Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Asterisk as useragent registered using 2 ac


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
rizwanhasham at gmail.com
Guest





PostPosted: Tue Feb 26, 2008 8:50 am    Post subject: [asterisk-users] Asterisk as useragent registered using 2 ac Reply with quote

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080226/0df58f21/attachment.htm
Back to top
igi-go at ya.ru
Guest





PostPosted: Wed Feb 27, 2008 6:36 am    Post subject: [asterisk-users] Asterisk as useragent registered using 2 ac Reply with quote

Rizwan Hisham wrote:
Quote:
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
Back to top
rizwanhasham at gmail.com
Guest





PostPosted: Fri Feb 29, 2008 9:34 am    Post subject: [asterisk-users] Asterisk as useragent registered using 2 ac Reply with quote

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:

Quote:
Rizwan Hisham wrote:
Quote:
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
Quote:
dialplan in such a way that both local accounts on AST1 use different
trunks
Quote:
to send the call to AST2 server. These 2 different trunks are for 2
accounts
Quote:
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080229/253ef5d6/attachment.htm
Back to top
rizwanhasham at gmail.com
Guest





PostPosted: Wed Mar 05, 2008 3:55 am    Post subject: [asterisk-users] Asterisk as useragent registered using 2 ac Reply with quote

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:

Quote:
Rizwan Hisham wrote:
Quote:
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
Quote:
dialplan in such a way that both local accounts on AST1 use different
trunks
Quote:
to send the call to AST2 server. These 2 different trunks are for 2
accounts
Quote:
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080305/7d4ccf07/attachment.htm
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services