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When setting type=friend for the incoming calls :<BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">; outgoing conversations</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[user1-out]</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">type=peer</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">host=sip.ITSP</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">username=user1</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">secret=secret1</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">fromuser=user1</FONT></FONT><BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">; incoming conversations</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[user1]</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">type=friend</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">host=sip.ITSP</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">context=user1incoming</FONT></FONT><BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">; outgoing conversations</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[user2-out]</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">type=peer</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">host=sip.ITSP</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">username=user2</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">secret=secret2</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">fromuser=user2</FONT></FONT><BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">; incoming conversations</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[user2]</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">type=friend</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">host=sip.ITSP</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">context=user2incoming</FONT></FONT><BR>
<BR>
I get the following message :<BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[Jan 29 18:49:07] NOTICE[6314]: chan_sip.c:14703 handle_request_invite: Call from 'user2' to extension "329990102" rejected because extension not found.</FONT></FONT><BR>
<BR>
The call in fact needs to come from user1 in stead of from user2. Of course the extension is not found as it is defined in the context for incoming calls of user 1.<BR>
<BR>
In sip.conf [user2] is defined after [user1] and I have the impression that the last definition of a user is always taken. [user2] is always taken for incoming calls.<BR>
<BR>
<BR>
How can I identify calls that are destined for user1 as defined in :<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">register => user1:passwd1@sip.ITSP</FONT></FONT><BR>
<BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">[user1]</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">type=friend</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">host=sip.ITSP</FONT></FONT><BR>
<FONT SIZE="2"><FONT COLOR="#0000ff">context=user1incoming</FONT></FONT><BR>
<BR>
<BR>
It must be possible to have several accounts with the same ITSP on the same Asterisk-server ?!<BR>
<BR>
<BR>
Jonas.<BR>
<BR>
<BR>
On Fri, 2010-01-29 at 16:51 +0000, Robert Lister wrote:
<BLOCKQUOTE TYPE=CITE>
<PRE>
On Fri, 2010-01-29 at 15:09 +0100, jonas kellens wrote:
> Hello list !
>
> Having troubles with multiple registrations to one and the same ITSP.
>
> This sip.conf :
>
> register => user1:<A HREF="mailto:passwd1@sip.ITSP">passwd1@sip.ITSP</A>
> register => user2:<A HREF="mailto:passwd2@sip.ITSP">passwd2@sip.ITSP</A>
>
> ; outgoing conversations
> [user1-out]
> type=peer
> host=sip.ITSP
Try setting type=friend instead of peer for these and see what happens.
</PRE>
</BLOCKQUOTE>
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