Thanks Anca. This is what i was looking for. Btw, if i understood correct. I can make current Opensips (TLS enabled - 1.6.2 (Proxy)) one to act as B2BUA. or is it i hv to uninstall the current one and install other 2BUA. Iam confused. Please clarify.
On Mon, Jul 26, 2010 at 6:15 PM, Anca Vamanu <[email protected]> wrote: > 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 Vamanuwww.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]> 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 Vamanuwww.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=;call_id= >> [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;call_id= >> [email protected];code=200;reason=OK >> >> Any Insight ? >> >> Thanks, >> Prem >> >> > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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