Skip to content
New issue

Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.

Sign up for GitHub

By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.

Already on GitHub? Sign in to your account

Jump to bottom

SVXLink and Asterisk ??? #250

Closed
sp2ong opened this issue Feb 22, 2017 · 56 comments
Closed

SVXLink and Asterisk ??? #250

sp2ong opened this issue Feb 22, 2017 · 56 comments

Comments

@sp2ong
Copy link

sp2ong commented Feb 22, 2017

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
Copy link
Contributor

dl1hrc commented Feb 22, 2017 via email

@sp2ong
Copy link
Author

sp2ong commented Feb 22, 2017

Ok good news :-)

73 Waldek SP2ONG

@dl1hrc
Copy link
Contributor

dl1hrc commented Feb 23, 2017 via email

@sp2ong
Copy link
Author

sp2ong commented Feb 23, 2017

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
Copy link
Contributor

dl1hrc commented Feb 23, 2017

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
Copy link
Author

sp2ong commented Feb 23, 2017

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

73 Waldek SP2ONG

@sp2ong
Copy link
Author

sp2ong commented Feb 24, 2017

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
Copy link
Owner

sm0svx commented Feb 24, 2017

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
Copy link
Contributor

dl1hrc commented Feb 25, 2017

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
Copy link
Author

sp2ong commented Feb 25, 2017

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
Copy link
Author

sp2ong commented Feb 25, 2017

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
Copy link
Author

sp2ong commented Feb 26, 2017 •
edited
Loading

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
Copy link
Contributor

dl1hrc commented Feb 26, 2017 via email

@sp2ong
Copy link
Author

sp2ong commented Feb 26, 2017

Hi Adi,

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

73 Waldek SP2ONG

@dl1hrc
Copy link
Contributor

dl1hrc commented Feb 26, 2017 via email

@sp2ong
Copy link
Author

sp2ong commented Feb 26, 2017

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
Copy link
Contributor

dl1hrc commented Feb 26, 2017 via email

@sp2ong
Copy link
Author

sp2ong commented Feb 26, 2017

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
Copy link
Contributor

dl1hrc commented Feb 26, 2017

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
Copy link
Author

sp2ong commented Feb 26, 2017

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
Copy link
Contributor

dl1hrc commented Feb 26, 2017

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
Copy link
Author

sp2ong commented Feb 26, 2017

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
Copy link
Contributor

dl1hrc commented Feb 26, 2017 via email

@sp2ong
Copy link
Author

sp2ong commented Feb 28, 2017

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
Copy link
Contributor

dl1hrc commented Feb 28, 2017 via email

@sp2ong
Copy link
Author

sp2ong commented Feb 28, 2017

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
Copy link
Contributor

dl1hrc commented Feb 28, 2017 via email

@VR2XKP
Copy link

VR2XKP commented Jun 2, 2017

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
Copy link
Author

sp2ong commented Jun 2, 2017 via email

@sm0svx
Copy link
Owner

sm0svx commented Jun 19, 2017

@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
Copy link
Contributor

dl1hrc commented Jun 20, 2017

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
Copy link
Author

sp2ong commented Jun 20, 2017

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
Copy link
Contributor

dl1hrc commented Jun 20, 2017

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

@sp2ong
Copy link
Author

sp2ong commented Jun 21, 2017

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
Copy link
Contributor

dl1hrc commented Jun 22, 2017

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

@sp2ong
Copy link
Author

sp2ong commented Jun 22, 2017

I have use OPUS but I will try RAW

@sp2ong
Copy link
Author

sp2ong commented Jun 23, 2017

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
Copy link
Author

sp2ong commented Jun 23, 2017

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
Copy link
Contributor

dl1hrc commented Jun 23, 2017

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

@sp2ong
Copy link
Author

sp2ong commented Jun 23, 2017

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
Copy link
Author

sp2ong commented Jun 23, 2017

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
Copy link
Author

sp2ong commented Jun 23, 2017

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
Copy link
Author

sp2ong commented Jun 23, 2017 •
edited
Loading

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
Copy link
Contributor

dl1hrc commented Jun 23, 2017

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
Copy link
Author

sp2ong commented Jun 23, 2017

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
Copy link
Author

sp2ong commented Jun 23, 2017

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
Copy link
Contributor

dl1hrc commented Jun 23, 2017

Ok, thanks, will try it here.

@dl1hrc
Copy link
Contributor

dl1hrc commented Jun 23, 2017

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
Copy link
Author

sp2ong commented Jun 23, 2017

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
Copy link
Author

sp2ong commented Jun 23, 2017

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

@sp2ong
Copy link
Author

sp2ong commented Jun 23, 2017

I have information about chan_alsaradio

https://yo3iiu.ro/blog/?p=361

and sources

http://www.yo3iiu.ro/archives/chan_alsaradio/chan_alsaradio.c_02.tar.gz

To correctly build the chan_alsaradio driver, the following line needs to be added in channels/Makefile:

chan_alsaradio.so: LIBS+=-lasound

But I don know that it is possible to add to current version asterisk

@dl1hrc
Copy link
Contributor

dl1hrc commented Jun 23, 2017

Hm, but maybe there is a chance to send a dtmf dial string to asterisk over another channel. There is no problem on SvxLink to put the received dmtf digits out (pty or tcl) and pipe them to a script that may initiate a dial command on asterisk. Could you find out that? Sri, I have less time this weekend :/

@sp2ong
Copy link
Author

sp2ong commented Jun 24, 2017 •
edited
Loading

I have tried complied chan_alsaradio
http://www.yo3iiu.ro/archives/chan_alsaradio/chan_alsaradio.c_02.tar.gz
with asterisk v13.8 but without success and it looks that this code is needed the rewrite to newest asterisk code because version 0.2 was created to asterisk 1.x
I think that chan_alsaradio source (of course after adoption to new asterisk code) is at current one way to have a link between svxlink and asterisk but we must find a developer who will do it..

@sp2ong
Copy link
Author

sp2ong commented Jun 29, 2017

I think that it maybe better solution will add module to svxlink IAXAsterisk to connect with asterisk via network TCP/IP

https://www.voip-info.org/wiki/view/Asterisk+IAX+channels
https://www.voip-info.org/wiki/view/IAX

this method will be independent to actually sources asterisk version
We can base on IAXClinet sources: https://sourceforge.net/projects/iaxclient/
and information https://www.codeproject.com/Articles/15277/Using-the-Asterisk-IAXClient-library-in-C

But maybe someone else has a better idea how to integrate svxlink with voip asterisk HamNet network

@ghost
Copy link

ghost commented Sep 15, 2017 •
edited by ghost
Loading

Hi, ive got some questions:
Is the "asterisk module" included in the current masterbranch?
How do i hang up the call on the radio side?
How can i dial on the radio side?

What will happen when i lost the connection to the SIP proxy?
I would like to built two testing gateways, thats why im asking.

Thanks in advance,
gamecompiler

@sp2ong
Copy link
Author

sp2ong commented Apr 8, 2019

#437

@sp2ong sp2ong closed this as completed Apr 8, 2019
Sign up for free to join this conversation on GitHub. Already have an account? Sign in to comment
Assignees
No one assigned
Labels
None yet
Projects
None yet
Milestone
No milestone
Development

No branches or pull requests
4 participants
@sm0svx @sp2ong @dl1hrc @VR2XKP