[asterisk-users] sip register peer (the quest for near 100% availability)
Hello. I am aware that 'sip show peers' will display my peers, and that 'sip unregister ' (where is the peer name) will unregister a peer - however, I want to force registration of a peer from the CLI. Is there any way to force this? I have several user agents and I want to achieve near 100% availability for all peers. I realise that the peer will be 'woken' up at my qualify intervals, but can I actually force registration from the CLI? -- _ -- 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] sip register peer (the quest for near 100% availability)
I am aware that the direction is from peer to asterisk. Its a valid question. If a solution did exist, guarantees near 100 per cent availability. Especially if the device is actually there. -- _ -- 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
[asterisk-users] Asterisk Messaging Refuses To Work!
I am pulling my hairs out here. This is my dialplan.
exten = 100,1,Set(AGISIGHUP=no)
exten = 100,n,AGI(a2billing.php,4,callingcard)
exten = 100,n,Set(__APP_MSG_IND=${APP_MSG_IND})
exten = 100,n,Set(__APP_MESSAGE=${APP_MESSAGE})
exten = 100,n,Hangup()
exten = h,1,GotoIf($[${APP_MSG_IND} = YES]?send-msg,1)
exten = h,n,Hangup()
exten = send-msg,1,SendText(${APP_MESSAGE})
exten = send-msg,n,Hangup()
I can see on the command line that the SendText() is actually being called, but
the softphone isnt getting the text.
What am I doing wrong?
Is there a variable to be set?
Any ideas will be most welcome
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Re: [asterisk-users] sip register peer (the quest for near 100% availability)
Thanks - I was hoping there was some silver bullet to use out there. Thanks anyway. -- _ -- 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 Messaging Refuses To Work!
I have changed the dial command to $tech/$dest,,gM(appmsg) and I have added [macro-appmsg] exten = s,1,SendText(Hello world) still no joy! :-( -- _ -- 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 Messaging Refuses To Work!
Hi there, it actually DID work - but the message was flashed on the receiving party's softphone -- _ -- 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] Limit registration concurrency per friend
Thanks - this is what I needed to know -- _ -- 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] Detect Low Quality Calls - Realtime
Thanks What would you use to measure jitter / packetloss in real time? -- _ -- 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] Detect Low Quality Calls - Realtime
Joachim, thanks for the reply - delay you can somewhat estimate prior to the call (with qualify for example) Pls be explicit. How do I use qualify to measure delay - The jitter / packetloss you can only figure out when the call is already up for a while. what would you use to measure jitter / packetloss in real time? -- _ -- 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
[asterisk-users] Limit registration concurrency per friend
Can I restrictthe number of concurrent registrations per friend? -- _ -- 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
[asterisk-users] MaxCallBR Peer Setting
Hi sip show peer 21342 gives me peer 21342's parameters. I am interested in the MaxCallBR line i.e. MaxCallBR: 384 kbps What exactly does this mean? -- _ -- 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] MaxCallBR Peer Setting
Its so obvious now that you've made it clear Thanks -- _ -- 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
[asterisk-users] Detect Low Quality Calls - Realtime
Hi there, I support a large number of enterprise users who contractually must connect to our support center via a 4G VOIP connection. I simply want to be able to auto detect all poor quality calls in realtme (as they are being made), play a message and drop the call - without user intervention. All decent call quality calls will be allowed through - to be handled by support staff. Its a challenging and tricky one as I cannot install any software on the callers endpoint. I can only detect calls as they hit our server, do the magic and based on latency, bandwidth and MOS (Meaning Opinion Score) - decide whether the call should be let through. I will accept all MOS values of 4.0 Any bright ideas? -- _ -- 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
