Re: [asterisk-users] Asterisk 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls
Le 18/02/2013 18:54, Chris Bagnall a écrit : On 18/2/13 5:39 pm, Administrator TOOTAI wrote: on incoming call we have exten => 100,n,Dial(SIP/Handset_102&SIP/Handset_103&SIP/Handset_104,,) and always only Handset_102 is ringing, we receive "busy" back from the 2 others but they are not. Any clue? It depends which base station you're using - some of the earlier ones only supported one or two simultaneous SIP calls (remember dialling counts as a call, even if it's not answered). I seem to recall the N300IP (the one we use) supports 3 concurrent SIP calls. The easiest workaround is probably to create a fourth SIP account called '102_103_104' or something that's set to ring all 3 handsets on the Gigaset web interface. You can then Dial(SIP/Handset_102_103_104) from Asterisk instead. That make sens, good idea, thanks ;-) Before I read your message I already splitted the call by calling at first the 2 most important handsets and in case of no answer, the third one with one of the first ones. Will apply your proposal if my setup isn't convenient to the customer. Thanks for your help -- Daniel -- _ -- 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 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls
On 18/2/13 5:39 pm, Administrator TOOTAI wrote: on incoming call we have exten => 100,n,Dial(SIP/Handset_102&SIP/Handset_103&SIP/Handset_104,,) and always only Handset_102 is ringing, we receive "busy" back from the 2 others but they are not. Any clue? It depends which base station you're using - some of the earlier ones only supported one or two simultaneous SIP calls (remember dialling counts as a call, even if it's not answered). I seem to recall the N300IP (the one we use) supports 3 concurrent SIP calls. The easiest workaround is probably to create a fourth SIP account called '102_103_104' or something that's set to ring all 3 handsets on the Gigaset web interface. You can then Dial(SIP/Handset_102_103_104) from Asterisk instead. A cautionary note with Gigasets in general: they claim each base station will support up to 7 handsets. In my experience, things start to get a bit "flaky" above 3 or 4 handsets (specific handsets not ringing periodically, etc.), so I suspect the base station might be CPU limited at some point, especially if you're asking it to use an expensive (computationally) codec like G.729. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls
Le 17/02/2013 18:27, Chris Bagnall a écrit : On 17/2/13 5:02 pm, Administrator TOOTAI wrote: customer102/Handset_102 xxx.yyy.zzz.153 D N 5062 OK (80 ms) customer103/Handset_103 xxx.yyy.zzz.153 D N 5062 OK (70 ms) customer104/Handset_104 xxx.yyy.zzz.153 D N 5062 OK (66 ms) That's perfectly normal with these phones, and shouldn't pose a problem. As you see, all handsets are identified with the same port, which means that on incoming call to one handset or when transfering a call with the asterisk transfer feature, all 3 handsets are ringing :-( You can specify which SIP account correlates to each handset in the Gigaset web interface. Go to Settings -> Telephony -> Number Assignment You want Handset 1 to use Connection 'Handset_102' for outgoing calls and for incoming calls (untick everything else except this for incoming calls). Likewise Handset 2 should use Connection 'Handset_103' for outgoing and incoming (again, untick everything but this option). Chris, one problem stays, you perhaps already face it: on incoming call we have exten => 100,n,Dial(SIP/Handset_102&SIP/Handset_103&SIP/Handset_104,,) and always only Handset_102 is ringing, we receive "busy" back from the 2 others but they are not. Any clue? -- Daniel -- _ -- 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 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls
Le 17/02/2013 18:27, Chris Bagnall a écrit : [...] Go to Settings -> Telephony -> Number Assignment You want Handset 1 to use Connection 'Handset_102' for outgoing calls and for incoming calls (untick everything else except this for incoming calls). Likewise Handset 2 should use Connection 'Handset_103' for outgoing and incoming (again, untick everything but this option). [...] Hi Chris, that did the trick, many thanks. -- Daniel -- _ -- 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 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls
On 17/2/13 5:02 pm, Administrator TOOTAI wrote: customer102/Handset_102 xxx.yyy.zzz.153 D N 5062 OK (80 ms) customer103/Handset_103 xxx.yyy.zzz.153 D N 5062 OK (70 ms) customer104/Handset_104 xxx.yyy.zzz.153 D N 5062 OK (66 ms) That's perfectly normal with these phones, and shouldn't pose a problem. As you see, all handsets are identified with the same port, which means that on incoming call to one handset or when transfering a call with the asterisk transfer feature, all 3 handsets are ringing :-( You can specify which SIP account correlates to each handset in the Gigaset web interface. Go to Settings -> Telephony -> Number Assignment You want Handset 1 to use Connection 'Handset_102' for outgoing calls and for incoming calls (untick everything else except this for incoming calls). Likewise Handset 2 should use Connection 'Handset_103' for outgoing and incoming (again, untick everything but this option). Rinse and repeat for other handsets. I can confirm it does work properly - we have dozens of clients with Gigaset phones and separate SIP registrations per handset. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls
Hi everybody, We installed the Gigaset C610IP to one of our customer, those phone are natted and connects to our Asterisk 1.8.19. Each handset has is own account on our Asterisk, lets say Handset_102, Handset_103 and Handset_104. Problem is this one, taken from a sip show peers: customer102/Handset_102 xxx.yyy.zzz.153 D N 5062 OK (80 ms) customer103/Handset_103 xxx.yyy.zzz.153 D N 5062 OK (70 ms) customer104/Handset_104 xxx.yyy.zzz.153 D N 5062 OK (66 ms) As you see, all handsets are identified with the same port, which means that on incoming call to one handset or when transfering a call with the asterisk transfer feature, all 3 handsets are ringing :-( We tried using fixed port (sample above with port 5062) as well as random, no changes. We know that few of you are using those phones, how did you manage to solve this problem? Would be great if you could share. Regards -- Daniel -- _ -- 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
