Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
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.com/asterisk/asterisk/pull/309 > > > The change "seems" to be working. > Will test more tomorrow - had to leave. > THANKS! > > Jerry > Yes - this fix is working for me. Only issue I have now is, I used to run asterisk like this: su silentm -c "/usr/sbin/asterisk -fn" I also tried su silentm -l -c "/usr/sbin/asterisk -fn" these do not work for the chan_console. I have to actually login as silentm and then run asterisks - to HEAR the audio. doing su above I do not hear the audio - but the CLI looks the same - no errors. Thoughts? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>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 leave. THANKS! Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
> 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 cannot run it after teh Dial() as the Dial(Console/default,20,g) never returns to the dial plan. jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>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 not doing the hangup or the system. jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>
>
> 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 this is a bummer
[Sep 8 08:45:28] WARNING[313684][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
--- <("<) --- Hangup on Console --- (>")> ---
[Sep 8 08:45:28] ERROR[313683]: utils.c:1027 lock_info_destroy: Thread
'stream_monitor started at [ 390] chan_console.c start_stream()'
still has a lock! - 'pvt' (0x55e647a245e0) from 'stream_monitor' in
chan_console.c:281!
How do we get this working
Jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
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 that - and just let it do the Dial() - I stopped
everything - got it running again. - and then the Dial() hangs on the
second call.
So both ChanIsAvail() or Dial() both hang on the second call in.
So only 1 call in will work.
Below is the CLI report of the call that works.
This is my context
[smvoice-mediacontroller-public-address]
exten => s,1,ChanIsAvail(Console/default)
exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/default)
exten => s,n,Hangup
Now what ???
Jerry
onnected to Asterisk 18.18.0 currently running on nuc11cdev2 (pid = 282195)
== Using SIP RTP CoS mark 5
> 0x7feeec0086b0 -- Strict RTP learning after remote address set to:
192.168.1.8:17526
-- Executing [public_address@smvoice-mediacontroller:1]
SoftHangup("SIP/devgeis_to_nuc11cdev2-", "ALSA/dummy") in new stack
-- Executing [public_address@smvoice-mediacontroller:2]
Goto("SIP/devgeis_to_nuc11cdev2-",
"smvoice-mediacontroller-public-address,s,1") in new stack
-- Goto (smvoice-mediacontroller-public-address,s,1)
-- Executing [s@smvoice-mediacontroller-public-address:1]
NoOp("SIP/devgeis_to_nuc11cdev2-", "JERRY") in new stack
-- Executing [s@smvoice-mediacontroller-public-address:2]
Playback("SIP/devgeis_to_nuc11cdev2-", "beep") in new stack
> 0x7feeec0086b0 -- Strict RTP switching to RTP target address
192.168.1.8:17526 as source
-- Playing 'beep.gsm' (language
'en')
-- Executing [s@smvoice-mediacontroller-public-address:3]
Dial("SIP/devgeis_to_nuc11cdev2-", "Console/default") in new stack
--- <("<) --- Call to device 'default' on console from 'MyName Here'
<2564286000> --- (>")> ---
--- <("<) --- Auto-answered --- (>")> ---
-- Called Console/default
-- Console/default answered SIP/devgeis_to_nuc11cdev2-
-- Channel Console/default joined 'simple_bridge' basic-bridge
[Sep 8 08:07:10] WARNING[282457][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
-- Channel SIP/devgeis_to_nuc11cdev2- joined 'simple_bridge'
basic-bridge
> 0x7feeec0086b0 -- Strict RTP learning complete - Locking on source
address 192.168.1.8:17526
-- Channel SIP/devgeis_to_nuc11cdev2- left 'simple_bridge'
basic-bridge
-- Channel Console/default left 'simple_bridge' basic-bridge
[Sep 8 08:07:17] WARNING[282457][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
--- <("<) --- Hangup on Console --- (>")> ---
== Spawn extension (smvoice-mediacontroller-public-address, s, 3) exited
non-zero on 'SIP/devgeis_to_nuc11cdev2-'
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
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
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
I found "console list available" === === - === Device Name: default === ---> Default Input Device === ---> Default Output Device === - === === - === Device Name: dmix === ---> Output Device === - === = dmix is there and default is there I tried both - and get the same error Console device "dmix" not found . etc. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 18.18.0 and chan_console
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 20.04 LTS ? I can "aplay /usr/share/sounds/alsa/Front_Center.wav" no problem. Thoughts? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 18.18.0 and chan_console
> 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 18.18.0 and chan_console
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 18.18.0 and chan_console
> 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 18.18.0 and chan_console
>
>
>
> 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 => s,1,ChanIsAvail(Console/dsp)
exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/dsp)
exten => s,n,Hangup
I also tried Console/ALSA and both gave errors.
Thanks
Jerry
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Re: [asterisk-users] asterisk 18.18.0 and chan_console
> > > 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 18.18.0 and chan_console
> 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 18.18.0 and chan_console
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 60+ devices in confbridge and dropping audio
> 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 60+ devices in confbridge and dropping audio
> > > 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 dropping ? > *>* not much else is running on the server besides - asterisk. > *> > Is it virtualized? What timer is being used? > > one way conf is ME - setting up users to be Muted. [ConfUserMuted] type=user quiet=yes startmuted=yes announce_only_user=no announce_user_count_all=no announce_join_leave=no The people at that environment shared that the audio is choppy or drops out. The server is Virtual server. How do you tell what "timer" is used ? Thanks jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 60+ devices in confbridge and dropping 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) Xeon(R) CPU E3-1240 v5 @ 3.50GHz > > What else might I tweak to get this working without audio dropping ? > not much else is running on the server besides - asterisk. > > Thanks > > Jerry > > What else have you tried ? jitterbuffer ? mixing_interval ? codecs ? I did just now change the mixing_interval to 40. Asked them to try again. WHich setting has the BIGGEST impact ? (typically) For codec all the endpoints are just ulaw and alaw. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 60+ devices in confbridge and dropping audio
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 audio dropping ? not much else is running on the server besides - asterisk. Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Media flow between them
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 AUDIO. > > I see this in the CLI > > > -- Channel SIP/63009-0013 joined 'simple_bridge' basic-bridge > <475050e7-9d99-43f0-a9bf-7aa581a97fd9> > -- Channel SIP/63000-0012 joined 'simple_bridge' basic-bridge > <475050e7-9d99-43f0-a9bf-7aa581a97fd9> >> Bridge 475050e7-9d99-43f0-a9bf-7aa581a97fd9: switching from > simple_bridge technology to native_rtp >> Remotely bridged 'SIP/63000-0012' and 'SIP/63009-0013' - > media will flow directly between them > > I added in general section of sip.conf (chan_sip in use) > directrtpsetup=no > directmedia=no > > but yet I still see "media will flow directly between them". > HOW do I turn this off - RTP has to go through the server. > > > Thanks > > Jerry > even easier: canreinvite=no I had yes. works Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Media flow between them
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' basic-bridge <475050e7-9d99-43f0-a9bf-7aa581a97fd9> -- Channel SIP/63000-0012 joined 'simple_bridge' basic-bridge <475050e7-9d99-43f0-a9bf-7aa581a97fd9> > Bridge 475050e7-9d99-43f0-a9bf-7aa581a97fd9: switching from simple_bridge technology to native_rtp > Remotely bridged 'SIP/63000-0012' and 'SIP/63009-0013' - media will flow directly between them I added in general section of sip.conf (chan_sip in use) directrtpsetup=no directmedia=no but yet I still see "media will flow directly between them". HOW do I turn this off - RTP has to go through the server. Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] audio from soft phone actual phone from cloud
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 call from Indiana to Chicago - the phone rings - but I do not get any audio? I have in sip.conf externip=real_ip_here localnet=172.31.17.0/255.255.255.0 localnet=192.168.11.0/255.255.255.0 localnet=192.168.1.0/255.255.252.0 localnet=10.0.0.0/255.255.255.0 One phone config: (both are the same) [Y] type=friend defaultname=Y defaultuser=Y secret=notshown dtmfmode=RFC2833 host=dynamic description=testing. context=some-context-that-works rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 callerid=YY qualify=yes insecure= canreinvite=yes timezone=0 nat=force_rport,comedia disallow=all allow=ulaw allow=alaw allow=gsm Which accounts for all locations. Why might I not be getting audio ? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 18.17.1 unreachable
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. then they come back online again and are OK. I'm using old chan_sip, I tried changing qualify to no - that did not help. What might I adjust to keep these SIP units alway ON ? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 401 error
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. >>> >>> 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 >>> >> >> Thanks I am using chan_sip. Turning on "sip set debug on" I do se it. >> >> >> >> Using INVITE request as basis request - >> 0gQAAC8WAAACBAAALxYAAJJTfs5d3xOERBazRh4eM17uDp4m3CsM7ZnaEtaSRzwP@IP >> Found peer 'JJ' for 'phone' from IP:5060 >> >> <--- Reliably Transmitting (no NAT) to IP:5060 ---> >> SIP/2.0 401 Unauthorized^M >> Via: SIP/2.0/UDP >> IP:5060;branch=z9hG4bK+13778ac6c6ac51ac853b3fb4a387e5b71+sip+3+ae5ab1a2;received=IP^M >> From: "Caller" ;tag=IP+3+67d18b6f+9e6ad02d^M >> To: ;tag=as128621a0^M >> Call-ID: >> 0gQAAC8WAAACBAAALxYAAJJTfs5d3xOERBazRh4eM17uDp4m3CsM7ZnaEtaSRzwP@IP^M >> CSeq: 503124310 INVITE^M >> Server: Asterisk PBX 18.14.0^M >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE^M >> Supported: replaces, timer^M >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", >> nonce="6cbb5c2f"^M >> Content-Length: 0^M >> >> I dont see a reason why it failed. >> I tried nat=yes, made no difference. >> I tried insecure=very, made no difference. >> >> I do have: >> externip=X >> localnet=Y >> localnet=Z >> set in sip.conf >> >> As I mentioned - I can call out over this SIP trunk. >> What next ? >> Jerry >> > > > Just added insecure=very again, stopped and started. > > > [JJ] > type=friend > dtmfmode=rfc2833 > secret=yes > username=NUMBER > defaultuser=NUMBER > disallow=all > allow=ulaw > allow=alaw > context=smvoice-incoming > host=dnsname > canreinvite=yes > qualify=yes > insecure=very > > Got the same 401. > Thanks > > Jerry > > Thank you for the suggestions - it got me to this insecure=port,invite This worked. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 401 error
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 the CLI. >> The person says the server is returning 401 >> >> How do I debug that. Using asterisk 18.8.0 >> >> Thanks >> >> Jerry >> > > Thanks I am using chan_sip. Turning on "sip set debug on" I do se it. > > > > Using INVITE request as basis request - > 0gQAAC8WAAACBAAALxYAAJJTfs5d3xOERBazRh4eM17uDp4m3CsM7ZnaEtaSRzwP@IP > Found peer 'JJ' for 'phone' from IP:5060 > > <--- Reliably Transmitting (no NAT) to IP:5060 ---> > SIP/2.0 401 Unauthorized^M > Via: SIP/2.0/UDP > IP:5060;branch=z9hG4bK+13778ac6c6ac51ac853b3fb4a387e5b71+sip+3+ae5ab1a2;received=IP^M > From: "Caller" ;tag=IP+3+67d18b6f+9e6ad02d^M > To: ;tag=as128621a0^M > Call-ID: > 0gQAAC8WAAACBAAALxYAAJJTfs5d3xOERBazRh4eM17uDp4m3CsM7ZnaEtaSRzwP@IP^M > CSeq: 503124310 INVITE^M > Server: Asterisk PBX 18.14.0^M > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE^M > Supported: replaces, timer^M > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", > nonce="6cbb5c2f"^M > Content-Length: 0^M > > I dont see a reason why it failed. > I tried nat=yes, made no difference. > I tried insecure=very, made no difference. > > I do have: > externip=X > localnet=Y > localnet=Z > set in sip.conf > > As I mentioned - I can call out over this SIP trunk. > What next ? > Jerry > Just added insecure=very again, stopped and started. [JJ] type=friend dtmfmode=rfc2833 secret=yes username=NUMBER defaultuser=NUMBER disallow=all allow=ulaw allow=alaw context=smvoice-incoming host=dnsname canreinvite=yes qualify=yes insecure=very Got the same 401. Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 401 error
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 that. Using asterisk 18.8.0 > > Thanks > > Jerry > Thanks I am using chan_sip. Turning on "sip set debug on" I do se it. Using INVITE request as basis request - 0gQAAC8WAAACBAAALxYAAJJTfs5d3xOERBazRh4eM17uDp4m3CsM7ZnaEtaSRzwP@IP Found peer 'JJ' for 'phone' from IP:5060 <--- Reliably Transmitting (no NAT) to IP:5060 ---> SIP/2.0 401 Unauthorized^M Via: SIP/2.0/UDP IP:5060;branch=z9hG4bK+13778ac6c6ac51ac853b3fb4a387e5b71+sip+3+ae5ab1a2;received=IP^M From: "Caller" ;tag=IP+3+67d18b6f+9e6ad02d^M To: ;tag=as128621a0^M Call-ID: 0gQAAC8WAAACBAAALxYAAJJTfs5d3xOERBazRh4eM17uDp4m3CsM7ZnaEtaSRzwP@IP ^M CSeq: 503124310 INVITE^M Server: Asterisk PBX 18.14.0^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE^M Supported: replaces, timer^M WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6cbb5c2f"^M Content-Length: 0^M I dont see a reason why it failed. I tried nat=yes, made no difference. I tried insecure=very, made no difference. I do have: externip=X localnet=Y localnet=Z set in sip.conf As I mentioned - I can call out over this SIP trunk. What next ? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 401 error
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Not reporting IP of the incoming connection 18.14.0
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] NOTICE[2959153][C-06d2] chan_sip.c: Failed to authenticate device ;tag=22518526 for INVITE, code = -1 [Feb 9 15:37:56] NOTICE[2959153][C-06d7] chan_sip.c: Failed to authenticate device ;tag=2145228727 for INVITE, code = -1 But how do I get what IP this message is coming from ? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi Compile on 22.04 LTS
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 have all these libusb installed apt list --installed | grep libusb libusb-0.1-4/jammy,now 2:0.1.12-32build3 amd64 [installed] libusb-1.0-0-dev/jammy-updates,now 2:1.0.25-1ubuntu2 amd64 [installed] libusb-1.0-0/jammy-updates,now 2:1.0.25-1ubuntu2 amd64 [installed] libusb-1.0-doc/jammy-updates,now 2:1.0.25-1ubuntu2 all [installed] libusb-dev/jammy,now 2:0.1.12-32build3 amd64 [installed] libusb-java-doc/jammy,now 0.8+ztex20090101-9build1 all [installed] libusb-java-lib/jammy,now 0.8+ztex20090101-9build1 amd64 [installed] libusb-java/jammy,now 0.8+ztex20090101-9build1 all [installed] libusb-libusb-perl/jammy,now 0.09-2build2 amd64 [installed] libusb-ocaml-dev/jammy,now 1.3.1-4build6 amd64 [installed] libusb-ocaml/jammy,now 1.3.1-4build6 amd64 [installed] libusb3380-0/jammy,now 0.0.1+git20190125.c83d1e9-2 amd64 [installed] libusb3380-dev/jammy,now 0.0.1+git20190125.c83d1e9-2 amd64 [installed] libusbauth-configparser-dev/jammy,now 1.0.3-1 amd64 [installed] libusbauth-configparser1/jammy,now 1.0.3-1 amd64 [installed] libusbguard0/jammy,now 1.1.1+ds-3 amd64 [installed] libusbmuxd-dev/jammy,now 2.0.2-3build2 amd64 [installed] libusbmuxd-tools/jammy,now 2.0.2-3build2 amd64 [installed] libusbmuxd6/jammy,now 2.0.2-3build2 amd64 [installed] libusbredirhost-dev/jammy,now 0.11.0-2build1 amd64 [installed] libusbredirhost1/jammy,now 0.11.0-2build1 amd64 [installed] libusbredirparser-dev/jammy,now 0.11.0-2build1 amd64 [installed] libusbredirparser1/jammy,now 0.11.0-2build1 amd64 [installed] What might I be missing ? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two calls from same server to end device
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 10001 and keep the line open - > then call 10002 this works. both calls are answered and speaking. > > So if I make a call from my asterisk server with "call files" one to the > 10001 it answers - and one to the 10002 - at different times - BOTH work. > > However if I make the 10001 call - and while its still speaking I call the > 10002 - the second call gets unanswered till the first call is completed. > I'm guessing that is a function of the SIPGW - good or bad > > Take that a step further - if I use the call file to call 10001 - it > answers - from the polycom I call 10002 - this works also - two > different sources I presume > > My question is can "in a call file" somehow - say I am a different source > or something ? > just like the two polycom phones that work - two different source > addresses or something. > > Anyone ran into this - or thoughts on something I might try to say the > calls are different sources ? > > Thank you. > > Jerry > Turns out simple setting the CALLERID to different values does the trick. Awesome! jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Two calls from same server to end device
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 speaking. So if I make a call from my asterisk server with "call files" one to the 10001 it answers - and one to the 10002 - at different times - BOTH work. However if I make the 10001 call - and while its still speaking I call the 10002 - the second call gets unanswered till the first call is completed. I'm guessing that is a function of the SIPGW - good or bad Take that a step further - if I use the call file to call 10001 - it answers - from the polycom I call 10002 - this works also - two different sources I presume My question is can "in a call file" somehow - say I am a different source or something ? just like the two polycom phones that work - two different source addresses or something. Anyone ran into this - or thoughts on something I might try to say the calls are different sources ? Thank you. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 18.14.0 console dsp
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 [public_address@smvoice-mediacontroller:1]
SoftHangup("SIP/nuc7cdev1-0002", "ALSA/dummy") in new stack
-- Executing [public_address@smvoice-mediacontroller:2]
Goto("SIP/nuc7cdev1-0002",
"smvoice-mediacontroller-public-address,s,1") in new stack
-- Goto (smvoice-mediacontroller-public-address,s,1)
-- Executing [s@smvoice-mediacontroller-public-address:1]
ChanIsAvail("SIP/nuc7cdev1-0002", "Console/Dsp") in new stack
<< Hangup on console >>
-- Executing [s@smvoice-mediacontroller-public-address:2]
GotoIf("SIP/nuc7cdev1-0002", "0?smvoice-busy,s,1") in new stack
-- Executing [s@smvoice-mediacontroller-public-address:3]
System("SIP/nuc7cdev1-0002", "/home/silentm/bin/smfunctions
-totem_pause") in new stack
-- Executing [s@smvoice-mediacontroller-public-address:4]
Playback("SIP/nuc7cdev1-0002", "beep") in new stack
> 0x7f47b80132a0 -- Strict RTP switching to RTP target address
192.168.1.8:19436 as source
-- Playing 'beep.gsm' (language
'en')
-- Executing [s@smvoice-mediacontroller-public-address:5]
Dial("SIP/nuc7cdev1-0002", "Console/dsp") in new stack
<< Call placed to 'dsp' on console >>
<< Auto-answered >>
-- Called Console/dsp
-- ALSA/default answered SIP/nuc7cdev1-0002
-- Channel ALSA/default joined 'simple_bridge' basic-bridge
<2df4409d-39c0-4b1e-bb6f-8485d3c331fc>
-- Channel SIP/devgeis_to_nuc7cdev1-0002 joined 'simple_bridge'
basic-bridge <2df4409d-39c0-4b1e-bb6f-8485d3c331fc>
[Nov 10 14:20:58] WARNING[15363][C-0003]: chan_alsa.c:573
alsa_indicate: Don't know how to display condition 26 on ALSA/default
-- Channel SIP/devgeis_to_nuc7cdev1-0002 left 'simple_bridge'
basic-bridge <2df4409d-39c0-4b1e-bb6f-8485d3c331fc>
-- Channel ALSA/default left 'simple_bridge' basic-bridge
<2df4409d-39c0-4b1e-bb6f-8485d3c331fc>
[Nov 10 14:21:04] WARNING[15363][C-0003]: chan_alsa.c:573
alsa_indicate: Don't know how to display condition 26 on ALSA/default
== Spawn extension (smvoice-mediacontroller-public-address, s, 5) exited
non-zero on 'SIP/nuc7cdev1-0002'
<< Hangup on console >>
What is clipping or warble from ?
I also tried the Console/dsp/answer and the same happens with the sound.
Thanks
Jerry
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[asterisk-users] asterisk kernel crash
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 [1768362.083217] do_idle+0x1f4/0x270 [1768362.083222] Hardware name: VMware, Inc. VMware Virtual Platform/440BX Desktop Reference Platform, BIOS 6.00 11/12/2020 [1768362.083227] cpu_startup_entry+0x20/0x30 [1768362.083230] RIP: 0010:dev_watchdog+0x27a/0x290 [1768362.083234] start_secondary+0x12a/0x180 [1768362.083245] secondary_startup_64_no_verify+0xc2/0xcb [1768362.083248] Code: eb 97 48 8b 5d d0 c6 05 01 49 6c 01 01 48 89 df e8 6b 78 f9 ff 44 89 e1 48 89 de 48 c7 c7 38 e2 ad a1 48 89 c2 e8 fe 54 19 00 <0f> 0b eb 80 e9 a [1768362.083259] RSP: 0018:abcb00524e80 EFLAGS: 00010282 [1768362.083262] [1768362.083550] asm_sysvec_apic_timer_interrupt+0x1a/0x20 [1768362.083558] RIP: 0010:___slab_alloc+0x564/0x860 [1768362.083577] Code: 74 24 48 4c 89 f7 4c 01 e8 48 89 c2 48 0f ca 49 33 96 b8 00 00 00 48 33 10 e8 e8 ef ff ff e9 97 fe ff ff fb 66 0f 1f 44 00 00 8b fe ff ff 65 8b 05 c8 75 8e 5f 48 98 49 8b b4 c6 d8 00 00 00 [1768362.083586] RSP: 0018:abcb053db740 EFLAGS: 0206 [1768362.083594] RAX: 8a5935eb5180 RBX: RCX: 034de982 [1768362.083600] RDX: 8a58542bda00 RSI: e2f3c550ae00 RDI: 0200 [1768362.083605] RBP: abcb053db810 R08: 00082cc0 R09: 0002 [1768362.083610] R10: R11: 8a58254eb200 R12: 8a5935eb5160 [1768362.083615] R13: 8a58542bd800 R14: 8a5800042b00 R15: 0002 [1768362.083630] ? e1000_xmit_frame+0xa14/0x1180 [e1000] [1768362.083655] ? __alloc_skb+0x92/0x1e0 [1768362.083668] ? __kmalloc_node_track_caller+0x1d0/0x4e0 [1768362.083676] ? __alloc_skb+0x16c/0x1e0 [1768362.083687] __kmalloc_node_track_caller+0x112/0x4e0 [1768362.083694] ? __alloc_skb+0x92/0x1e0 [1768362.083704] kmalloc_reserve+0x31/0x90 [1768362.083715] __alloc_skb+0x92/0x1e0 [1768362.083725] alloc_skb_with_frags+0x4e/0x1e0 [1768362.083739] sock_alloc_send_pskb+0x207/0x230 [1768362.083751] sock_alloc_send_skb+0x18/0x20 [1768362.083758] __ip_append_data.isra.0+0x7ed/0xe60 [1768362.083774] ? ip_reply_glue_bits+0x50/0x50 [1768362.083787] ? ip_reply_glue_bits+0x50/0x50 [1768362.083799] ip_make_skb+0xd4/0x110 [1768362.083815] udp_sendmsg+0xb74/0xe50 [1768362.083826] ? ip_reply_glue_bits+0x50/0x50 [1768362.083843] ? copyout+0x20/0x40 [1768362.083853] ? __cond_resched+0x19/0x40 [1768362.083863] ? aa_sk_perm+0x43/0x1c0 [1768362.083895] inet_sendmsg+0x6d/0x80 [1768362.083908] ? inet_sendmsg+0x6d/0x80 [1768362.083919] sock_sendmsg+0x62/0x70 [1768362.083929] __sys_sendto+0x113/0x1a0 [1768362.083944] ? exit_to_user_mode_prepare+0x3d/0x1c0 [1768362.083955] ? syscall_exit_to_user_mode+0x27/0x50 [1768362.083967] ? __do_sys_gettid+0x1b/0x30 [1768362.083980] __x64_sys_sendto+0x29/0x40 [1768362.083990] do_syscall_64+0x59/0xc0 [1768362.084000] ? syscall_exit_to_user_mode+0x27/0x50 [1768362.084011] ? __x64_sys_read+0x1a/0x20 [1768362.084020] ? do_syscall_64+0x69/0xc0 [1768362.084027] ? do_syscall_64+0x69/0xc0 [1768362.084035] ? do_syscall_64+0x69/0xc0 [1768362.084042] ? do_syscall_64+0x69/0xc0 [1768362.084050] ? do_syscall_64+0x69/0xc0 [1768362.084057] ? sysvec_apic_timer_interrupt+0x4e/0x90 [1768362.084069] entry_SYSCALL_64_after_hwframe+0x61/0xcb [1768362.084082] RIP: 0033:0x7fd5ca0028a4 [1768362.084091] Code: c2 f7 ff ff 44 8b 4c 24 2c 4c 8b 44 24 20 89 c5 44 8b 54 24 28 48 8b 54 24 18 b8 2c 00 00 00 48 8b 74 24 10 8b 7c 24 08 0f 05 <48> 3d 00 f0 ff ff 77 30 89 ef 48 89 44 24 08 e8 e8 f7 ff ff 48 8b [1768362.084098] RSP: 002b:7fd55ce74b50 EFLAGS: 0293 ORIG_RAX: 002c [1768362.084107] RAX: ffda RBX: 00ac RCX: 7fd5ca0028a4 [1768362.084112] RDX: 00ac RSI: 7fd54006a36c RDI: 0104 [1768362.084117] RBP: R08: 7fd55ce74c40 R09: 0010 [1768362.084130] R10: R11: 0293 R12: [1768362.084135] R13: 7fd4e00287c0 R14: 7fd4e00287c0 R15: 7fd55ce74c40 [1768362.084146] [1768362.084150] ---[ end trace 72a16d2ead475475 ]--- [1768362.103485] e1000 :02:01.0 ens33: Reset adapter [1768362.214181] e1000: ens33 NIC Link is Up 1000 Mbps Full Duplex, Flow Control: None What do I do about that ? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confbridge for 80 devices
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 crappy audio with about 80 devices and a 1 - way conf. other thoughts on what is happening ? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confbridge for 80 devices
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 adpcm slin8 slin12 slin16 slin24 slin32 slin44 slin48 slin96 slin192 lpc10 speex8 speex16 speex32 ilbc g722 testlaw ulaw - 9150 15000 1500015000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 23000 23000 15000 17250 15000 alaw 9150 - 15000 1500015000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 23000 23000 15000 17250 15000 gsm 15000 15000 - 1500015000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 23000 23000 15000 17250 15000 g726 15000 15000 15000 -15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 23000 23000 15000 17250 15000 g726aal2 15000 15000 15000 15000- 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 23000 23000 15000 17250 15000 adpcm 15000 15000 15000 1500015000 - 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 23000 23000 15000 17250 15000 slin8 6000 6000 6000 6000 6000 6000 - 8000 8000 8000 8000 8000 8000 80008000 6000 6000 14000 14000 6000 82506000 slin12 14500 14500 14500 1450014500 14500 8500 - 8000 8000 8000 8000 8000 80008000 14500 14500 14000 14000 14500 14000 14500 slin16 14500 14500 14500 1450014500 14500 8500 8500 - 8000 8000 8000 8000 80008000 14500 145006000 14000 14500 6000 14500 slin24 14500 14500 14500 1450014500 14500 8500 8500 8500 - 8000 8000 8000 80008000 14500 14500 14500 14000 14500 14500 14500 slin32 14500 14500 14500 1450014500 14500 8500 8500 8500 8500 - 8000 8000 80008000 14500 14500 145006000 14500 14500 14500 slin44 14500 14500 14500 1450014500 14500 8500 8500 8500 8500 8500 - 8000 80008000 14500 14500 14500 14500 14500 14500 14500 slin48 14500 14500 14500 1450014500 14500 8500 8500 8500 8500 8500 8500 - 80008000 14500 14500 14500 14500 14500 14500 14500 slin96 14500 14500 14500 1450014500 14500 8500 8500 8500 8500 8500 8500 8500 -8000 14500 14500 14500 14500 14500 14500 14500 slin192 14500 14500 14500 1450014500 14500 8500 8500 8500 8500 8500 8500 8500 8500 - 14500 14500 14500 14500 14500 14500 14500 lpc10 15000 15000 15000 1500015000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 - 15000 23000 23000 15000 17250 15000 speex8 15000 15000 15000 1500015000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 - 23000 23000 15000 17250 15000 speex16 23500 23500 23500 2350023500 23500 17500 17500 9000 17000 17000 17000 17000 17000 17000 23500 23500 - 23000 23500 15000 23500 speex32 23500 23500 23500 2350023500 23500 17500 17500 17500 17500 9000 17000 17000 17000 17000 23500 23500 23500 - 23500 23500 23500 ilbc 15000 15000 15000 1500015000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 23000 23000 - 17250 15000 g722 15600 15600 15600 1560015600 15600 9600 17500 9000 17000 17000 17000 17000 17000 17000 15600 15600 15000 23000 15600 - 15600 testlaw 15000 15000 15000 1500015000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 23000 23000 15000 17250 - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confbridge for 80 devices
[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 config ; is not currently used, but reserved ; for future use. ; ; --- Default Information --- ; The default_user and default_bridge sections are applied ; automatically to all ConfBridge instances invoked without ; a user, or bridge argument. No menu is applied by default. ; ; --- ConfBridge User Profile Options --- [default_user] type=user ;admin=yes ; Sets if the user is an admin or not. Off by default. ;marked=yes; Sets if this is a marked user or not. Off by default. ;startmuted=yes; Sets if all users should start out muted. Off by default ;music_on_hold_when_empty=yes ; Sets whether MOH should be played when only ; one person is in the conference or when the ; the user is waiting on a marked user to enter ; the conference. Off by default. ;music_on_hold_class=default ; The MOH class to use for this user. ;quiet=yes ; When enabled enter/leave prompts and user intros are not played. ; There are some prompts, such as the prompt to enter a PIN number, ; that must be played regardless of what this option is set to. ; Off by default ;announce_user_count=yes ; Sets if the number of users should be announced to the ; caller. Off by default. ;announce_user_count_all=yes ; Sets if the number of users should be announced to ; all the other users in the conference when someone joins. ; This option can be either set to 'yes' or a number. ; When set to a number, the announcement will only occur ; once the user count is above the specified number. ;announce_only_user=yes ; Sets if the only user announcement should be played ; when a channel enters a empty conference. On by default. ;wait_marked=yes ; Sets if the user must wait for a marked user to enter before ; joining the conference. Off by default. ;end_marked=yes ; This option will kick every user with this option set in their ; user profile after the last Marked user exists the conference. ;dsp_drop_silence=yes ; This option drops what Asterisk detects as silence from ; entering into the bridge. Enabling this option will drastically ; improve performance and help remove the buildup of background ; noise from the conference. Highly recommended for large conferences ; due to its performance enhancements. ;dsp_talking_threshold=128 ; The time in milliseconds of sound above what the dsp has ; established as base line silence for a user before a user ; is considered to be talking. This value affects several ; operations and should not be changed unless the impact on ; call quality is fully understood. ; ; What this value affects internally: ; ; 1. Audio is only mixed out of a user's incoming audio stream ;if talking is detected. If this value is set too ;loose the user will hear themselves briefly each ;time they begin talking until the dsp has time to ;establish that they are in fact talking. ; 2. When talk detection AMI events are enabled, this value ;determines when talking has begun which results in ;an AMI event to fire. If this value is set too tight ;AMI events may be falsely triggered by variants in ;room noise. ; 3. The drop_silence option depends on this value to determine ;when the user's audio should be mixed into the bridge ;after periods of silence. If this value is too loose ;the beginning of a user's speech will get cut off as they ;transition from silence to talking. ; ; By default this value is 160 ms. Valid values are 1 through 2^31 ;dsp_silence_threshold=2000 ; The time in milliseconds of sound falling within the what ; the dsp has established as baseline
Re: [asterisk-users] Confbridge for 80 devices
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 80 Confbridge items. timing test on BOTH return the same... 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 I dont think either system is "correctly" using the dahdi timer. Thoughts ? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confbridge for 80 devices
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 is 1G. > > Asterisk 18.14.0 > > I would think this should be able to handle 80 calls (one way audio). > > How can I tell if asterisk is an able to handle this - or how can I find > the bottle neck? > > Thanks > > Jerry > So I did the "timing test" got timing test Attempting to test a timer with 50 ticks per second. Using the 'timerfd' timing module for this test. in modules I added [modules] autoload=yes preload => res_timing_dahdi.so preload => res_timing_pthread.so Re-ran the timing test got the same thing. Notice it says specifically timerfd - not timing_dahdi or timing_pthread. System is CentOS 7. lsmod | grep dahdi lsmod | grep dahdi dahdi_transcode14291 1 wctc4xxp dahdi_voicebus 59241 1 wctdm24xxp dahdi 228002 9 wctdm24xxp,wcaxx,dahdi_transcode,oct612x,dahdi_voicebus,wcb4xxp,wct4xxp,wcte43x,wcte13xp crc_ccitt 12707 2 wctdm24xxp,dahdi Is my timing OK - or not ? Note /var/log/asterisk/messages says "nothing" about timing even though the resourse were loaded. Thoughts ? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Confbridge for 80 devices
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 way audio). How can I tell if asterisk is an able to handle this - or how can I find the bottle neck? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP audio
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" when calling in. > once in a great while - person they hear the beep - but do not hear the > audio public address. > > What would I be looking for to track this beast down ? > > This is my SIP trunk > [LSVOIP] > type=friend > dtmfmode=rfc2833 > secret=password > username=LSVOIP > defaultuser=LSVOIP > disallow=all > allow=ulaw > allow=alaw > context=incoming > host=172.1.1.1 > canreinvite=yes > qualify=yes > insecure=invite > > Thoughts? > > Jerry > Is there any kind of pjsip vs old SIP (which I am using) issue happening here. (asterisk 18.14.0) Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP audio
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 public address. What would I be looking for to track this beast down ? This is my SIP trunk [LSVOIP] type=friend dtmfmode=rfc2833 secret=password username=LSVOIP defaultuser=LSVOIP disallow=all allow=ulaw allow=alaw context=incoming host=172.1.1.1 canreinvite=yes qualify=yes insecure=invite Thoughts? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 18.14.0 connected to Call Manager
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Muliticast not connecting
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 nothing connects on the multicast. What might this be? How can I tell what is happening and why it does not connect? is it valid to put : exten => 141,n,Dial(MulticastRTP/basic/239.168.4.90:30410//t(15)) exten => 141,n,Dial(MulticastRTP/basic/239.168.4.90:30410//t(15)) So if the first one doesnt connect perhaps the second one will ??? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying asterisk on AWS
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 > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying asterisk on AWS
>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 section of sip.conf its still sending to the local IP. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying asterisk on AWS
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 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 comes into my linphone softphone - but I get no audio on my >> linphone softphone. >> What might I be missing to allow the audio ? >> Volume is up. >> >> Thanks >> >> Jerry >> > > > I just noticed the RTP log is sending to 192.168.2.0 which is my local lan > address of the linphone - it should be sending to the NAT address and is > not. > What did I not set correctly ? > I am not using pjsip - but the older asterisk. > > Thanks > > Jerry > >Have you configured chan_sip to know it is behind NAT itself and what its >public IP address is? If not, then you'll get no audio. I'm thinking I have not. What did I miss ? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying asterisk on AWS
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 > allow=gsm > canreinvite=yes > > I enable "rtp set debug on" and the console is printing info. > > The call comes into my linphone softphone - but I get no audio on my > linphone softphone. > What might I be missing to allow the audio ? > Volume is up. > > Thanks > > Jerry > I just noticed the RTP log is sending to 192.168.2.0 which is my local lan address of the linphone - it should be sending to the NAT address and is not. What did I not set correctly ? I am not using pjsip - but the older asterisk. Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trying asterisk on AWS
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 comes into my linphone softphone - but I get no audio on my linphone softphone. What might I be missing to allow the audio ? Volume is up. Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 8.14.0 and multicast sometimes not hear anythign
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 thoughts on why once in a great while nothing would be heard ? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] libpri compile ubuntu 22.04
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:
> > >
> > > 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 zero-length array ‘u_int8_t[0]’ {aka ‘unsigned char[]’}
> > > [-Werror=zero-length-bounds]
> > > 1333 | if ((h->u.ft == 3) && (h->u.m3 == 0) &&
> > (h->u.m2 ==
> > > 0) && (h->u.data[0] == 0x0f)) {
> > >|
> > > ~^~~
> > >
> > > EVEN though I added the "-Werror=zero-length-bounds" to the
> > makefile as you
> > > can see.
> > > How do I get libpri-1.6.0 to compile on Ubuntu 22.04 LTS ?
> > Just informationally, there is an open issue for this[1], but it
> > hasn't
> > been worked on.
> > I am not sure if changing the [0] to [] will do the trick as I
> > haven't
> > been able to look at this yet, but you could try that.
> >
> > [1] https://issues.asterisk.org/jira/browse/PRI-189
> >
> >
> > Problem is it references [0], [1], [2], [3], and [4]. Not just [0], so
> > I don't think so.
> > Thank you.
>
> [0] is a notation used for flexible array members, placed at the end of
> a struct. It does not actually mean the array is of size 0. It seems
> that [0] may have fallen out of use now in favor of [] which is how it
> is defined in the C99 standard.
> I have not tested this in any way whatsoever, but here is a quick
> patch[1] that makes it compile again, that you could try. If you do try
> it, let us know how it works.
>
> [1] https://gerrit.asterisk.org/c/libpri/+/19311
These changes did allow the compile now... Thanks!
jerry
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Re: [asterisk-users] libpri compile ubuntu 22.04
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
> > -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 zero-length array ‘u_int8_t[0]’ {aka ‘unsigned char[]’}
> > [-Werror=zero-length-bounds]
> > 1333 | if ((h->u.ft == 3) && (h->u.m3 == 0) &&
> (h->u.m2 ==
> > 0) && (h->u.data[0] == 0x0f)) {
> >|
> > ~^~~
> >
> > EVEN though I added the "-Werror=zero-length-bounds" to the makefile as
> you
> > can see.
> > How do I get libpri-1.6.0 to compile on Ubuntu 22.04 LTS ?
> Just informationally, there is an open issue for this[1], but it hasn't
> been worked on.
> I am not sure if changing the [0] to [] will do the trick as I haven't
> been able to look at this yet, but you could try that.
>
> [1] https://issues.asterisk.org/jira/browse/PRI-189
Problem is it references [0], [1], [2], [3], and [4]. Not just [0], so I
don't think so.
Thank you.
jerry
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[asterisk-users] libpri compile ubuntu 22.04
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 zero-length array ‘u_int8_t[0]’ {aka ‘unsigned char[]’}
[-Werror=zero-length-bounds]
1333 | if ((h->u.ft == 3) && (h->u.m3 == 0) && (h->u.m2 ==
0) && (h->u.data[0] == 0x0f)) {
|
~^~~
EVEN though I added the "-Werror=zero-length-bounds" to the makefile as you
can see.
How do I get libpri-1.6.0 to compile on Ubuntu 22.04 LTS ?
Thanks
Jerry
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[asterisk-users] Two quick questions
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 that making a full release ? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multicast codec
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on resources
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 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 spikes up the CPU to 270%. > > > > Is there something I dont have set right to take advantage to > > the resourses? > > Thanks > > > > Jerry > > > > Hi Jerry, > > If I recall correctly, there was a talk at an AstriCon or a web page > somewhere that I came across at one point (I'm having a hard time > finding it now) that dove in fairly deep into Asterisk performance > related to multiple cores. > > And if I recall correctly, the conclusion was that the drop-off was > around 8-12 cores -- and beyond that the extra cores aren't doing much > other than helping schedule work and you can't really get more > concurrent calls by adding more cores. > > Someone who is a bit more well-versed in large-machine performance with > Asterisk can certainly chime in here, but from what I gather, throwing > 40 cores at a single Asterisk instance is not the magic bullet to > support a massive number of calls. > > > Thanks Mark, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multicast on asterisk 13.30.0 weird issue.
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 multicast group that device should listen for. How can this be ? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on resources
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 spikes up the CPU to 270%. Is there something I dont have set right to take advantage to the resourses? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Video - how many calls at once using ConfBridge
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 cpu, 8 core, 8 gig ram...Asterisk 18.12.1 etc... Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi
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 support newer kernels
>>
>> yes the install from git works - but I prefer to grab a real release.
> There should hopefully be a new release sometime within the next month,
> according to the DAHDI team at Sangoma (with which I am not affiliated).
> A lot of people are frustrated at the inactivity DAHDI has seen under Sangoma
> in the past couple years. The DAHDI team has acknowledged this and is trying
> to get a release out soon. It's just taken them longer than expected.
> Again, not affiliated with the team, but passing along some info to hopefully
> provide some insight. I'm as anxious as you are as well.
>
> Not that even Git head will not necessarily install on newer kernels. I have
> an install script that adds in some patches to get DAHDI Linux and Tools to
> properly install on the newest systems. Without it, installation will fail,
> even if your base is the "next" branch ("master" / 3.1.0 will fail - same for
> Tools).
Thanks for the update. I’m mainly using Ubuntu 20.04 and 22.04. Hopefully they
will be supported
Jerry
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[asterisk-users] dahdi
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi start up under systemctl
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 Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting new registrations
How does an external program get notification of "new" registrations ? Would that come over the AMI or anything ? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd issue CentOS 7 vs Ubuntu 20.04
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 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 up the polycom phone and dial it connects. >>>> I hear the other ends 'tone" - but when I press digits - >>>> nothing happens (to select a port) >>>> Seems everything is set for rfc2833. >>>> >>>> The devices are a TOA SIP Gateway, and a TOA N-8000 device connected to >>>> the GW. >>>> >>>> I have compared the settings of the polycom extension on both boxes - >>>> they match and also the SIP gateway. >>>> >>>> I tried to compare the sip debug from the Ubuntu to the centos and >>>> "looked" the same to me. >>>> >>>> Where might I look next or what might I look at ? >>>> >>>> Thanks, >>>> >>>> Jerry >>>> >>> >>> >>> ok - if I "rtp set debug on " on the CentOS 7 server I get a tone of >>> logging. >>> >>> if I do the same on the ubuntu 20.04 all i get is like 2 lines. >>> I have done "systemctl stop firewalld" on the ubuntu box - same result. >>> >>> Where do I look next ? >>> >>> Jerry >>> >> >> >> I dont get it - I certainly getting RTP traffic because I defined an >> extension to playback the demo-congrats messages. >> I call that extension - and ALL kinds of RTP traffic prints on teh >> console. >> >> But when I call the one extension - 103 - all it prints is 2 lines. >> >> I also removed the source tree - un tarred - ran the >> contrib/scripts/install_prereq install script, it did install a couple >> packages - I dont think they mattered. >> do the ./configure, make, make install and started up again - same issue >> though. >> >> Jerry >> > > > > So - still on this... > > I was just dialing the SIP Gateway with Dial(SIP/103) > > if I change my Dial command to this: > > Dial(SIP/103,20,D(15)) > So I send out the DTMF in the dial command - this works and connects me > and the DTMF is delivered and I get the right port. > > The problem still remains - Dialing just Dial(SIP/103) from the polycom > phone - and then doing 15 for DTMF does not work. Cant figure out why ? > > Any thoughts ? > > Jerry > This ended up being a simple canreinvite situation... I had yes - and needed to be set to NO. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd issue CentOS 7 vs Ubuntu 20.04
>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 will then open the NAT/firewall. -- Hi Josh - Thanks - I have also turned off the firewall with "systemctl stop firewalld". Did not make a differernce. I am not at the site to change the to rtpkeepalive - will try that monday. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd issue CentOS 7 vs Ubuntu 20.04
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. >>> >>> 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 digits - nothing happens >>> (to select a port) >>> Seems everything is set for rfc2833. >>> >>> The devices are a TOA SIP Gateway, and a TOA N-8000 device connected to >>> the GW. >>> >>> I have compared the settings of the polycom extension on both boxes - >>> they match and also the SIP gateway. >>> >>> I tried to compare the sip debug from the Ubuntu to the centos and >>> "looked" the same to me. >>> >>> Where might I look next or what might I look at ? >>> >>> Thanks, >>> >>> Jerry >>> >> >> >> ok - if I "rtp set debug on " on the CentOS 7 server I get a tone of >> logging. >> >> if I do the same on the ubuntu 20.04 all i get is like 2 lines. >> I have done "systemctl stop firewalld" on the ubuntu box - same result. >> >> Where do I look next ? >> >> Jerry >> > > > I dont get it - I certainly getting RTP traffic because I defined an > extension to playback the demo-congrats messages. > I call that extension - and ALL kinds of RTP traffic prints on teh console. > > But when I call the one extension - 103 - all it prints is 2 lines. > > I also removed the source tree - un tarred - ran the > contrib/scripts/install_prereq install script, it did install a couple > packages - I dont think they mattered. > do the ./configure, make, make install and started up again - same issue > though. > > Jerry > So - still on this... I was just dialing the SIP Gateway with Dial(SIP/103) if I change my Dial command to this: Dial(SIP/103,20,D(15)) So I send out the DTMF in the dial command - this works and connects me and the DTMF is delivered and I get the right port. The problem still remains - Dialing just Dial(SIP/103) from the polycom phone - and then doing 15 for DTMF does not work. Cant figure out why ? Any thoughts ? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd issue CentOS 7 vs Ubuntu 20.04
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 >> 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 digits - nothing happens >> (to select a port) >> Seems everything is set for rfc2833. >> >> The devices are a TOA SIP Gateway, and a TOA N-8000 device connected to >> the GW. >> >> I have compared the settings of the polycom extension on both boxes - >> they match and also the SIP gateway. >> >> I tried to compare the sip debug from the Ubuntu to the centos and >> "looked" the same to me. >> >> Where might I look next or what might I look at ? >> >> Thanks, >> >> Jerry >> > > > ok - if I "rtp set debug on " on the CentOS 7 server I get a tone of > logging. > > if I do the same on the ubuntu 20.04 all i get is like 2 lines. > I have done "systemctl stop firewalld" on the ubuntu box - same result. > > Where do I look next ? > > Jerry > I dont get it - I certainly getting RTP traffic because I defined an extension to playback the demo-congrats messages. I call that extension - and ALL kinds of RTP traffic prints on teh console. But when I call the one extension - 103 - all it prints is 2 lines. I also removed the source tree - un tarred - ran the contrib/scripts/install_prereq install script, it did install a couple packages - I dont think they mattered. do the ./configure, make, make install and started up again - same issue though. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd issue CentOS 7 vs Ubuntu 20.04
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 up the polycom phone and dial it connects. > I hear the other ends 'tone" - but when I press digits - nothing happens > (to select a port) > Seems everything is set for rfc2833. > > The devices are a TOA SIP Gateway, and a TOA N-8000 device connected to > the GW. > > I have compared the settings of the polycom extension on both boxes - they > match and also the SIP gateway. > > I tried to compare the sip debug from the Ubuntu to the centos and > "looked" the same to me. > > Where might I look next or what might I look at ? > > Thanks, > > Jerry > ok - if I "rtp set debug on " on the CentOS 7 server I get a tone of logging. if I do the same on the ubuntu 20.04 all i get is like 2 lines. I have done "systemctl stop firewalld" on the ubuntu box - same result. Where do I look next ? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd issue CentOS 7 vs Ubuntu 20.04
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 digits - nothing happens (to select a port) Seems everything is set for rfc2833. The devices are a TOA SIP Gateway, and a TOA N-8000 device connected to the GW. I have compared the settings of the polycom extension on both boxes - they match and also the SIP gateway. I tried to compare the sip debug from the Ubuntu to the centos and "looked" the same to me. Where might I look next or what might I look at ? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge user joining not getting video
> > > 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 that - doing better - its negotiating video now - but still not showing me video for a conference. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge user joining not getting video
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: > 4013c15f1f4cdff (ulaw|h264) No Tx: ACK > so clearly the caller has h264. > > Then when I "automatically" request another softphone to join my conf > bridge... > the soft phone rings, and answers - all I get is audio and sip show > channels for that device: > 5c77cf1455e4afc (ulaw) No Tx: ACK > > How do I get Video in the confbridge ? > > Thanks > > Jerry > hi Josh, here is the sip debug... It shows the the first call negotiate video - but the second call to bring the end video device into the conf - no video negotitation. Audio is at 15542 Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Thanks, Jerry Asterisk 18.8.0, Copyright (C) 1999 - 2021, Sangoma Technologies Corporation and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Running as user 'silentm' Running under group 'silentm' Connected to Asterisk 18.8.0 currently running on DevKaufer (pid = 597669) Really destroying SIP dialog 'c24843e8-d7f1-0740-08dd-8b79fe39a15a' Method: REGISTER <--- SIP read from UDP:192.168.2.22:5060 ---> <-> == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 Audio is at 17816 Video is at 192.168.1.6:10746 Adding video codec vp8 to SDP Adding codec ulaw to SDP Adding codec opus to SDP Reliably Transmitting (NAT) to 192.168.1.6:48124: INVITE sips:mason.kaufer.visualcampus@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss SIP/2.0 Via: SIP/2.0/WS 192.168.1.6:5060;branch=z9hG4bK73b689b9;rport Max-Forwards: 70 From: "Mason Kaufer 34" ;tag=as101db932 To: Contact: Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 18.8.0 Date: Thu, 13 Jan 2022 13:46:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 1106 v=0 o=root 1174630673 1174630673 IN IP4 192.168.1.6 s=Asterisk PBX 18.8.0 c=IN IP4 192.168.1.6 b=CT:5120 t=0 0 m=audio 17816 UDP/TLS/RTP/SAVPF 0 107 a=rtpmap:0 PCMU/8000 a=rtpmap:107 opus/48000/2 a=maxptime:60 a=ice-ufrag:4ff9bfd157a3896a6bc7f86d312dde00 a=ice-pwd:2c8b8f052875a1cd7096d71478ff3567 a=candidate:Hc0a80106 1 UDP 2130706431 192.168.1.6 17816 typ host a=candidate:Hc0a80106 2 UDP 2130706430 192.168.1.6 17817 typ host a=connection:new a=setup:passive a=fingerprint:SHA-256 0D:4D:96:97:A7:EB:A2:62:4E:43:10:C9:64:9E:5D:43:C5:00:08:2D:D7:1D:01:0C:83:99:B9:49:77:64:24:AF a=rtcp-mux a=sendrecv m=video 10746 UDP/TLS/RTP/SAVPF 100 a=ice-ufrag:0c2c1ae221c4578666475d5455d11e6f a=ice-pwd:7eadf35c40a33c2f2bd87431669b60b2 a=candidate:Hc0a80106 1 UDP 2130706431 192.168.1.6 10746 typ host a=candidate:Hc0a80106 2 UDP 2130706430 192.168.1.6 10747 typ host a=connection:new a=setup:passive a=fingerprint:SHA-256 0D:4D:96:97:A7:EB:A2:62:4E:43:10:C9:64:9E:5D:43:C5:00:08:2D:D7:1D:01:0C:83:99:B9:49:77:64:24:AF a=rtpmap:100 VP8/9 a=rtcp-fb:* ccm fir a=rtcp-mux a=sendrecv --- -- Called mason.kaufer.visualcampus <--- SIP read from WS:192.168.1.6:48124 ---> SIP/2.0 100 Trying (sent from the Transaction Layer) Via: SIP/2.0/WS 192.168.1.6:5060;rport=5060;branch=z9hG4bK73b689b9 From: "Mason Kaufer 34";tag=as101db932 To: Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060 CSeq: 102 INVITE Content-Length: 0 <-> --- (7 headers 0 lines) --- <--- SIP read from WS:192.168.1.6:48124 ---> SIP/2.0 180 Ringing Via: SIP/2.0/WS 192.168.1.6:5060;rport=5060;branch=z9hG4bK73b689b9 From: "Mason Kaufer 34";tag=as101db932 To: ;tag=HULiDWhvD78SNfAPBUqC Contact: Call-ID: 4c4ab96b31b9fa52416eacf563ae2bf1@192.168.1.6:5060 CSeq: 102 INVITE Content-Length: 0 Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE <-> --- (9 headers 0 lines) --- sip_route_dump: route/path hop: -- SIP/mason.kaufer.visualcampus-004b is ringing > 0x7f8eac0141f0 -- Strict RTP learning after remote address set to: 192.168.1.6:56634 > 0x7f8eac00b800 -- Strict RTP learning after remote address set to: 192.168.1.6:32953 <--- SIP read from WS:192.168.1.6:48124 ---> SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.1.6:5060;rport=5060;branch=z9hG4bK73b689b9 From: "Mason Kaufer 34
[asterisk-users] ConfBridge user joining not getting video
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 when I "automatically" request another softphone to join my conf bridge... the soft phone rings, and answers - all I get is audio and sip show channels for that device: 5c77cf1455e4afc (ulaw) No Tx: ACK How do I get Video in the confbridge ? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extensions.conf asterisk 18.8.0 question
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 silentm silentm 3 Jan 10 14:02 /tmp/test.incoming.txt
Thanks!
Jerry
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[asterisk-users] sipml5
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sipml5
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 - and plugged into the
javascript function below
The console log says 403 forbidden.
Is there a trick to get the API working ?
Any pointers to share ? Thanks.
Jerry
SIPml.init(
function(e){
var stack = new SIPml.Stack({realm: 'example.org',
impi: 'bob', impu: 'sip:b...@example.org', password: 'mysecret',
events_listener: { events: 'started', listener:
function(e){
var callSession =
stack.newSession('call-audiovideo', {
video_local:
document.getElementById('video-local'),
video_remote:
document.getElementById('video-remote'),
audio_remote:
document.getElementById('audio-remote')
});
callSession.call('alice');
}
}
});
stack.start();
}
);
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[asterisk-users] sipml5 how many video connections
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18 with SIPml5 - Wrong password
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 checked that I entered the correct password on the website, > I can see the password on Asterisk sip.conf - Why would I get this wrong > password message? > > Thanks > > Jerry > Hello all - I continue to not be able to login with the web page above - and asterisk 18.8.0 for SIPML5. I am using chan_sip still on this one - and I have enabled websocket_enabled=yes in my sip.conf file. I also have a valid dtlscertfile= and that seems to be working fine. All I see in the asterisk message files is : [Nov 17 15:05:09] NOTICE[23632] chan_sip.c: Registration from '"SIPMl5"' failed for 'IP' - Wrong password What might I be missing to get this to work? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 18 with SIPml5 - Wrong password
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 sip.conf - Why would I get this wrong password message? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk bring in RTP audio
>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. I can perhaps see 'sending" audio out - but I'm looking for both directions. Would this be the syntax ? Channel: UnicastRTP/basic/IP:port Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk bring in RTP audio
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] originate call in dial plan to join confbridge
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 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 => 63(join_conf),n,ConfBridge(63,MyConfBridge,MyConfUser)
> > >
> > > The problem is this waits till 401 answers -before calling 402
> > and so
> > > forth.
> > > I need to "call" each one "now" and bring into the conference.
> > >
> > > How do I do that ? I want all 3 ringing at the same time - and
> > then as
> > > they answer they are brought into the conference.
> >
> > Use the "a" option for asynchronous operation.
> > e.g. exten => 63,1,Originate(SIP/401,exten,63,join_conf,,a)
> >
> > > Thanks
> > >
> > > Jerry
> >
> >
> > I tried this and got an error:
> > Originate("SIP/122-0021", "SIP/401,exten,63,join_conf,,a") in new
> > stack
> > [Sep 29 16:35:48] NOTICE[49708][C-27b0]: app_originate.c:222
> > originate_exec: Invalid timeout: 'a'. Setting timeout to 30 second
> Whoops, you need 2 commas actually, not 1, since you didn't specify a
> priority.
> Full usage explained here:
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Application_Originate
Awesome this does work.
Originate(SIP/401,exten,63,join_conf,,,a)
jerry
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Re: [asterisk-users] originate call in dial plan to join confbridge
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,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 => 63(join_conf),n,ConfBridge(63,MyConfBridge,MyConfUser)
> >
> > The problem is this waits till 401 answers -before calling 402 and so
> > forth.
> > I need to "call" each one "now" and bring into the conference.
> >
> > How do I do that ? I want all 3 ringing at the same time - and then as
> > they answer they are brought into the conference.
>
> Use the "a" option for asynchronous operation.
> e.g. exten => 63,1,Originate(SIP/401,exten,63,join_conf,,a)
>
> > Thanks
> >
> > Jerry
>
>
I tried this and got an error:
Originate("SIP/122-0021", "SIP/401,exten,63,join_conf,,a") in new stack
[Sep 29 16:35:48] NOTICE[49708][C-27b0]: app_originate.c:222
originate_exec: Invalid timeout: 'a'. Setting timeout to 30 second
Jerry
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[asterisk-users] originate call in dial plan to join confbridge
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 => 63(join_conf),n,ConfBridge(63,MyConfBridge,MyConfUser) The problem is this waits till 401 answers -before calling 402 and so forth. I need to "call" each one "now" and bring into the conference. How do I do that ? I want all 3 ringing at the same time - and then as they answer they are brought into the conference. Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIPml5
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 https://www.doubango.org/sipml5/call.htm and I click Login... I get this error on asterisk /var/log/asterisk/messages [Sep 27 14:58:08] NOTICE[46486] chan_sip.c: Registration from '"SIPMl5"< sip:jerry.g...@somewhere.com>' failed for 'IP:37994' - Wrong password my sip.conf entry is this: [jerry.g...@somewhere.com] type=friend defaultname=jerry.g...@somewhere.com defaultuser=jerry.g...@somewhere.com secret=(matches that on the website page) encryption=yes avpf=yes force_avpf=yes dtlsenable=yes dtlsverify=fingerprint dtlscertfile=/etc/letsencrypt/live/mypath/asterisk.pem dtmfmode=RFC2833 host=dynamic description=Test context=sipml5 qualify=yes rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 callerid="Jerry.Geis" qualify=no canreinvite=yes timezone=1 nat=force_rport,comedia disallow=all allow=ulaw allow=alaw allow=gsm Not sure what I dont have that is not accespting the PW. I am using 18.6.0 and still on the OLD chan_sip.c (which is working fine for everything else). Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up sipml5
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/letsencrypt/live/mpname/fullchain.pem > > The SIPMl log just says: > WebSocket connection to 'wss://myIP:8088/' failed: > > Is there something easy I'm missing to allow websockets on Asterisk ? > Thanks > > Jerry > Kevin - THANKS - my url was missing the "/ws" at the end it should have been "wss://myip:8088/ws". Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting up sipml5
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: Is there something easy I'm missing to allow websockets on Asterisk ? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk18.6.0 with old chan_sip and microsoft teams
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. > > Does Asterisk 18.6.0 work with Microsoft teams ? my issue is no audio. > > I have done a tcpdump -i eth0 and I see the audio traffic going back on > that network - so I presume I am good - What can I do so teams sees the > audio ? > > Thanks, > > Jerry > > The configuration is this: "initiating a call from Teams, that call gets routed through an Audiocodes 2600 SBC. That 2600 SBC then passes the call over to the Marion Audiocodes Mediant 1000. " So teams -> 2600 SBC -> Mediant 1000 -> Asterisk 18.6.0 - Call does come in - looks normal but they report no audio heard back at source. Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk18.6.0 with old chan_sip and microsoft teams
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 done a tcpdump -i eth0 and I see the audio traffic going back on that network - so I presume I am good - What can I do so teams sees the audio ? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failed to authenticate
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 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 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 has <8124> >>>>> [Jul 27 10:08:50] NOTICE[89791][C-0068] chan_sip.c: Failed to >>>>> authenticate device "USCOL TEST" ;tag=1c1947164290 for >>>>> INVITE, code = -2 >>>>> >>>>> How I allow this ? I want to allow any SIP call to 3381. >>>>> Using Astering 18.4.0 >>>>> >>>>> Thanks, >>>>> >>>>> Jerry >>>>> >>>> >>>> Sure here it is: >>>> [general](+) >>>> register => 3382:XX@IP/3382 >>>> >>>> ; Description: Connection to PBX >>>> [3382] >>>> type=friend >>>> defaultname=3382 >>>> defaultuser=3382 >>>> secret=XX >>>> dtmfmode=RFC2833 >>>> host=IP >>>> description=Connection to PBX >>>> context=incoming >>>> rtptimeout=60 >>>> rtpholdtimeout=60 >>>> rtpkeepalive=60 >>>> callerid=3382 >>>> qualify=no >>>> canreinvite=no >>>> nat=never >>>> disallow=all >>>> allow=ulaw >>>> allow=alaw >>>> allow=gsm >>>> >>>> Thanks >>>> Jerry >>>> >>>> >>> > What's the association between 3381 and 3382? >>> >>> 3381 is the number they want to dial into my asterisk. 3382 is the >>> registered extension to their system. >>> >>> Jerry >>> >>> >>> >>>> >>>> >>> >> >You register as 3382. That means that if someone on their system dials >> 3382, >> >your Asterisk server gets the call. >> >> >> I think at first I was only using 3381. That was the extension I >> registered. There was no 3382. Something was going wrong there also. >> (Might have been a similar error), >> and I could not get that to work either. >> >> Jerry >> > > > Well my issue has changed now. I have dropped the 3382. Changed back to > 3381. So I am registering 3381 to the other server. > The other server is 10.35.229.5. My IP is 10.35.229.11. > I have two network cards. > > 10.35.229.11 is Eth0 > 192.168.1.60 is Eth1 > > route looks OK > route -n > Kernel IP routing table > Destination Gateway Genmask Flags Metric RefUse > Iface > 0.0.0.0 192.168.1.1 0.0.0.0 UG0 00 > eth1 > 10.35.229.0 0.0.0.0 255.255.255.0 U 0 00 > eth0 > 169.254.0.0 0.0.0.0 255.255.0.0 U 1002 00 > eth0 > 169.254.0.0 0.0.0.0 255.255.0.0 U 1003 00 > eth1 > 192.168.1.0 0.0.0.0 255.255.255.0 U 0 00 > eth1 > > The issue is that the call comes in but the user hears no audio. > There is any crazy networking going on - why would the user not hear audio > ? > Thanks > > Jerry > Hello All, I got more information about the "no audio". The incoming call is from 10.37.229.5 - I have two network cards in the box. 10.35.229.11 eth0 192.168.1.60 eth1 When I noticed the incoming address was 10.37.229.5 I thought the audio packets are sending out the default route of eth1. SO I tried to add a route: route -n Kernel IP routing table Destination Gateway Genmask Flags Metric RefUse Iface 0.0.0.0 192.168.1.1 0.0.0.0 UG0 00 eth1 10.35.229.0 0.0.0.0 255.255.255.0 U 0 00 eth0 10.37.229.0 0.0.0.0 255.255.255.0 U 0 00 eth0 169.254.0.0 0.0.0.0 255.255.0.0 U 1002 00 eth0 169.254.0.0 0.0.0.0 255.255.0.0 U 1003 00 eth1 192.168.1.0 0.0.0.0 255.255.255.0 U 0 00 eth1 But I am still not getting audio. Anything else I might try ? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failed to authenticate
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: >>> >>>> 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 has <8124> >>>> [Jul 27 10:08:50] NOTICE[89791][C-0068] chan_sip.c: Failed to >>>> authenticate device "USCOL TEST" ;tag=1c1947164290 for >>>> INVITE, code = -2 >>>> >>>> How I allow this ? I want to allow any SIP call to 3381. >>>> Using Astering 18.4.0 >>>> >>>> Thanks, >>>> >>>> Jerry >>>> >>> >>> Sure here it is: >>> [general](+) >>> register => 3382:XX@IP/3382 >>> >>> ; Description: Connection to PBX >>> [3382] >>> type=friend >>> defaultname=3382 >>> defaultuser=3382 >>> secret=XX >>> dtmfmode=RFC2833 >>> host=IP >>> description=Connection to PBX >>> context=incoming >>> rtptimeout=60 >>> rtpholdtimeout=60 >>> rtpkeepalive=60 >>> callerid=3382 >>> qualify=no >>> canreinvite=no >>> nat=never >>> disallow=all >>> allow=ulaw >>> allow=alaw >>> allow=gsm >>> >>> Thanks >>> Jerry >>> >>> >> > What's the association between 3381 and 3382? >> >> 3381 is the number they want to dial into my asterisk. 3382 is the >> registered extension to their system. >> >> Jerry >> >> >> >>> >>> >> > >You register as 3382. That means that if someone on their system dials > 3382, > >your Asterisk server gets the call. > > > I think at first I was only using 3381. That was the extension I > registered. There was no 3382. Something was going wrong there also. > (Might have been a similar error), > and I could not get that to work either. > > Jerry > Well my issue has changed now. I have dropped the 3382. Changed back to 3381. So I am registering 3381 to the other server. The other server is 10.35.229.5. My IP is 10.35.229.11. I have two network cards. 10.35.229.11 is Eth0 192.168.1.60 is Eth1 route looks OK route -n Kernel IP routing table Destination Gateway Genmask Flags Metric RefUse Iface 0.0.0.0 192.168.1.1 0.0.0.0 UG0 00 eth1 10.35.229.0 0.0.0.0 255.255.255.0 U 0 00 eth0 169.254.0.0 0.0.0.0 255.255.0.0 U 1002 00 eth0 169.254.0.0 0.0.0.0 255.255.0.0 U 1003 00 eth1 192.168.1.0 0.0.0.0 255.255.255.0 U 0 00 eth1 The issue is that the call comes in but the user hears no audio. There is any crazy networking going on - why would the user not hear audio ? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failed to authenticate
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 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 has <8124> >>> [Jul 27 10:08:50] NOTICE[89791][C-0068] chan_sip.c: Failed to >>> authenticate device "USCOL TEST" ;tag=1c1947164290 for >>> INVITE, code = -2 >>> >>> How I allow this ? I want to allow any SIP call to 3381. >>> Using Astering 18.4.0 >>> >>> Thanks, >>> >>> Jerry >>> >> >> Sure here it is: >> [general](+) >> register => 3382:XX@IP/3382 >> >> ; Description: Connection to PBX >> [3382] >> type=friend >> defaultname=3382 >> defaultuser=3382 >> secret=XX >> dtmfmode=RFC2833 >> host=IP >> description=Connection to PBX >> context=incoming >> rtptimeout=60 >> rtpholdtimeout=60 >> rtpkeepalive=60 >> callerid=3382 >> qualify=no >> canreinvite=no >> nat=never >> disallow=all >> allow=ulaw >> allow=alaw >> allow=gsm >> >> Thanks >> Jerry >> >> > > What's the association between 3381 and 3382? > > 3381 is the number they want to dial into my asterisk. 3382 is the > registered extension to their system. > > Jerry > > > >> >> > >You register as 3382. That means that if someone on their system dials 3382, >your Asterisk server gets the call. I think at first I was only using 3381. That was the extension I registered. There was no 3382. Something was going wrong there also. (Might have been a similar error), and I could not get that to work either. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failed to authenticate
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 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 has <8124> >> [Jul 27 10:08:50] NOTICE[89791][C-0068] chan_sip.c: Failed to >> authenticate device "USCOL TEST" ;tag=1c1947164290 for >> INVITE, code = -2 >> >> How I allow this ? I want to allow any SIP call to 3381. >> Using Astering 18.4.0 >> >> Thanks, >> >> Jerry >> > > Sure here it is: > [general](+) > register => 3382:XX@IP/3382 > > ; Description: Connection to PBX > [3382] > type=friend > defaultname=3382 > defaultuser=3382 > secret=XX > dtmfmode=RFC2833 > host=IP > description=Connection to PBX > context=incoming > rtptimeout=60 > rtpholdtimeout=60 > rtpkeepalive=60 > callerid=3382 > qualify=no > canreinvite=no > nat=never > disallow=all > allow=ulaw > allow=alaw > allow=gsm > > Thanks > Jerry > > > What's the association between 3381 and 3382? 3381 is the number they want to dial into my asterisk. 3382 is the registered extension to their system. Jerry > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failed to authenticate
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 10:08:50] WARNING[89791][C-0068] chan_sip.c: username > mismatch, have <3381>, digest has <8124> > [Jul 27 10:08:50] NOTICE[89791][C-0068] chan_sip.c: Failed to > authenticate device "USCOL TEST" ;tag=1c1947164290 for > INVITE, code = -2 > > How I allow this ? I want to allow any SIP call to 3381. > Using Astering 18.4.0 > > Thanks, > > Jerry > Sure here it is: [general](+) register => 3382:XX@IP/3382 ; Description: Connection to PBX [3382] type=friend defaultname=3382 defaultuser=3382 secret=XX dtmfmode=RFC2833 host=IP description=Connection to PBX context=incoming rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 callerid=3382 qualify=no canreinvite=no nat=never disallow=all allow=ulaw allow=alaw allow=gsm Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Failed to authenticate
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 has <8124> [Jul 27 10:08:50] NOTICE[89791][C-0068] chan_sip.c: Failed to authenticate device "USCOL TEST" ;tag=1c1947164290 for INVITE, code = -2 How I allow this ? I want to allow any SIP call to 3381. Using Astering 18.4.0 Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pjsip in 18.5.X
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
