On Wed, Sep 13, 2023 at 5:20 PM Jerry Geis wrote:
> >An issue[1] was already created by asterisk at phreaknet.org and they
> also put
> >a fix up for review and inclusion[2].
>
> >[1] https://github.com/asterisk/asterisk/issues/308
> >[2] https://github.
>An issue[1] was already created by asterisk at phreaknet.org and they also
put
>a fix up for review and inclusion[2].
>[1] https://github.com/asterisk/asterisk/issues/308
>[2] https://github.com/asterisk/asterisk/pull/309
The change "seems" to be working.
Will test more tomorrow - had to
I have found that I can add the restart of asterisk (killall -9 asterisk)
to the h extension- BOY is that UGLY.
chan_console is not a testing device - how can we get this nasty bug fixed ?
Jerry
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> After a hung call, can you run core restart now from the asterisk console?
Doing a "killall -9 asterisk" is the only thing that works
I tried killall asterisk - does not free up the channel
the asterisk "core restart now" takes like a good 20 seconds to return but
does work.
The issue is I
>Using system() you could issue a asterisk -rx 'core restart now'
So I tried this
exten => s,1,Playback(beep)
exten => s,n,Dial(Console/default,20,g)
exten => s,n,Hangup
exten => s,n,System(asterisk -rx 'core restart now')
But it does not continue. Last thing I see is "Exited non zero"
so its
I have noticed that once my message speaks - the server thinks its done and
HUNGUP,
the endpoint STILL thinks the channel is active - the last message says
"Rx: BYE" on sip show channels
I tried ADDING to Dial() ,20,g and then had a Hangup after teh dial.
Its NOT getting there to hangup.
Jerry
--
Is there a dial plan call that can "exit asterisk" or completely reload
everything - killall active calls and start over ?
seems the console/dummy (chan_console) is holding some resource. How do I
just "exit" and start over after call came in ?
Thanks
Jerry
--
So I have done through chan_console.c and searched for console_pct_lock() -
every one - has a matching console_pvt_unlock()
How is the deadlock occurring ?
jerry
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>
>
> Not sure if this is the same thing you're seeing, but chan_console
> currently has a known deadlock issue that has not been resolved:
> https://issues-archive.asterisk.org/ASTERISK-30481
> It's quite easy to reproduce, so it may be the case that the module is
> currently unusable.
>
Well
Some progress to report:
I had to run asterisk as the user logged in - actually not even that. I
could not "su user -c " to that user - I had to run it as that user.
Then I did a test and got audio! Great...
But when I do a second test. Asterisk HANGS on ChanIsAvail()
Then I thought lets SKIP
ok switching to "Console/default" does show the text
--- <("<) --- Call to device 'default' on console from 'default'
<2564286000> --- (>")> ---
--- <("<) --- Auto-answered --- (>")> ---
However I don't hear any audio.
Thanks
Jerry
--
I found "console list available"
===
=== -
=== Device Name: default
=== ---> Default Input Device
=== ---> Default Output Device
=== -
===
===
Joshua
Asterisk 18.14.0 with chan_alsa and Console/dsp works.
This does not work in 18.18.0 with chan_console enabled.
I am on Ubuntu 20.04 LTS.
Is there a howto for the new chan_console ?
how can I get this working again ?
I am trying to just play audio on pulse audio.
Thanks,
Jerry
--
I am trying to get audio to play on Pulse - so just the monitor basically.
I have tried Console/dsp, Console/dmix, Console/pulse, and a couple others.
The error is always the same "console_request: Console device 'dmix' not
found.
What is the correct "Console/" to play on pulse for UBuntu
> What is the device that you're connecting to?
I am trying to just play on PulseAudio actually.
This used to work - I have just recently updated to 18.18.0, so I'm puzzled.
Jerry
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This is hte error I get for Console/dsp or console/dsp
ERROR[230711][C-0001]: chan_console.c:477 console_request: Console
device 'dsp' not found
Jerry
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> I don't use it; just figured I'd try to help
Thanks Doug...
So then for the list - I have chan_console working now
But I am trying Console/dsp and Console/ALSA and both give an error about
not found.
What have I missed ?
Thanks
Jerry
--
>
>
>
> Oh that is a good one - I thought I did - but apparently not. menuconfig
> now shows "*"
>
> So is chan_alsa going away ? What is it being replaced with?
>
> thank you!
>
> Jerry
>
hi Doug - so what device do you use? I am getting and error for Console/dsp
exten =>
>
>
> Just to verify that you did rerun configure after installing the libraries?
>
> Doug
>
Oh that is a good one - I thought I did - but apparently not. menuconfig
now shows "*"
So is chan_alsa going away ? What is it being replaced with?
thank you!
Jerry
--
> portaudio19-dev
Thanks doug - I did that - still showing XXX for chan_console
libportaudio2/focal,now 19.6.0-1build1 amd64 [installed]
libportaudiocpp0/focal,now 19.6.0-1build1 amd64 [installed,automatic]
portaudio19-dev/focal,now 19.6.0-1build1 amd64 [installed]
Jerry
--
I am still using chan_console.
I compiled 18.18.0 and chan_console is not built.
I am using ubuntu 20.04.6 LTS
make menuselect says XXX chan_consoel and it needs "portaudio"
What do I do next ?
Also menuconfig is saying XXX on Also - what alsa library is needed ?
Thanks
jerry
--
> The "timing test" CLI command will state it. Does the VM have guaranteed
resources?
timing test
Attempting to test a timer with 50 ticks per second.
Using the 'timerfd' timing module for this test.
It has been 1000 milliseconds, and we got 50 timer ticks
Jerry
--
>
>
> What is a one-way conf?
>
>
> >>* 60+ devices and packets choppy or dropping audio.
> *>
> How have you determined the packets choppy/dropping audio?
>
>
> >>* The CPU is decent at Intel(R) Xeon(R) CPU E3-1240 v5 @ 3.50GHz
> *>>* What else might I tweak to get this working without audio
On Mon, Aug 21, 2023 at 10:23 AM Jerry Geis wrote:
> I am using asterisk 18.14.0 and chan_sip.
> confbridge has dsp_drop_silence=yes
> The conf joins all the endpoints in a one-way conf.
>
> 60+ devices and packets choppy or dropping audio.
>
> The CPU is decent at Intel(R
I am using asterisk 18.14.0 and chan_sip.
confbridge has dsp_drop_silence=yes
The conf joins all the endpoints in a one-way conf.
60+ devices and packets choppy or dropping audio.
The CPU is decent at Intel(R) Xeon(R) CPU E3-1240 v5 @ 3.50GHz
What else might I tweak to get this working without
On Thu, Jul 20, 2023 at 10:24 AM Jerry Geis wrote:
> I have a hosted server.
> I have TWO different locations what have phones. Chicago and Indiana
> If I send audio direct from server to Chicago I hear it - same with
> indiana.
> But if indiana calls chicago - NO A
I have a hosted server.
I have TWO different locations what have phones. Chicago and Indiana
If I send audio direct from server to Chicago I hear it - same with indiana.
But if indiana calls chicago - NO AUDIO.
I see this in the CLI
-- Channel SIP/63009-0013 joined 'simple_bridge'
I have a cloud server...
I have a phone in Chicago
I have a phone in Indiana.
Both are registered to the cloud server - using chan_sip and Asterisk
18.18.0
I can send a pre-recorded message to Chicago it auto answers and hear audio.
I can do the same to the phone in indiana.
however - when i
I have 4 devices that I connect here local and there is no issue.
I have those same 4 devices connecting from another location across the
internet.
They all boot up, connect and register I can send audio to them and they
play.
- then at times they show UNREACHABLE and I can no longer send audio.
On Fri, Mar 10, 2023 at 10:04 AM Jerry Geis wrote:
>
>
> On Fri, Mar 10, 2023 at 9:49 AM Jerry Geis wrote:
>
>>
>>
>> On Thu, Mar 9, 2023 at 9:42 PM Jerry Geis wrote:
>>
>>> I have a SIP trunk - calls going out work fine.
>>>
>>> T
On Fri, Mar 10, 2023 at 9:49 AM Jerry Geis wrote:
>
>
> On Thu, Mar 9, 2023 at 9:42 PM Jerry Geis wrote:
>
>> I have a SIP trunk - calls going out work fine.
>>
>> Trying to setup an incoming call with a DNIS
>>
>> When I dial the number - I see nothing
On Thu, Mar 9, 2023 at 9:42 PM Jerry Geis wrote:
> I have a SIP trunk - calls going out work fine.
>
> Trying to setup an incoming call with a DNIS
>
> When I dial the number - I see nothing on the CLI.
> The person says the server is returning 401
>
> How do I debug tha
I have a SIP trunk - calls going out work fine.
Trying to setup an incoming call with a DNIS
When I dial the number - I see nothing on the CLI.
The person says the server is returning 401
How do I debug that. Using asterisk 18.8.0
Thanks
Jerry
--
I see this in my logs:
[Feb 9 15:25:27] NOTICE[2959153][C-06c8] chan_sip.c: Failed to
authenticate device ;tag=1019177874 for INVITE, code = -1
[Feb 9 15:29:44] NOTICE[2959153][C-06cd] chan_sip.c: Failed to
authenticate device ;tag=1301847080 for INVITE, code = -1
[Feb 9 15:33:56]
I am compiling 3.2.0 on Ubuntu 22.04
checking for error_at_line... yes
checking for GNU libc compatible malloc... yes
./configure: line 20019: syntax error near unexpected token `LIBUSBX,'
./configure: line 20019: ` PKG_CHECK_MODULES(LIBUSBX, libusb-1.0,'
make: *** [Makefile:10: all] Error 2
I
On Wed, Dec 7, 2022 at 2:14 PM Jerry Geis wrote:
> Hi All,
>
> I have a physical SIP gateway device. It has 5 SIP extensions connected to
> Asterisk 10001-10005.
> These are all registered - will call this unit the SIPGW.
>
> If I use Two different phones one to call 10
Hi All,
I have a physical SIP gateway device. It has 5 SIP extensions connected to
Asterisk 10001-10005.
These are all registered - will call this unit the SIPGW.
If I use Two different phones one to call 10001 and keep the line open -
then call 10002 this works. both calls are answered and
Hi am using asterisk 18.14.0 with pulse audio and dialing console dsp and
getting a warble or a clipping in my audio.
This is my cli log
== Using SIP RTP CoS mark 5
> 0x7f47b80132a0 -- Strict RTP learning after remote address set to:
192.168.1.8:19436
-- Executing
I am running ubuntu 20.04 fully patched along with Asterisk 18.8.0
This is a VM environment with VMWare.
I found this in the logs today.
[1768362.083207] CPU: 2 PID: 1939739 Comm: asterisk Tainted: G
OEL5.15.0-52-generic #58~20.04.1-Ubuntu
[1768362.083209] call_cpuidle+0x23/0x50
On Thu, Oct 20, 2022 at 6:15 PM Eric Wieling wrote:
>
> https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces
>
>
Thanks - so based on this wiki - seems like "The only functionality that
requires internal timing is IAX2 trunking" - which I am not using .
Just ConfBridge... And getting
This is on the bare metal machine
Recalculating Codec Translation (number of sample seconds: 1)
Translation times between formats (in microseconds) for one second
of data
Source Format (Rows) Destination Format (Columns)
ulaw alaw gsm g726 g726aal2
[modules]
autoload = yes
noload = res_timing_pthread
noload = res_timing_timerfd
SO I "dont" want to load res_timing_anything ???
I have preload on res_timing_dahdi, then res_timing_pthread and not
res_timing_timerfd at all.
confbridge.conf is below
[general]
; The general section of this
What is the trick to get "preload => res_timing_dahdi" working ?
I have tried to add to both a CentOS 7 (metal box) and Ubuntu 20.04 (VMware
guest) system
restart asterisk and neither print anything about res_timing_dahdi in the
/var/log/asterisk/messages file.
Both are having issues with around
On Thu, Oct 20, 2022 at 1:53 PM Jerry Geis wrote:
> Bringing 80 devices into a conf bridge with CPU: AMD Phenom(tm) II X6
> 1045T Processor at 2.7G and audio is reported as staticy or not the best
> audio quality.
>
> Network is r8169 :02:00.0 eth0: RTL8168e/8111
> Link i
Bringing 80 devices into a conf bridge with CPU: AMD Phenom(tm) II X6 1045T
Processor at 2.7G and audio is reported as staticy or not the best audio
quality.
Network is r8169 :02:00.0 eth0: RTL8168e/8111
Link is 1G.
Asterisk 18.14.0
I would think this should be able to handle 80 calls (one
On Tue, Oct 18, 2022 at 3:38 PM Jerry Geis wrote:
> Has there been issues where "once in a while" RTP audio does not work ?
>
> Example: connection to Cisco call manager - works mostly all the time.
>
> once in a great while - person does not hear the "beep"
Has there been issues where "once in a while" RTP audio does not work ?
Example: connection to Cisco call manager - works mostly all the time.
once in a great while - person does not hear the "beep" when calling in.
once in a great while - person they hear the beep - but do not hear the
audio
ANyone ever ran into a situation when Call coming from Call Manager into
asterisk, is successful coming across - but no Audio ???
But then the next call - audio is heard - its once in a great while no
audio - most time it works.
Anything I might look for ? How do I debug that?
Thanks
jerry
--
I have a simple dialplan with asterisk 18.14.0
exten => 141,1,Answer
exten => 141,n,Noop(MC)
exten => 141,n,Playback(beep)
exten => 141,n,Dial(MulticastRTP/basic/239.168.4.90:30410//t(15))
exten => 141,n,Hangup
Most times this works just fine ... Once in a while the person hears the
beep - but
Thanks for the information
This is now working...
externip=EC2 public IP
localnet=EC2 local range
nat=force_rport,comedia
I got audio, Fantastic
Jerry
>
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>The sample configuration file outlines how things work, and the options for
>it:
>https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L874
>in general localnet and externip (or externaddr, or externhost)
I added:
externip=xxx
nat=force_rport,comedia
to the general
On Thu, Oct 6, 2022 at 9:02 AM Jerry Geis wrote:
>
>
> On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis wrote:
>
>> I am trying to get audio to work on AWS using asterisk 18.14.0
>>
>> I have enabled the firewall to allow ALL UDP on AWS
>>
>> My SIP extension
On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis wrote:
> I am trying to get audio to work on AWS using asterisk 18.14.0
>
> I have enabled the firewall to allow ALL UDP on AWS
>
> My SIP extension has
> nat=force_rport,comedia
> qualify=yes
> allow=ulaw
> allow=alaw
> all
I am trying to get audio to work on AWS using asterisk 18.14.0
I have enabled the firewall to allow ALL UDP on AWS
My SIP extension has
nat=force_rport,comedia
qualify=yes
allow=ulaw
allow=alaw
allow=gsm
canreinvite=yes
I enable "rtp set debug on" and the console is printing info.
The call
I am just doing a basic call in.
exten => 140,1,Answer
exten => 140,n,Playback(beep)
exten => 140,n,Dial(MulticastRTP/basic/239.168.4.90:3040//t(15))
exten => 140,n,Hangup
this works - but "sometimes" I get reports that "nothing" was heard.
Is there anything special to do for multicast ?
Any
On Tue, Sep 27, 2022 at 9:09 AM wrote:
> On 9/26/2022 8:25 PM, Jerry Geis wrote:
> > On Mon, Sep 26, 2022 at 8:09 PM > <mailto:aster...@phreaknet.org>> wrote:
> >
> > On 9/26/2022 8:00 PM, Jerry Geis wrote:
> > > I am getting a compile error:
On Mon, Sep 26, 2022 at 8:09 PM wrote:
> On 9/26/2022 8:00 PM, Jerry Geis wrote:
> > I am getting a compile error:
> >
> > gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes
> > -Werror=zero-length-bounds -fPIC -O2 -MD -MT q921.o -MF .q921.o.d -MP
> -c
I am getting a compile error:
gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes
-Werror=zero-length-bounds -fPIC -O2 -MD -MT q921.o -MF .q921.o.d -MP -c
-o q921.o q921.c
q921.c: In function ‘q921_dump’:
q921.c:1333:85: error: array subscript 0 is outside the bounds of an
interior
hi All
How do I restart logging in /var/log/asterisk/messages ?
asterisks is still running - but logging stopped. I think a process "trims"
the file.
How (with stopping and starting) do I get logging to happen again.
I see downloads.Asterisk.org has a dahdi release candidate from Jun... when
is
Hello - I am using asterisk 18.14.0
I think multicast uses codec g711 pcmu
is there any way to change or set the codec I want to use - like g722 ?
How would I do that?
Thanks
Jerry
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On Mon, Sep 5, 2022 at 9:16 AM Mark Murawski
wrote:
> On 8/4/22 20:32, Jerry Geis wrote:
> > I am running Asterisk 13.30.0
> > 40 core CPU (VM) VMware.
> > CentOS 7
> > 32 G ram
> > 10G vmx network
> >
> > Should be plenty of room for anythin
I am having a weird issue.
To different locations are speaking live at perhaps the same time.
Both are just connecting to multicast groups.
239.168.4.90:3041
and
239.168.4.90:3042
Somehow I am hearing audio from 3041 on the 3042 devices.
My config files for my devices show just the single
I am running Asterisk 13.30.0
40 core CPU (VM) VMware.
CentOS 7
32 G ram
10G vmx network
Should be plenty of room for anything...
Yes asterisk is running 270% CPU...
Is it not taking advantage of the 40 cores ?
I am bring around 300 SIP endpoints in a muted audio conference (so one
way) and this
I'm trying to get a sense for how many video calls with the Confbridge
can be active "when" dropping the incoming video with the confbridge setup.
So its really just the main person's video is showing "out" to all the
endpoints. So its a one to many kind of situation.
Assume normal machine: 2G
Sent from my iPhone
> On May 21, 2022, at 3:56 PM, aster...@phreaknet.org wrote:
>
> On 5/21/2022 3:49 PM, Jerry Geis wrote:
>> It has been a COUPLE years since a release of DAHDI ...
>>
>> Is there going to be one ?
>> I "desire" a release to su
It has been a COUPLE years since a release of DAHDI ...
Is there going to be one ?
I "desire" a release to support newer kernels
yes the install from git works - but I prefer to grab a real release.
Jerry
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What is the command to install dahdi on a systemctl type startup ?
I just installed dahdi from git (so latest) and did :
cd dahdi-linux-complete
ls shows
dahdi-linux and dahdi-tools
find . | grep service
shows nothing.
in dahdi-tools there is the OLD dahdi.init file - but that is the OLD
init.d
How does an external program get notification of "new" registrations ?
Would that come over the AMI or anything ?
Thanks
Jerry
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Check out the new
On Fri, Feb 4, 2022 at 12:42 PM Jerry Geis wrote:
>
>
> On Wed, Feb 2, 2022 at 1:06 PM Jerry Geis wrote:
>
>>
>>
>> On Wed, Feb 2, 2022 at 10:44 AM Jerry Geis wrote:
>>
>>>
>>>
>>> On Wed, Feb 2, 2022 at 9:26 AM Jerry Geis wr
>The usage of D(15) causes Asterisk to produce RTP on its own. Without it,
>it merely forwards RTP. If a NAT/firewall requires media to be sent before
>allowing media in, then you'll have no media flow. You can use the
>"rtpkeepalive" option to have the RTP stack produce keepalive packets,
>which
On Wed, Feb 2, 2022 at 1:06 PM Jerry Geis wrote:
>
>
> On Wed, Feb 2, 2022 at 10:44 AM Jerry Geis wrote:
>
>>
>>
>> On Wed, Feb 2, 2022 at 9:26 AM Jerry Geis wrote:
>>
>>> So I have CentOS 7 server running asterisk 18.8.0 - all is good.
>>
On Wed, Feb 2, 2022 at 10:44 AM Jerry Geis wrote:
>
>
> On Wed, Feb 2, 2022 at 9:26 AM Jerry Geis wrote:
>
>> So I have CentOS 7 server running asterisk 18.8.0 - all is good.
>>
>> I unplug that server - plug in a ubuntu 20.04 server at the same IP
>> ad
On Wed, Feb 2, 2022 at 9:26 AM Jerry Geis wrote:
> So I have CentOS 7 server running asterisk 18.8.0 - all is good.
>
> I unplug that server - plug in a ubuntu 20.04 server at the same IP
> address.
> let my 3 devices reconnect to the ubuntu server
>
> When I pick
So I have CentOS 7 server running asterisk 18.8.0 - all is good.
I unplug that server - plug in a ubuntu 20.04 server at the same IP address.
let my 3 devices reconnect to the ubuntu server
When I pick up the polycom phone and dial it connects.
I hear the other ends 'tone" - but when I press
>
>
> Hi Josh
>chan_sip did not add a video stream. What is the actual configuration for
> it? What is the actual call file used for it?
sip.conf has videosupport in the general section.
I did find that where I am "joining" the person in the conference I did not
have the Codecs: set. I added
On Wed, Jan 12, 2022 at 5:09 PM Jerry Geis wrote:
> I am running 18.8.0 - videosupport is enabled. I get video calls no
> problem.
>
> However when I make a call file to a soft phone and include:
> Codecs: ulaw,h264
> in the call file...
>
> sip show channels - shows:
I am running 18.8.0 - videosupport is enabled. I get video calls no
problem.
However when I make a call file to a soft phone and include:
Codecs: ulaw,h264
in the call file...
sip show channels - shows:
4013c15f1f4cdff (ulaw|h264) No Tx: ACK
so clearly the caller has h264.
Then
I am trying to run this command:
exten => _4XX,n,System(/usr/bin/rm /tmp/test.incoming.txt)
>From the log:
Executing [402@smvoice-sip:7] System("SIP/103-0018", "/usr/bin/rm
/tmp/test.incoming.txt") in new stack
Is "rm" not an allowed command - the above file is not removed.
-rw-rw-rw- 1
I have a machine that is completely NOT on the internet - closed network.
Can sipml5 work there ? how ?
It cannot use LetsEncrypt or anything. can self sign certs work ?
IS there another way.
Thanks
Jerry
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I have my asterisk 18 working with
https://www.doubango.org/sipml5/call.htm?svn=252#
I then tried to take the 15 lines of javascript library API (below) and
when it runs I get
asterisk console message about "failed to authenticate".I took ALL the
same settings I was using in the above URL -
Hi - Any one using SIPML5 ? How many video connections can a "normal"
asterisk server box (2.2G 8GIG ram) handle in a SINGLE video session ?
Thanks,
Jerry
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On Sat, Nov 13, 2021 at 9:41 AM Jerry Geis wrote:
> I am trying to use the SIPML5 at
> https://www.doubango.org/sipml5/call.htm?svn=252
> and when I hit the login button - and asterisk says "wrong password" and
> the web page says Forbidden.
>
> I have triple checke
I am trying to use the SIPML5 at
https://www.doubango.org/sipml5/call.htm?svn=252
and when I hit the login button - and asterisk says "wrong password" and
the web page says Forbidden.
I have triple checked that I entered the correct password on the website, I
can see the password on Asterisk
>Hello,
>You may use a UnicastRTP channel. It allows you to specify an IP/port to
>connect to.
>Regards, Jean Aunis
Hi Jean
interesting - was not aware of the unicastrtp channel - been looking
for more information on it - not finding much.
Is there anyway to bring "in" audio with unicastrtp.
Hi -
I have a device that has 16 RTP ports. I desire to bring that audio into
Asterisk... is that possible ?
The device does not run SIP at all just RTP audio. I am using Asterisk 18.
How might I do that ?
Thanks,
Jerry
--
_
On Wed, Sep 29, 2021 at 4:40 PM wrote:
> On 9/29/2021 4:38 PM, Jerry Geis wrote:
> >
> >
> > On Wed, Sep 29, 2021 at 4:31 PM > <mailto:aster...@phreaknet.org>> wrote:
> >
> > On 9/29/2021 4:08 PM, Jerry Geis wrote:
> > > I ne
On Wed, Sep 29, 2021 at 4:31 PM wrote:
> On 9/29/2021 4:08 PM, Jerry Geis wrote:
> > I need to call 1 number and that number and bring 3 phones into a
> > confbridge.
> > I tried this:
> >
> > ; PHONE CONF - Phone group Conf
> >
> > exten => 63,1
I need to call 1 number and that number and bring 3 phones into a
confbridge.
I tried this:
; PHONE CONF - Phone group Conf
exten => 63,1,Originate(SIP/401,exten,63,join_conf)
exten => 63,2,Originate(SIP/402,exten,63,join_conf)
exten => 63,3,Originate(SIP/404,exten,63,join_conf)
exten =>
Hi All - I am playing with SIPML5.
I was getting an error about wss I fixed that by doing :
cat privkey.pem > asterisk.pem
cat fullchain.pem >> asterisk.pem
with my letsencrypt certificate. and setting
tlscertfile=/etc/letsencrypt/live/mypath/asterisk.pem
But now when I use the
On Fri, Sep 10, 2021 at 1:44 PM Jerry Geis wrote:
> HI All,
>
> I am trying to get SIPml5 working with 18.6.0.
> My http.conf file:
> enabled=yes
> bindaddr=myip
> bindport=8088
> serverName=MyName
> tlsenabled=true
> tlsbindaddr=myip
> tlscertfile=/etc/lets
HI All,
I am trying to get SIPml5 working with 18.6.0.
My http.conf file:
enabled=yes
bindaddr=myip
bindport=8088
serverName=MyName
tlsenabled=true
tlsbindaddr=myip
tlscertfile=/etc/letsencrypt/live/mpname/fullchain.pem
The SIPMl log just says:
WebSocket connection to 'wss://myIP:8088/' failed:
On Fri, Aug 13, 2021 at 2:21 PM Jerry Geis wrote:
> Hi,
>
> I had a different thread going about about no audio with asterisk - I
> thought it was due to two network cards - but I dont think so any more.
> The endpoint is microsoft teams - and I think that might be the issue.
&g
Hi,
I had a different thread going about about no audio with asterisk - I
thought it was due to two network cards - but I dont think so any more.
The endpoint is microsoft teams - and I think that might be the issue.
Does Asterisk 18.6.0 work with Microsoft teams ? my issue is no audio.
I have
On Mon, Aug 9, 2021 at 11:05 AM Jerry Geis wrote:
>
>
> On Mon, Aug 9, 2021 at 8:32 AM Jerry Geis wrote:
>
>>
>>
>> On Mon, Aug 9, 2021 at 8:11 AM Jerry Geis wrote:
>>
>>>
>>>
>>> On Mon, Aug 9, 2021 at 7:57 AM Jerry Geis wrot
On Mon, Aug 9, 2021 at 8:32 AM Jerry Geis wrote:
>
>
> On Mon, Aug 9, 2021 at 8:11 AM Jerry Geis wrote:
>
>>
>>
>> On Mon, Aug 9, 2021 at 7:57 AM Jerry Geis wrote:
>>
>>>
>>>
>>> On Sun, Aug 8, 2021 at 3:18 PM Jerry Geis wrote:
&g
On Mon, Aug 9, 2021 at 8:11 AM Jerry Geis wrote:
>
>
> On Mon, Aug 9, 2021 at 7:57 AM Jerry Geis wrote:
>
>>
>>
>> On Sun, Aug 8, 2021 at 3:18 PM Jerry Geis wrote:
>>
>>> I am not using a SIP trunk as I normally do.
>>>
>>> I have
On Mon, Aug 9, 2021 at 7:57 AM Jerry Geis wrote:
>
>
> On Sun, Aug 8, 2021 at 3:18 PM Jerry Geis wrote:
>
>> I am not using a SIP trunk as I normally do.
>>
>> I have an extensions 3382 setup that my server registers to the other SIP
>> system.
>> When
On Sun, Aug 8, 2021 at 3:18 PM Jerry Geis wrote:
> I am not using a SIP trunk as I normally do.
>
> I have an extensions 3382 setup that my server registers to the other SIP
> system.
> When the other system calls 3381 on my system I am getting this error:
>
> [Jul 27 10:08
I am not using a SIP trunk as I normally do.
I have an extensions 3382 setup that my server registers to the other SIP
system.
When the other system calls 3381 on my system I am getting this error:
[Jul 27 10:08:50] WARNING[89791][C-0068] chan_sip.c: username mismatch,
have <3381>, digest
Is there a way to "not" compile/configure pjsip in 18 ?
I am still using the older SIP channel driver and have not converted over
just yet.
Thanks,
Jerry
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