Closed sp2ong closed 5 years ago
I have information about chan_alsaradio
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
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 :/
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..
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
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
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 ???