAT situations in people's homes p
some versions of BIOS give me a wierd "boot freeze scenerio" running Centos 6
with grub, once grub loads the bootloader and progresses on, the APU2 will
freeze solid for about 5 minutes before the kernel loads..
then the system will boot fully.. ive noticed it on APU2 only and not APU1.
(I
for whatever its worth.. from a hardware standpoint I have probably 200 APU2s
in the field and dont have then just freeze like that.. granted they arent
running astlinux, but just mentioning it from a power / hardware point of view.
mine are all running Centos 6.X and asterisk 11 or 13
I hav
ried in Asterisk the option „c“ of the „dial“ command?
Sent from a mobile device.
Michael Keuter
Am 15.06.2020 um 22:21 schrieb nedi :
Thanks , I will try tomorow.Regards nedi
Am 15.06.2020 um 22:16 schrieb The Cadillac Kid via Astlinux-users
:
hit send too quick..
this is the SIP sent bac
3:54:19 PM EDT, nedi wrote:
The phone showing message "call completed elsewhere" on the screen coming
but missed call is still there on all phones.Nedi
Am 15.06.2020 um 21:45 schrieb The Cadillac Kid via Astlinux-users
:
this was a PJ issue wasnt it? im still using Chan_sip
The phone showing message "call completed elsewhere" on the screen coming
but missed call is still there on all phones.Nedi
Am 15.06.2020 um 21:45 schrieb The Cadillac Kid via Astlinux-users
:
this was a PJ issue wasnt it? im still using Chan_sip and my snoms dont do it
if I
this was a PJ issue wasnt it? im still using Chan_sip and my snoms dont do it
if I answer at another phone, they actually show call completed elsewhere on
the screen... im using 10.1.49.X firmware on my snoms..
I dont remember having to set anything different in the config to make it work.
:
Hey Christopher,
Have you tried the GL-iNet GL-MT300N-V2 (mango) Wireless Router/VPN, $20.49 USD.
https://www.gl-inet.com/products/gl-mt300n-v2/
It supports WireGuard. An alternative to the MikroTik.
Lonnie
> On Jun 9, 2020, at 7:01 AM, The Cadillac Kid via Astlinux-users
>
what features are customers wanting that having a smart app is important? Lync
style integration? VPN has been my method for securing remote clients for
awhile.. many of them are logging into the their site anyway to work remotely,
so we integrate to their networks so the phone will just work.
ive got some of my group on softphones.. others we set up L2TP server on a
MikroTik router at the home office and then sent them and their hard phones
home with another MikroTik set as L2TP client, we use the MikroTiks with POE
and 48 volt PS so they just plug in and go with their hardphones.
Ive had this same issue.. while im not running on astlinux on my production
sites.. ive found that I needed to do one of 2 things..
anchor the media at the asterisk server, so the directmedia = no on the SIP
trunk, and in the SIP settings I needed to have my externip = my.public.ip.here
a
we have kept it simple in that regard.. we use IPSEC tunnels from our linked
sites to the Host for Hosted systems and L2TP client to the site for prem
systems where they hasve remote teleworkers.. (the remote teleworker receives
a MikroTik router ) which connects to the site..
NAT issues a
I think I started using astlinux somewhere around the 0.28 or so builds.. it
was real close to the beginning.. I had played with asterisk since about 02 or
03 around there.. my first astlinux was on a Net4801 board also.. that board
later became a MikroTik Router in my office until its flas
the big issue i have with AWS is no ability t ogain console access when needed
to get things working right. I have quite a few servers running on AWS, however
my Asterisk servers are not among them.. AWS doesnt do real well at limiting
someones Rogue instance and i had issues with latencies and
I do exactly that, i preface my tenant id to my mailbox.. 0001-1234. And
0002-1234. Since i handle vm dial in and auth in my dialplan things dont break.
I do the tenant preface for all comtexts for my stations, sip ID etc.
This is done with my cloud asterisk platform where many tenants are on on
I didnt count lightning strikes.. I have had 2 lost in lightning strikes..
MAJOR strikes.. since we often have systems connected via T1 / PRI. by way of
an Adtran TA900, the point of lightning failure (even with SNEAK) tends to be
the T1's get popped.. either the can or the pole depending
same experience on my end.. i had 3 in the field.. 2 were lost when the phone
room temp went consistently above 80 degrees for the summer months. the other
is still in service, however in a very controlled IT environment.
we havent yet lost any APU or APU2 boards and have 100s out in the fie
I’ve done it and it didn’t seem to hurt though I’d never do it in production.
I’m not sure how much data is cached by asterisk so I end up always handling
the DB through AMI Christopher
On Wednesday, July 4, 2018, 6:23:56 PM EDT, Michael Knill
wrote:
Does anyone know if there is an
and we use blue and red but not black lol
On Sunday, March 25, 2018, 3:14:37 PM EDT, Michael Knill
wrote:
Thanks all for your comments and yes Chris you make a good point.
And of course there are other memory upgradeable hardware options like the
quite fast Qotom box.
But a
I never used the DPMA with astlinux but when I used it with regular asterisk.
it seemed pretty CPU intensive esp when updates were being pushed / pulled..
dbus, avahi, mDNS were some of the things that it required to work.klougey at
best is how I found it.. im sure they ironed some of that op
another way to look at it would be are you going to out grow CPU along with ram
as opposed to outgrowing ram on its own.. in my experience as my projects grow
or get more intense, the requirements of both seem to go up over time.. if
astlinux ever needed more than 2 GB would it even run on the
I use the 4 GB boxes in my asterisk installations as we do a lot of processing
in RAMdisk for non critical functions.. fast and saves the wear on the
mSATA.the APU and APU2 boards have been a great replacement and upgrade once we
ran the Soekris to its limit.-Christopher
On Sunday, March
theres a lot of asterisk 11 production systems out there in the real world.. i
know I have a few hundred myself.. its been one of the good ones!! 11.20 has
been a good build of asterisk for me.. ive got systems with > 500 day
uptimes..
I was glad to see asterisk 13 get a longer stay.. as it
ive got about 150 APU1's and about 20 APU2's in the field... not on astlinux
but on centos 6.X and asterisk 11 or 13 depending on version and so far all are
solid.. for some reason the kernel takes a long time to boot on my APU2's I
think I need a kernel bump and that will fix it.. but once u
> exten => _NXXNXX,n,NoOp(inbound-phone-call)
> > exten => _NXXNXX,n,Set(boxnumber=${EXTEN}) ; set a variable for box
> number
> > exten => _NXXNXX,n,NoOp(${boxnumber}) ; test for variable
> > exten => _NXXNXX,n,Voicemail(${boxn
set a variable first... the issue is that ${EXTEN} changes to 'a' when you *
out... ${EXTEN} is the current extension you are workign with and you want to
go to the original dialed extension.
[inbound]
exten => _NXXNXX,1,Answer
exten => _NXXNXX,n,NoOp(inbound-phone-call)
; set a variabl
I have found hangup handlers to be MUCH more accurate than the h extension..
it seemed whe ni was using the h extension that it wouldnt trigger.. or
triggered in a different context than I expected when calls failed.-Christopher
From: Michael Keuter
To: AstLinux Users Mailing List
I do it within asterisk itself, I use custom cel events that get logged to
SQLite on the box, there's also an agi that gets run in the event I run out of
trunks. This notifies my noc, all failed trunk calls are logged specially, this
is extra important on my hosted sites as I should not fail tru
I run my asterisk DB in RAM on my regular asterisk servers.. I have an agent
that automatically backs that up every so often to set a of files.. if my
asterisk system crashes then it reloads my asterisk DB on boot.. persistence
is maintained via storing my asterisk DB as a datadump style file
I pulled out my few-years-old NUC, and it went on to HDD boot when it couldnt
get a valid filename from the TFTP.. im not using DNSMASQ. but i put the MAC
address of it in my dhcpd.conf with a option 66 that went nowhere..bogus
filename so it failed to get a boot file and just went on to boot
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