Re: [SR-Users] Contents of sr-users Digest, Vol 96, Issue 68

Thanks for the quick reply. 1. Re: SCTP question (Daniel-Constantin Mierla) What I meant is if the SCTP association goes down from the Invite, 200 ok w/sdp to BYE. Should there be not an error at t_relay() level , since the association does not exists ? Currently I don't get any error when I do if (!t_relay()) { xlog("L_INFO", "T_Relay return code is $retcode\n"); sl_reply_error(); } I have set up $du with the IP:port of the host where I received the Invite and ACK to 200 ok w/sdp. I have shut down the other side , so I know that SCTP is not up. Thanks, --Jignesh -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request(a)lists.sip-router.org Sent: Monday, May 20, 2013 2:25 PM To: sr-users(a)lists.sip-router.org Subject: sr-users Digest, Vol 96, Issue 68 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: SCTP question (Daniel-Constantin Mierla) 2. Re: [SR-Users] http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour (Daniel-Constantin Mierla) 3. Re: Kamailio + Siremis Outbound route (Daniel-Constantin Mierla) 4. SCTP support of MultiHoming... (Jignesh Gandhi) 5. Re: SCTP support of MultiHoming... (Daniel-Constantin Mierla) ---------------------------------------------------------------------- Message: 1 Date: Mon, 20 May 2013 16:56:55 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: Jignesh Gandhi <jigpgandhi(a)gmail.com>om>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] SCTP question Message-ID: <519A39B7.3050503(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Hello, On 5/20/13 6:12 AM, Jignesh Gandhi wrote: By middle of the call, do you mean in between the INVITE and the BYE? I am not that familiar with SCTP, but probably a new one is created with the BYE. The same should be happening with tcp. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu *