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Summary:ASTERISK-01308: Using one SIP account and login from several locations
Reporter:kibo (kibo)Labels:
Date Opened:2004-03-28 08:25:09.000-0600Date Closed:2011-06-07 14:04:40
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
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Description:I tried to login with one account from different locations. I saw everything works fine except the fact that when you try to send a call from that channel, you won't know which one is going to ring. Although, I checked the logs and see each two registeration has got unique random postfix. So normally I expect if you place the call including the random postfix you should get through with the right party. Basicly this might cause a source for confusion. I guess, a new registeration should somehow either override the previous one or we should have a way to tell how many registrations we have and how can we reach them. I guess we should have both. we can place an option that states the number of registration allowed and base on that we can develop the rest.

Cheers
Kiarash
Comments:By: kibo (kibo) 2004-03-28 09:02:08.000-0600

Reminder sent to kibo

I just wanted to add, apparently the registeration will override and the last one is the valid one. But as two phones send registration periodicly. They behave sporadic. While both phones will be able to dial extentions simultanously. The last one registered will receive the calls. I guess two things are needed. First only the registered phone should be able to dial calls. In the case the phone is able to make conference calling and have numerious channels. There should be a way that someone can restrict the number of incoming and outgoing. The usually scenario is that there is only one channel which can be used either as incoming or outgoing. So if one call is in progress, the second call should get busy message.

By: Olle Johansson (oej) 2004-03-28 19:22:44.000-0600

SIP is documented to support only ONE registration per account. If you want multiple phones, each phone needs one SIP account in sip.conf.

By: Olle Johansson (oej) 2004-03-28 19:23:15.000-0600

--No bug, maybe documentation problem --