Manolo Quijano
2013-Aug-27 00:11 UTC
[asterisk-users] Interconnect Radio units using asterisk
Hi all, This my first mail in the community. My name is Manolo. I'm new in Asterisk. My objective is to control some radio using Asterisk via web. In google I could see that the application app_rpt had this goal, but currently is not in the new Asterisk 1.8 or higher. Thinking in future, I thought the easer way to do that is to buy some RoIP unit and do the communication with asterisk via SIP or IAX2. Someone with experience in these kind of units, could give me some link or P/N of some equipmet that work fine with current version of Asterisk? Thanks in advance, Manolo -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130827/6c7398fb/attachment.htm>
James Sharp
2013-Aug-27 05:53 UTC
[asterisk-users] Interconnect Radio units using asterisk
On Aug 26, 2013, at 8:11 PM, Manolo Quijano <manolo.quijano at gmail.com> wrote:> Hi all, > > This my first mail in the community. My name is Manolo. I'm new in Asterisk. My objective is to control some radio using Asterisk via web. > > In google I could see that the application app_rpt had this goal, but currently is not in the new Asterisk 1.8 or higher. > > Thinking in future, I thought the easer way to do that is to buy some RoIP unit and do the communication with asterisk via SIP or IAX2. > > Someone with experience in these kind of units, could give me some link or P/N of some equipmet that work fine with current version of Asterisk? > > Thanks in advance, > Manolo >Hi Manolo, While not specifically "RoIP" boxes, I've used the Multitech MVP-200 and 400 series VOIP gateways to interface radio systems over SIP. The MVP series boxes come in a model that supports 4-wire E&M interfaces which are pretty common in radio systems. I programmed the units to auto dial a SIP address when the E&M interface was seized. They didn't specifically use Asterisk, but I don't see why you couldn't stick Asterisk somewhere in there since SIP is SIP. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130827/3533a052/attachment.htm>