[Asterisk-Users] Fwd: sip.conf and jump from register to the extension

Thomas Winter thowinter at googlemail.com
Thu Apr 20 01:29:10 MST 2006


Am Thursday 20 April 2006 01:21 schrieb tom:
> Thomas Winter wrote:
> > I have done additional tests, because the documentation sample was not
> > 100 % identical to my register command.
> >
> > OK:
> > register => 44198:password at provider.com/200
> > This jumps to 200, s is also working
> >
> > NOT OK:
> > user:password at provider.com/200
> > It looks for extension user and is ignoring 200 or anythink else
> >
> > I think the non numeric username is the problem.
> >
> > Yes, I have done an restart of Asteriks after changing the sip.conf.
>
> Excuse me for sounding silly, but isn't the extension you mark at the
> end sent to your provider as the extension that they should use when
> calling you (ie. in the authentication statement, the remote server
> tries to connect as 200@<yourhost-ip> ) which is why some providers with
> broken sip implementations require you to have a specific extension
> after the /.
>
> ie. the extension they call on, is not neccesarily what is stated in the
> register statement, that's just the extension you've told them to call
> you on.
>
> Do a sip debug provider.com in the asterisk CLI to see what happens when
> the call comes in.

I have used Ethereal.
The initial call comes in: SIP/SDP Request Invite sip:user at myIP
user is allways the username from the register , there is no information 
regards the user extension from the register command in. (or I didnt see 
them)

If * is register at the sip provider there are contact bindings send to the 
SIP-proxy with the extension from the register command and some messages with 
bindings are comming back. I dont know how this is related to the call 
INVITE.

Anyway, I have tested 4 different provider, two numeric and two alphanumeric 
username.
numeric is working and alphanumeric is not working.
If the provider software is broken it would be also good to know. 












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