[asterisk-dev] SIP Outbound Registrations

[Johan Wilfer]

Olle E Johansson wrote:
> 17 nov 2007 kl. 06.45 skrev Johan Wilfer:
>
>   
>> Hello,
>>
>> I've tried to understand how Asterisk handles outbound sip  
>> registrations. In production I am using 1.4 and when I add
>> more than one account that uses the same sip server, only one of the  
>> will function. On the other one I get a error-tone,
>> while calling. I read something about this behaviour and that it is  
>> a limitation of the Asterisk sip implementation.
>> This is 100% reproducible, Asterisk claims both numbers are  
>> registerd, I see the sip dialogues with the registration
>> data, but only one of the numbers work at the time. My pstn-provider  
>> use OpenSER.
>>
>> Yesterday I started to play with the current svn source for  
>> Asterisk, and sadly this is still the way Asterisk behaves.
>> For my production server I've solved the problem by using the  
>> excellent service at mysipswitch.com. But this isn't
>> the way I would like to have it...
>>
>> I have to ask, isn't this a very big limitation of Asterisk? Or am I  
>> doing it terribly wrong?
>>     
> Asterisk certainly handles more than one registration per server. This  
> is not, however, a development
> question, so please search voip-info.org or use the asterisk-users  
> mailing list. 
>
> Best regards,
>
> /Olle
>   
Ok, sorry. I've asked these questions previously at the digium forum and 
then got the response that
Asterisk didn't handle multiple registrations at the same sip-server 
very well.

After trying with the latest SVN version and still having the same 
problem I figured I ask you about this.
But I guess I should be happy about being wrong, because that means it 
could be fixed easier.

The strange thing is that my registrations works just fine - one at a 
time. But when I have both of them
active only on of the registered accounts are working. And Asterisk 
thinks that both are working...
This confuses me..

Greetings anyway
/Johan
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>