Hi Prem,
Yes, your scenario is possible with OpenSIPS B2BUA and it will work also
with TCP/TLS.
When asterisk gets the destination it can send a BYE to OpenSIPS with a
header ( with any name you like) containing the destination. You just
have to take care to put this name in the scenario ( modify this
http://www.opensips.org/Resources/B2buaTutorial#toc15).
Regards,
--
Anca Vamanu
www.voice-system.ro
On 07/26/2010 02:38 PM, Premalatha Kuppan wrote:
Thanks Anca.
Iam going to try this now and let you the results.
BTW, i have one question:
Iam using Opensips 1..6.2 TLS enabled, can i make it to works as well
as B2BUA ? If so how ?
Actaully, my scenario is All users will get registered with opensips
and all calls (Incoming Invite) would be forwarded to Asterisk
(1.4.3.1, it has no TLS, TCP support) for IVR. After IVR, asterisk
will get the destination address.
In my case, Destination address is either TLS/TCP/UDP enabled.
I was searching little bit, and found Opensips can handle Refere-to
header. From asterisk can i do transfer to Opensips and Opensips
should terminate the call log with asterisk and connect source and
destination party.
Will this logic work and opensips has that capabilities? If so what
needs to be configured to make it as B2BUA with the existing setup.
Please advice.
Thanks,
Prem
On Mon, Jul 26, 2010 at 3:50 PM, Anca Vamanu <[email protected]
<mailto:[email protected]>> wrote:
Hi Prem,
There are some things that you have to take care if you want TCP
to work:
1) set the script parameter : tcp_accept_aliases = 1'
2) the top most Via header in Register from the client must
contain 'alias' parameter. To check this print the Register
message from the script: xlog("$mb\n"); If you don't see this
parameter, you can fix it from the server by calling
force_tcp_alias() on that Register.
3) the ip and port in the top most Via for Register must match
exactly the ip and port in the Contact header for tcp reusage to
work. Check this.
Regards,
--
Anca Vamanu
www.voice-system.ro <http://www.voice-system.ro/>
On 07/26/2010 08:54 AM, Premalatha Kuppan wrote:
Hi,
I have Integrated setup of Opensips(TLS) (1.6.2)) and Asterisk
(1.4.3.1). When i try to make call to TCP enabled client. The
call fails.
Below is the Error observed at Opensips.
Jul 23 13:54:45 204548-4 /usr/local/sbin/opensips[8374]
: new branch at sip:[email protected]:1036;transport=TCP
Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
ERROR:core:tcp_blocking_connect: timeout 10 s elapsed from 10 s
Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
ERROR:core:tcpconn_connect: tcp_blocking_connect failed
Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
ERROR:core:tcp_send: connect failed
Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
ERROR:tm:msg_send: tcp_send failed
Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]:
ERROR:tm:t_forward_nonack: sending request failed
Jul 23 13:54:55 204548-4 /usr/local/sbin/opensips[8374]: ACC:
call missed:
timestamp=1279907695;method=INVITE;from_tag=as2bf86f8d;to_tag=;[email protected]
<mailto:[email protected]>;code=477;reason=Request
Failure
Jul 23 13:54:57 204548-4 /usr/local/sbin/opensips[8374]: ACC:
transaction answered:
timestamp=1279907697;method=BYE;from_tag=as7d156ab4;to_tag=3488896563-937568;[email protected]
<mailto:[email protected]>;code=200;reason=OK
Any Insight ?
Thanks,
Prem
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