asterisk users - Feb 2008 - Asterisk as useragent registered using 2 accounts

Hi all,
I am having a strange problem. I am using my asterisk server AST1 to
register with another asterisk server AST2 using 2 accounts (2 register
commands in sip.conf). I have made 2 local users in AST1, and configured my
dialplan in such a way that both local accounts on AST1 use different trunks
to send the call to AST2 server. These 2 different trunks are for 2 accounts
i have registered on AST1.


line1 ---> trunk1(ON AST1) ---> AST2
line2 ---> trunk2(ON AST1) ---> AST2

These 2 trunks are to differentiate that the call is coming from one of the
2 registered accounts on AST1.

The problem is, my AST2 server cannot differentiate between 2 accounts. It
always dumps the cdr at the end of every call against only one of the 2
registered accounts (acc2 even if im dialing from acc1) on AST1 i.e. the
call always goes out using account-2 even if i dial from accout-1. Here is
my sip.conf

TRUNKS

[acc1]
username=acc1
type=friend
secret=123
qualify=yes
port=9060
nat=yes
insecure=port,invite
host=ip-of-my-AST2
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm

[acc2]
username=acc2
type=friend
secret=123
qualify=yes
port=9060
nat=yes
insecure=port,invite
host=ip-of-my-AST2
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm


REGSITRATION

register => acc1:123 at ip-of-my-AST2:9060
register => acc2:123 at ip-of-my-AST2:9060

local lines on AST1 use trunk acc1 and acc2 to throw calls to my AST2.

But it seems AST2 does not recognise that calls are coming from 2 different
accounts.

How can i make asterisk realize it?

-- 
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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Rizwan Hisham wrote:
> I am having a strange problem. I am using my asterisk server AST1 to
> register with another asterisk server AST2 using 2 accounts (2 register
> commands in sip.conf). I have made 2 local users in AST1, and configured my
> dialplan in such a way that both local accounts on AST1 use different
trunks
> to send the call to AST2 server. These 2 different trunks are for 2
accounts
> i have registered on AST1.
> (skiped)
>
> How can i make asterisk realize it?
>   
You must enable authentication of INVITE that AST1 send to AST2. Now you 
have no authentication of incoming INVITE and AST2 make decision about 
used account based only on IP address of caller peer.

Changing insecure=port,invite to insecure=port should help.

-- 
Best regards,
Igor A. Goncharovsky

Thanx for the tip. It has erased the problem i was having using
authentication but another problem has arised. i am now able to call with
only one user from AST1 to AST2. If i dial using the other user, my AST2
shows the following warning and responds with a "403 forbidden"
sip response:

*WARNING[13520]: chan_sip.c:8117 check_auth: username mismatch, have
<adf>,
digest has <abc>*

Any solutions to this problem?


On Wed, Feb 27, 2008 at 4:36 PM, Igor A. Goncharovsky <igi-go at ya.ru>
wrote:
> Rizwan Hisham wrote:
> > I am having a strange problem. I am using my asterisk server AST1 to
> > register with another asterisk server AST2 using 2 accounts (2
register
> > commands in sip.conf). I have made 2 local users in AST1, and
configured
> my
> > dialplan in such a way that both local accounts on AST1 use different
> trunks
> > to send the call to AST2 server. These 2 different trunks are for 2
> accounts
> > i have registered on AST1.
> > (skiped)
> >
> > How can i make asterisk realize it?
> >
> You must enable authentication of INVITE that AST1 send to AST2. Now you
> have no authentication of incoming INVITE and AST2 make decision about
> used account based only on IP address of caller peer.
>
> Changing insecure=port,invite to insecure=port should help.
>
> --
> Best regards,
> Igor A. Goncharovsky
>
>
> _______________________________________________
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> To UNSUBSCRIBE or update options visit:
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>


-- 
Best Regards
Rizwan Hisham
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Adding "fromuser" option in trunk declaration in AST1 has solved all
problems though.

On Wed, Feb 27, 2008 at 4:36 PM, Igor A. Goncharovsky <igi-go at ya.ru>
wrote:
> Rizwan Hisham wrote:
> > I am having a strange problem. I am using my asterisk server AST1 to
> > register with another asterisk server AST2 using 2 accounts (2
register
> > commands in sip.conf). I have made 2 local users in AST1, and
configured
> my
> > dialplan in such a way that both local accounts on AST1 use different
> trunks
> > to send the call to AST2 server. These 2 different trunks are for 2
> accounts
> > i have registered on AST1.
> > (skiped)
> >
> > How can i make asterisk realize it?
> >
> You must enable authentication of INVITE that AST1 send to AST2. Now you
> have no authentication of incoming INVITE and AST2 make decision about
> used account based only on IP address of caller peer.
>
> Changing insecure=port,invite to insecure=port should help.
>
> --
> Best regards,
> Igor A. Goncharovsky
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
Best Regards
Rizwan Hisham
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