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sm0svx / svxlink

Advanced repeater system software with EchoLink support for Linux including a GUI, Qtel - the Qt EchoLink client
http://svxlink.org/
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SVXLink and Asterisk ??? #250

Closed sp2ong closed 5 years ago

sp2ong commented 7 years ago

What about possibility connect svxlink with asterisk PBX VoIP ?? We have a lot of running asterisk PBX in Europe

https://www.afu.rwth-aachen.de/dundicrawler/

it will be nice to have possibility connect to local echolink node from VoIP hardware phone. And from echolink node use DTMF can connect to VoIP phone.

I have found on video which show this possibility:

https://www.youtube.com/watch?v=ym4WwEuQYxk

Anybody know how to configure svxlink / svxserver and asterisk to do this ???

dl1hrc commented 7 years ago

Am 22.02.2017 um 19:26 schrieb Waldek:

What about possibility connect svxlink with asterisk PBX VoIP ??

working on it...

73s de Adi / DL1HRC

We have a lot of running asterisk PBX in Europe

https://www.afu.rwth-aachen.de/dundicrawler/

it will be nice to have possibility connect to local echolink node from VoIP hardware phone. And from echolink node use DTMF can connect to VoIP phone.

I have found on video which show this possibility:

https://www.youtube.com/watch?v=ym4WwEuQYxk

Anybody know how to configure svxlink / svxserver and asterisk to do this ???

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sp2ong commented 7 years ago

Ok good news :-)

73 Waldek SP2ONG

dl1hrc commented 7 years ago

Am 22.02.2017 um 19:42 schrieb Waldek:

Ok good news :-)

73 Waldek SP2ONG

Hello Waldek,

first tests were sucessful. It's in a very early state so some errors and problems may still occur! Here a very short howto for people who know what they are doing. Will try to make a documentation (at first in German on http://svxlink.de) when I find the time.

You need to have asterisk and SvxLink running on the same hardware. Audio will be routed through alsa loopback devices. With "modprobe snd-aloop" you get up to 8 alsa lb devices:

Karte 2: Loopback [Loopback], Gerät 0: Loopback PCM [Loopback PCM] Sub-Geräte: 7/8 Sub-Gerät #0: subdevice #0 Sub-Gerät #1: subdevice #1 Sub-Gerät #2: subdevice #2 Sub-Gerät #3: subdevice #3 Sub-Gerät #4: subdevice #4 Sub-Gerät #5: subdevice #5 Sub-Gerät #6: subdevice #6 Sub-Gerät #7: subdevice #7 Karte 2: Loopback [Loopback], Gerät 1: Loopback PCM [Loopback PCM] Sub-Geräte: 8/8 Sub-Gerät #0: subdevice #0 Sub-Gerät #1: subdevice #1 Sub-Gerät #2: subdevice #2 Sub-Gerät #3: subdevice #3 Sub-Gerät #4: subdevice #4 Sub-Gerät #5: subdevice #5 Sub-Gerät #6: subdevice #6 Sub-Gerät #7: subdevice #7

in Asterisk:

Look for asterisk "Console" configuration. If you compile it, then enable support for alsa and portaudio. In asterisk/alsa.conf define the audio ports pointing to alsa loopback devices:

input_device=hw:2,0,0 output_device=hw:2,0,1

in asterisk/extensions.conf you need a reference to call the console when an incoming call is received:

exten => 2###0,1,Dial(Console/default) exten => 2###0,n,Hangup

SvxLink:

Define an own Logic for Asterisk as a repeater, e.g.:

[AsteriskLogic] TYPE=Repeater RX=RxAsterisk TX=TxAsterisk OPEN_ON_SQL=100 OPEN_ON_DTMF=* DEFAULT_LANG=de_DE OPEN_SQL_FLANK=OPEN IDLE_TIMEOUT=10 NO_REPEAT=1 CALLSIGN=DB0HRC-A EVENT_HANDLER=/usr/share/svxlink/events.tcl DEFAULT_LANG=de_DE

[RxAsterisk] TYPE=Local AUDIO_DEV=alsa:plughw:2,1,1 AUDIO_CHANNEL=0 SQL_DET=VOX SQL_START_DELAY=0 SQL_DELAY=0 SQL_HANGTIME=1300 VOX_FILTER_DEPTH=20 VOX_THRESH=1000 DEEMPHASIS=0 SQL_TAIL_ELIM=0 DTMF_DEC_TYPE=INTERNAL DTMF_MUTING=1 DTMF_HANGTIME=100 DTMF_SERIAL=/dev/ttyS0 DTMF_MAX_FWD_TWIST=18 DTMF_MAX_REV_TWIST=12

[TxAsterisk] TYPE=Local AUDIO_DEV=alsa:plughw:2,1,0 AUDIO_CHANNEL=0 PTT_TYPE=NONE TIMEOUT=9000 TX_DELAY=0 PREEMPHASIS=0

Connect the AsteriskLogic and your RepeaterLogic by LogicLinking, remember there is no security at the moment (e.g. PIN-query).

Good luck, vy 73s de Adi / DL1HRC

sp2ong commented 7 years ago

Hi Adi,

many thanks for info and no problem to for me to move svxlink to physical one server. I have asterisk v13.x own compiled and not problem to recompile with alsa and portaudio support

I will try to do according to your suggestion

73 Waldek SP2ONG

dl1hrc commented 7 years ago

Hi Waldek,

it was my first contact with asterisk two days ago, so I'm an absolute beginner here and very interested in having further information on pin queries, incoming call handling, e.g. call to 555-1234-001 connects to DB0HRC, 555-1234-002 to DB0HAL and so on. Please let me know what you find out, can also be that alsa loopback's are not the best way.

vy 73s de Adi / DL1HRC

sp2ong commented 7 years ago

Ok I will be contact with you when I make progress with my configuration or find problems

73 Waldek SP2ONG

sp2ong commented 7 years ago

Hi Adi,

Your proposal is nice when we have possibility move/run echolink on one PC but in many cases echolink nodes are running on small terminal and mounted very close to radio installation. But asterisk PBX in many cases are part of HamNET infrastructure and installed on more powerful CPU server for asterisk for this reason we need still second solution to have possibility connect echonlink node via network with asterisk. Or write small addon to echolink similar svxserver 'asterlink' use IAX technology and OPUS codec to connect with asterisk or write addon to asterisk similar like Allstar Link. The Allstar link unfortunately base on old version asterisk 1.x we are use news version asterisk 13.x or 14.x .

I would like thank you that you have find solution for echolink and asterisk installed on one PC but maybe in future we find solution connect via network echolink with asterisk

In this weekend I will be try install svxlink on my Asterisk PBX server to see how this solution working

73 Waldek SP2ONG

sm0svx commented 7 years ago

You could do a setup where you run a RemoteTrx on the Asterisk system and the main SvxLink system somewhere completely different. On the RemoteTrx side use the same rx and tx config as Adi suggest.

dl1hrc commented 7 years ago

Yes, I think it's a good solution to do it this way, since the native implementation of protocols like IAX2, SIP, ... into SvxLink needs a lot of external libraries and of course much more manpower in SvxLink development. At the moment I'm looking for a solution to connect the alsa device to a chat room (bridge) while asterisk is starting up. A pin queue (to access the chat room from the phone line) is the condition to a asterisk<->SvxLink interconnect. 73s de Adi / DL1HRC

sp2ong commented 7 years ago

Hi,

Adi, connected svxlink to conference bridge will be nice but I think only allow speak with other who connected to conference root from asterisk users.

But I think maybe not now but in near future, we have possibility interconnect svxlink with asterisk. When I will be use only radio and connect to echolink node which have interconnect with asterisk I can send via DTMF for example your phone VoIP number 315310427323 ( if I know that you have and use asterisk https://www.oe2wnl.at/calltodtmf-voip.php?call=DL1HRC&submit=Convert ) and I will be can speak with you from may handy radio via echolink node i Poland Torun with you, where you will be speak to your hardware/software VoIP asterik phone in Germany :-) Our asterisk PBX in HamNEt Europe use dundi to interconnect with other asterisk PBX for this reason I can select voip phone number on my local asterisk to any hamradio operator in HamNET Europe to have possibility speak with other hams in Europe. We use 2 different voip technology like echolink and asterisk (SIP/IAX) and only we need merge these. I have wrote that mayb we need look to Allstar sources to find idea or other to do this because Allstar is hybrid base on asterisk where exist asterisk SIP/IAX, Allstar node and have possibility configure as echolink node. But one problem with Allstart that use very old asterisk 1.x and don't exist app_rpt module to new version asterisk

I know that it is not easy task to do.

73 Waldek SP2ONG

sp2ong commented 7 years ago

I have found app_rpt sources in svn allstar link:

http://svn.ohnosec.org/svn/projects/allstar/astsrc-1.4.23-pre/trunk/asterisk/apps/app_rpt.c

sp2ong commented 7 years ago

Hi Adi,

I have try to do following with your instruction on:

http://svxlink.de/?page_id=3411

but I have following errors:

=============================================================== SvxLink v1.5.99.5 Copyright (C) 2003-2017 Tobias Blomberg / SM0SVX

SvxLink comes with ABSOLUTELY NO WARRANTY. This is free software, and you are welcome to redistribute it in accordance with the terms and conditions in the GNU GPL (General Public License) version 2 or later.

Using configuration file: /etc/svxlink/svxlink.conf

Starting logic: SimplexLogic Loading RX: Rx1 Loading TX: Tx1 Loading module "ModuleHelp" into logic "SimplexLogic" Found /usr/lib/i386-linux-gnu/svxlink/ModuleHelp.so Module Help v1.0.0 starting... Loading module "ModuleParrot" into logic "SimplexLogic" Found /usr/lib/i386-linux-gnu/svxlink/ModuleParrot.so Module Parrot v1.1.1 starting... Loading module "ModuleEchoLink" into logic "SimplexLogic" Found /usr/lib/i386-linux-gnu/svxlink/ModuleEchoLink.so Module EchoLink v1.3.99.0 starting... SimplexLogic: Event handler script successfully loaded.

Starting logic: RepeaterLogic Loading RX: Rx1 Loading TX: Tx1 Loading module "ModuleHelp" into logic "RepeaterLogic" Found /usr/lib/i386-linux-gnu/svxlink/ModuleHelp.so Module Help v1.0.0 starting... Loading module "ModuleParrot" into logic "RepeaterLogic" Found /usr/lib/i386-linux-gnu/svxlink/ModuleParrot.so Module Parrot v1.1.1 starting... Loading module "ModuleEchoLink" into logic "RepeaterLogic" Found /usr/lib/i386-linux-gnu/svxlink/ModuleEchoLink.so Module EchoLink v1.3.99.0 starting... RepeaterLogic: Event handler script successfully loaded.

Starting logic: AsteriskLogic Loading RX: RxAsterisk Loading TX: TxAsterisk AsteriskLogic: Event handler script successfully loaded. *** ERROR: Unable to handle event: AsteriskLogic::startup in logic AsteriskLogic (invalid command name "AsteriskLogic::startup") Activating link AsteriskRepeaterLink EchoLink directory status changed to ON EchoLink directory status changed to ON --- EchoLink directory server message: --- EchoLink Server v2.5.9997

ECHO6: N.Virginia, USA

======================================

My base configuration for this test before changes was: [GLOBAL] LOGICS=SimplexLogic CFG_DIR=svxlink.d TIMESTAMP_FORMAT="%c"

I have add all according your describe in svxlink.conf with change GLOBAL follow: [GLOBAL] LOGICS=SimplexLogic,RepeaterLogic,AsteriskLogic CFG_DIR=svxlink.d TIMESTAMP_FORMAT="%c" LINKS=AsteriskRepeaterLink

dl1hrc commented 7 years ago

yes, forgot to tell you that you need a new Tcl-Namespace "AsteriskLogic"

cp /usr/share/svxlink/events.d/RepeaterLogic.tcl /usr/share/svxlink/events.d/AsteriskLogic.tcl

and change the line at the beginning of the file from

namespace eval RepeaterLogic {

to

namespace eval AsteriskLogic {

vy 73s de Adi

Am 26.02.2017 um 12:31 schrieb Waldek:

Hi Adi, > > I have try to do following with your instruction on: > > http://svxlink.de/?page_id=3411 > > > but I have following errors: > > SvxLink v1.5.99.5 Copyright (C) 2003-2017 Tobias Blomberg / SM0SVX > > SvxLink comes with ABSOLUTELY NO WARRANTY. This is free software, and > you are welcome to redistribute it in accordance with the terms and > conditions in the GNU GPL (General Public License) version 2 or > later. Using configuration file: /etc/svxlink/svxlink.conf > > Starting logic: SimplexLogic Loading RX: Rx1 Loading TX: Tx1 Loading > module "ModuleHelp" into logic "SimplexLogic" Found > /usr/lib/i386-linux-gnu/svxlink/ModuleHelp.so Module Help v1.0.0 > starting... Loading module "ModuleParrot" into logic "SimplexLogic" > Found /usr/lib/i386-linux-gnu/svxlink/ModuleParrot.so Module Parrot > v1.1.1 starting... Loading module "ModuleEchoLink" into logic > "SimplexLogic" Found > /usr/lib/i386-linux-gnu/svxlink/ModuleEchoLink.so Module EchoLink > v1.3.99.0 starting... SimplexLogic: Event handler script successfully > loaded. > > Starting logic: RepeaterLogic Loading RX: Rx1 Loading TX: Tx1 Loading > module "ModuleHelp" into logic "RepeaterLogic" Found > /usr/lib/i386-linux-gnu/svxlink/ModuleHelp.so Module Help v1.0.0 > starting... Loading module "ModuleParrot" into logic "RepeaterLogic" > Found /usr/lib/i386-linux-gnu/svxlink/ModuleParrot.so Module Parrot > v1.1.1 starting... Loading module "ModuleEchoLink" into logic > "RepeaterLogic" Found > /usr/lib/i386-linux-gnu/svxlink/ModuleEchoLink.so Module EchoLink > v1.3.99.0 starting... RepeaterLogic: Event handler script > successfully loaded. > > Starting logic: AsteriskLogic Loading RX: RxAsterisk Loading TX: > TxAsterisk AsteriskLogic: Event handler script successfully loaded. > *** ERROR: Unable to handle event: AsteriskLogic::startup in logic > AsteriskLogic (invalid command name "AsteriskLogic::startup") > Activating link AsteriskRepeaterLink EchoLink directory status > changed to ON EchoLink directory status changed to ON --- EchoLink > directory server message: --- EchoLink Server v2.5.9997 > > > ECHO6: N.Virginia, USA > > My base configuration for this test before changes was: [GLOBAL] > LOGICS=SimplexLogic CFG_DIR=svxlink.d TIMESTAMP_FORMAT="%c" > > I have add all according your describe in svxlink.conf with change > GLOBAL follow: [GLOBAL] > LOGICS=SimplexLogic,RepeaterLogic,AsteriskLogic CFG_DIR=svxlink.d > TIMESTAMP_FORMAT="%c" LINKS=AsteriskRepeaterLink > > — You are receiving this because you commented. Reply to this email > directly, view it on GitHub > https://github.com/sm0svx/svxlink/issues/250#issuecomment-282549588, > or mute the thread > https://github.com/notifications/unsubscribe-auth/AH2kwUcfs_j0HLDuvDJOeV5uh74dvgxPks5rgWKkgaJpZM4MJAzu.

sp2ong commented 7 years ago

Hi Adi,

Ok, thanks now svxlink running without errors. Next, I will be try tune configuration asterisk

73 Waldek SP2ONG

dl1hrc commented 7 years ago

There ist one remaining thing:

Since SvxLink is running with 48k sampling rate (normally) and Asterisk expect 8k, you have to start Asterisk first, then SvxLink. Otherwise you will get an error message from Asterisk.

The better way would be to have an Alsa samplerate converter in the asound.conf.

vy 73s de Adi / DL1HRC

Am 26.02.2017 um 13:03 schrieb Waldek:

Hi Adi,

Ok, thanks now svxlink running without errors. Next, I will be try tune configuration asterisk

73 Waldek SP2ONG

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sp2ong commented 7 years ago

Ok Adi but I have small problem with asterisk , when I add and changed all necessary files for asterisk and restart asterisk in log I have fond following message

[Feb 26 13:12:40] WARNING[1501] channel.c: Already have a handler for type 'Console' [Feb 26 13:12:40] ERROR[1501] chan_alsa.c: Unable to register channel class 'Console'

and asterisk not running It is problem when is loading chan_console.so

Do you have example asound.conf to convert sampling rate ?

dl1hrc commented 7 years ago

Same message her, no idea what's the reason is for it. But it's working well here. I think we have to google for a solution.

BTW: There are a lot of notice messages here since I use the default configuration.

I do not have a asound.conf for now, my test configs were creating errors. Will look later when I have more time :)

Am 26.02.2017 um 13:37 schrieb Waldek:

Ok Adi but I have small problem with asterisk , when I add and changed all necessary files for asterisk and restart asterisk in log I have fond following message

[Feb 26 13:12:40] WARNING[1501] channel.c: Already have a handler for type 'Console' [Feb 26 13:12:40] ERROR[1501] chan_alsa.c: Unable to register channel class 'Console'

and asterisk not running It is problem when is loading chan_console.so

Do you have example asound.conf to convert sampling rate ?

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sp2ong commented 7 years ago

Ok I will ignore this but can you explain more about

exten => 2####0,1,Authenticate(1234) exten => 2####0,2,Playback(access-granted) exten => 2####0,3,Dial(Console/default) exten => 2####0,n,Hangup()

I am not sure what I need add to [default]

My extension.conf

[general] static=yes autofallthrough=yes writeprotect=no clearglobalvars=no autofallthrough=yes

[default] exten => _X.,1,Hangup()

[dundiextens] exten => _7471206000,1,Goto(conference,${EXTEN},1)

[conference] exten => 7471206000,1,Noop() same => n,ConfBridge(room-one,default_bridge,default_user,sample_user_menu)

[lookupdundi] switch => DUNDi/priv

[local] include => dundiextens

exten => _Z.,1,Dial(SIP/${EXTEN},30) same => n,Voicemail(${EXTEN}@voicemail) same => n,Goto(lookupdundi,${EXTEN},1) same => n,Hangup()

exten => 7471206000,1,Noop() same => n,ConfBridge(room-one,default_bridge,default_user,sample_user_menu)

[incomingdundi] exten => _Z.,1,Goto(local,${EXTEN},1)

dl1hrc commented 7 years ago

puuhhh, I'm still a beginner in asterisk ;) You need a call handler like this, where 2####0 stands for your sip number (e.g. 234580)

exten => 2####0,1,Authenticate(1234) exten => 2####0,2,Playback(access-granted) exten => 2####0,3,Dial(Console/default) exten => 2####0,n,Hangup()

At the moment it's not possible to connect the Console to a conference because Dial(Console/default) does not return the connection handler the to be connected to the ConfBridge(xx,yy,zz) I recommend to do it first as it explained in my example to be sure that it's working well. Then we can looking for better solutions.

sp2ong commented 7 years ago

Ok Adi,

If I am understood that any number 2xxxx0 incoming from console for example

212340 or 234230 etc

will be pass to Hangup because we use phone number converted from callsign to DTMF code and are long for exmaple my number is 747120636241 I need make aliases shorter to use via DTMF from radio channel

I am not sure that I need this

exten => 2####0,1,Authenticate(1234) exten => 2####0,2,Playback(access-granted) exten => 2####0,3,Dial(Console/default) exten => 2####0,n,Hangup()

move to [local] part extensions.conf

It is look that I must read documentation asterisk about extensions.conf

73 Waldek SP2ONG

dl1hrc commented 7 years ago

The function I wanted to have first was a simple way to dial-in into hamradio repeater-network from outside (not hamnet). It's just a beginning, asterisk has almost infinite functions and facilities to connect other networks. So have fun to extend the functionality, maybe we will have a direct connection (IAX,...) to asterisk later.

vy 73s de Adi / DL1HRC

sp2ong commented 7 years ago

Adi,

I have try understand how this all working and read about alsa loop. I have found example

http://www.getreu.net/public/downloads/doc/snd-aloop-device/snd-aloop-device.html#connect

and it is look like in asla.conf we need ??? following :

input_device=hw:2,0,0 output_device=hw:2,1,0

I thing in my case extensions.conf I don't need dial plane with 2####0 I have similar configuration like use Frank DL3DCW in hamserverpi but still don't have success with calling VoIP number from echolink node.

73 Waldek SP2ONG

dl1hrc commented 7 years ago

the reason may be the LogicLinking, do you use DTMF_MUTING=1 in your RX-configuration?

You can try a different configuration with a Voter and MultiTx too

Am 26.02.2017 um 20:35 schrieb Waldek:

Adi,

I have try understand how this all working and read about alsa loop. I have found example

http://www.getreu.net/public/downloads/doc/snd-aloop-device/snd-aloop-device.html#connect

and it is look like in asla.conf we need ??? following :

input_device=hw:2,0,0 output_device=hw:2,1,0

I thing in my case extensions.conf I don't need dial plane with 2####0 I have similar configuration like use Frank DL3DCW in hamserverpi but still don't have success with calling VoIP number from echolink node.

73 Waldek SP2ONG

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sp2ong commented 7 years ago

Hi Adi,

I don't use DTM_MUTING=1 in my RX configuration. I have still problem maybe with pass DTMF tones ? Because for tests I have run on server where I have asterisk run svxlink without radio access. I have connect via network use echolink software clinet and send DTMF tones from my online DTMF tones It maybe for this reason I have problem. Configuration of svxlink is simple base on distribution svxlink only add your suggested options to have link with asterisk. Maybe when I send DTMF tones from my echolink clinet software they are not passed to asterisk.

dl1hrc commented 7 years ago

Am 28.02.2017 um 07:00 schrieb Waldek:

Hi Adi,

I don't use DTM_MUTING=1 in my RX configuration. I have still problem maybe with pass DTMF tones ? Because for tests I have run on server where I have asterisk run svxlink without radio access. I have connect via network use echolink software clinet and send DTMF tones from my online DTMF tones It maybe for this reason I have problem. Configuration of svxlink is simple base on distribution svxlink only add your suggested options to have link with asterisk. Maybe when I send DTMF tones from my echolink clinet software they are not passed to asterisk.

Hi Waldek,

I think it's not quite simple since EchoLink normally uses the gsm codec as a voice codec. The tones were not clearly transmitted over this way. You should try to use the message channel for it.

sp2ong commented 7 years ago

Adi,

I am not sure but we can send DTMF codes and work only when we connected directly via radio with echolink node which have interlink with asterisk.

Do you try test that other or you connect to your echolink node not directly via radio but from other echolink node and try make connect to asterisk use VoIP number via DTMF ???

73 Waldek

dl1hrc commented 7 years ago

At the moment I do not test it to dial from RF into the phone network since it's not needed at the moment here. I have a commercial sip contract and for me it's to not safe enough over the air with simple DTMF.

Maybe I can test it locally next time without any RF environment, but it will take some days to realize it.

73 Adi

Am 28.02.2017 um 19:56 schrieb Waldek:

Adi,

I am not sure but we can send DTMF codes and work only when we connected directly via radio with echolink node which have interlink with asterisk.

Do you try test that other or you connect to your echolink node not directly via radio but from other echolink node and try make connect to asterisk use VoIP number via DTMF ???

73 Waldek

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VR2XKP commented 7 years ago

That's great Svxlink worked with Asterisk. However, I would like to ask does the above configuration request svxlink & asterisk install to the same machine (e.g. Raspberry pi 2 or 3), or 2 individual machine.

Tks, Francis

sp2ong commented 7 years ago

Hi Francis,

Adi described ( https://svxlink.de/?page_id=3411 ) asterisk configuration that must be installed on the same computer on which svxlink is running.

73 Waldek

2017-06-02 2:57 GMT+02:00 VR2XKP notifications@github.com:

That's great Svxlink worked with Asterisk. However, I would like to ask does the above configuration request svxlink & asterisk install to the same machine (e.g. Raspberry pi 2 or 3), or 2 individual machine.

Tks, Francis

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sm0svx commented 7 years ago

@VR2XKP: As stated above, it may be possible to use a RemoteTrx if you need to have Asterisk on one computer and the main SvxLink node on another. However, I have not studied Adi's instruction in detail so there may be something that prevent it. Adi can probably clarify this.

dl1hrc commented 7 years ago

I never tried it but I'm sure that you can use the RemoteTrx as well. Since the alsa loopback devices are used either SvxLink or RemoteTrx must be run on the same engine.

sp2ong commented 7 years ago

Wow, nice idea with integration Asterisk on another server - that was my problem where I have asterisk HamNET is running on another server than echolink node. In remotetrx.conf we need only define RxAsterisk and TxAsterisk with loopback sounds for NetUplinkTrx to interact with Asterisk

dl1hrc commented 7 years ago

Waldek, please could you give me a short feedback when it works as expected? Thank you.

sp2ong commented 7 years ago

I have run remotetrx on server Asterisk and link with svxlink server. I have recorded sound file from loopback device defined in asterisk alsa.conf

input_device=hw:2,0,0

use following command:

arecord -r 16000 -f S16_LE -D hw:2,0,0 test.wav

I have play DTMF number via radio and it looks that DTMF numbers were recorded in test.wav file for this reason I know that DTMF is passed from svxlink server to remotrx and appear on loopback device hw:2,0,0 But asterisk doesn't decode or another problem is that I can not connect to the phone number. I can not idea how to debug more asterisk to find a problem. I have use CLI asterisk

asterisk -r vvvvvvvv

but I can not see any messages to help me understand a problem. The alsa.conf and console.conf I have from your description only in the extension configuration I have put for test:

[globals] CONSOLE=Console/default

[default] exten =>12345,1,Dial(SIP/747120636241,30) exten =>12345,n,Hangup()

to redirect short number 12345 to my SIP Hamnet phone number

I have loaded modules chan_console and chan_alsa and no load chan_oss 73 Waldek

dl1hrc commented 7 years ago

What codec do you use between RemoteTrx and SvxLink node? Could you try with RAW again?

sp2ong commented 7 years ago

I have use OPUS but I will try RAW

sp2ong commented 7 years ago

Hi Adi, I have switched to RAW but still is problem decode DTMF by Asterisk. But I need confirmation that Asterisk working correctly via alsa channel. Do you or others have console tools to play DTMF code and send to loopback audio ?? I have found simple tool with wav files

http://sp2pmk.tvk.torun.pl/uploads/dtmf.tar.gz But when play DTMF code to dial number still asterisk don't work. For this tools need mplayer or wvaplay https://sourceforge.net/projects/wavplay/

I have check asterisk that chan_alsa is loaded with

chan_alsa.so ALSA Console Channel Driver 0 Running extended chan_console.so Console Channel Driver 0 Not Running extended

Adi do you have an idea how to check that channel alsa asterisk work without svxlink ?

73 Waldek

sp2ong commented 7 years ago

I have use script dmtf which play numbers

dtmf 1234

but asterisk don't decode via alsa channel input. I have use wav file with sample rate 8kHz and 16 kHz

dl1hrc commented 7 years ago

As far as I know asterisk is working only with a 8k stream over the alsa channel.

sp2ong commented 7 years ago

Ok Adi I have replaced dtmf.tar.gz with wav files to 8 k sample rate if anybody would like use this tools to send directly DTMF code to alsa channel input of asterisk to check that will be decode and dial

sp2ong commented 7 years ago

Adi can you try use this tools to send dtmf code to alsa channel input asterisk to check that on your configuration working ? For play wav files via this script you must install mplayer

sp2ong commented 7 years ago

I am not sure but maybe we need create file /etc/asound.conf with the definition of loopback device which is used in asterisk ???

pcm_slave.hw_loopback { pcm "hw:Loopback,0,2" channels 2 format RAW rate 8000 }

pcm.plug_loopback { type plug slave hw_loopback ttable { 0.0 = 1 0.1 = 1 } }

sp2ong commented 7 years ago

Adi,

I have found information in asterisk sources modelus.conf.sample that: ; Load one of: chan_oss, alsa, or console (portaudio). ; By default, load chan_oss only (automatically). ; load => chan_alsa.so noload => chan_oss.so noload => chan_console.so

If we use ALSA channel we don't need load chan_console and define this module.

With 1.2.x versions of Asterisk, only one console channel can be created, and only one call can be placed to/from the console channel at a given time. If the console channel is in use when an incoming call is routed to it, the destination will be treated as busy. Only one of either the ALSA or OSS channel drivers may be loaded (in modules.conf) at a given time.

in CLI asterisk run command: core show channeltypes Type Description Devicestate Indications Transfer

CBAnn Conference Bridge Announcing Channel no yes no
CBRec Conference Bridge Recording Channel no no no
IAX2 Inter Asterisk eXchange Driver (Ver 2) yes yes yes
SIP Session Initiation Protocol (SIP) yes yes yes
Console ALSA Console Channel Driver no yes no
Local Local Proxy Channel Driver yes yes no

7 channel drivers registered.

dl1hrc commented 7 years ago

Hm, version 1.2.x? Maybe is to old? I'm running 14.3.0 BTW: Sorry, I'm quite busy at the moment an can only answer with a high delay :/

sp2ong commented 7 years ago

I have use asterisk v13..0.8 and when I have load only chan_alsa I don't have any erroros about channel console and from CLI I can dial via alsa for example:

console dial 1234

It is loo that for asterisk version 1.2 and higger we need decised use only one console channel chan_alsa , chan_oss or chan_console (which use portaudio).

Informatin this: " Load one of: chan_oss, alsa, or console (portaudio)." is from sources v13.0.8

sp2ong commented 7 years ago

Adi this information about load only one channel console type you can find:

https://github.com/asterisk/asterisk/blob/master/configs/samples/modules.conf.sample

dl1hrc commented 7 years ago

Ok, thanks, will try it here.

dl1hrc commented 7 years ago

ok, with noload => chan_console.so it's working as well :) I think a good way would be to register at https://community.asterisk.org/ to ask for help since I have very low experiences with asterisk.

sp2ong commented 7 years ago

I have put on extensions.conf in default following

[default] exten => _1234,1,Dial(SIP/74712063624110,30) exten => _1234,n,Hangup()

When I run in CLI asterisk command:

CLI> console dial 1234

asterisk connect to my SIP hamnet phone and I can under linux play DTMF code with script

dtmf 12234

and I hear this DTMF code in my SIP phone

but when I have disconect sip phone and agin play DTMC code under linux it is look like asterisk don't decode DTMF sounds in my asterisk. Maybe I need upgrade my asterisk to version from 13 to 14

sp2ong commented 7 years ago

Adi

I have asked on Asterisk community and I have the answer:

"The alsa channel has no built in support for looking for DTMF in the audio."

others channesl like chan_oss and chan_consloe have the same problem and:

"... those modules expect to receive an audio stream that would contain DTMF in them, so none will do that."

It is not good news. Allstar code have own version alsa channel "chan_alsaradio" but is is code for old version asterisk It is look at current we don't have way to send DTMF code from svxlink to asterisk with decode dtmf tones via asterisk channel