Re: [asterisk-users] Asterisk as a distributed paging system
On Wed, 22 Sep 2010, Matteo Fortini wrote: > I'm building a paging system composed of roughly 10 switches in daisy > chain, with an embedded box with a speaker and a microphone for each > switch. The embedded box runs my software. > > I need the system to be resilient to any network partition, so that > anyone can send announces from any mic to all the reachable clients. I'd > need also to page a subset of all the speakers. > > I'm currently using some software I wrote which sends voice over > multicast RTP and coordinates all the sites with multicast messages. > > I don't own the switches so each site will be assigned an address by > DHCP, that's why I'm using multicast. > > I heard of asterisk and SIP as a possible alternative to my software, > and I'd rather use tested and widely adopted software. > > Is there a way asterisk could be of use, or would I need to bend it too > much? It does this as standard. However, to make it work you need to write some Asterisk dialplan code, and have SIP devices (phones) that can auto-answer and go into speakerphone mode. It also doesn't use multicast, so the number of 'speakers' (phones) you have might be a limitation. I've tested it with 20 without any issues, however as it uses the internal conferernce facilities (meetme), I know that there are people out there using asterisk to host some very large conferernces, so I suspect for your implementation it won't be an issue... Your requirement of any to all reachable, even if the network is broken is tricky - you might end up having an asterisk box at each switch (which I assume is an Ethernet switch) although then you'll have problems with the SIP device registrations. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a distributed paging system
With a proper setup and asynchronous dialing, this can be done in a relatively seamless (although not as simple as this indicates) fashion. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a distributed paging system
Hi! > I need the system to be resilient to any network partition, so that > anyone can send announces from any mic to all the reachable clients. > I'd need also to page a subset of all the speakers. Most of the major phone vendors (that are employed by the users of this list) have support for multi-cast of some sort. In recent firmware release notes I have read that SNOM has now also added a feature to feed multicast directly from a phone (and not just play multicast audio on the speaker as long as the phone is not in use). > I'm currently using some software I wrote which sends voice over > multicast RTP and coordinates all the sites with multicast messages. app_page has been around for quite some in Asterisk, and the new Asterisk 1.8 now also adds the channel driver "MulticastRTP". > Is there a way asterisk could be of use, or would I need to bend it > too much? Look here: http://www.voip-info.org/wiki/view/Asterisk+cmd+Page http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom I have made good experience with MAST for multicasting SNOM phones: http://www.aelius.com/njh/mast/ Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a distributed paging system
Hi! thank you for your good answers. Another related question: I tried using Page() and it works perfectly, but I need to implement a slightly different behavior, and I'm looking into ways of implementing it. When a user picks up the phone and chooses to page the speakers, the call should start (so that it's ready for talking), but in muted status. When the user pushes a push-to-talk button, then a bell sound needs to be played through all the speakers, then she can start talking freely. Everytime the PTT button is released, the mic needs to mute, but that's something I can work out in the softphone. How can I implement it? I am thinking of using some call parking method and some DTMF code to pass to the next state, but I am open to advice, since I'm quite new to Asterisk. Could I also create a macro to do the same thing Page is doing, but with ConfBridge? Last question: is there a way of reinviting periodically remotes to the conference, so that they can recover after e.g. a reboot? Thank you in advance, Matteo Il 22/09/2010 21:51, Philipp von Klitzing ha scritto: > Hi! > > >> I need the system to be resilient to any network partition, so that >> anyone can send announces from any mic to all the reachable clients. >> I'd need also to page a subset of all the speakers. >> > Most of the major phone vendors (that are employed by the users of this > list) have support for multi-cast of some sort. In recent firmware > release notes I have read that SNOM has now also added a feature to feed > multicast directly from a phone (and not just play multicast audio on the > speaker as long as the phone is not in use). > > >> I'm currently using some software I wrote which sends voice over >> multicast RTP and coordinates all the sites with multicast messages. >> > app_page has been around for quite some in Asterisk, and the new Asterisk > 1.8 now also adds the channel driver "MulticastRTP". > > >> Is there a way asterisk could be of use, or would I need to bend it >> too much? >> > Look here: > http://www.voip-info.org/wiki/view/Asterisk+cmd+Page > http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom > > I have made good experience with MAST for multicasting SNOM phones: > http://www.aelius.com/njh/mast/ > > Philipp > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a distributed paging system
On 09/22/2010 05:49 PM, Matteo Fortini wrote: > I'm building a paging system composed of roughly 10 switches in daisy > chain, with an embedded box with a speaker and a microphone for each > switch. The embedded box runs my software. I'm not sure if this is much use to your setup - but I've implemented a public announcement system (in UK sometimes called tannoy) using a pc and a pair of loud speakers. I used linphonec command line sip client - with auto-answer set. This has the disadvantage, of course, that it can't be used as a phone as well, to make phone calls from the machine driving the speakers (unless you install another sip client) - just as a speaker. You might be able to run linphonec on an embedded platform as well - with a speaker attached. I know there were some compiled for the NSLU2/arm platform already. You will have to combine it with Page() to get multiple nodes working at the same time. I only used one. I did a write-up here - see if you find any useful information: http://forum.voxilla.com/asterisk-support-forum/asterisk-public-announcement-system-loud-ringer-bell-49339.html Sebastian > > I need the system to be resilient to any network partition, so that > anyone can send announces from any mic to all the reachable clients. I'd > need also to page a subset of all the speakers. > > I'm currently using some software I wrote which sends voice over > multicast RTP and coordinates all the sites with multicast messages. > > I don't own the switches so each site will be assigned an address by > DHCP, that's why I'm using multicast. > > I heard of asterisk and SIP as a possible alternative to my software, > and I'd rather use tested and widely adopted software. > > Is there a way asterisk could be of use, or would I need to bend it too > much? > > Thank you in advance, > Matteo > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
