Hi.
Just as curiosity, what would cause an ACK lost in your system? Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: Users <users-boun...@lists.opensips.org> em nome de Ben Newlin <ben.new...@genesys.com> Enviado: terça-feira, 27 de março de 2018 11:18 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? Rodrigo, Yes, they do. I am using them to do exactly what you describe. The final reply (fr) timer is how long a transaction will wait to receive a final reply (>=200). If the timer expires without receiving a final reply the transaction will be canceled and failure route will be triggered with, I think, a local 408 response. As for $DLG_timeout, you can set that value multiple times in a call. We do this as well. Prior to the call being ACKed we set this value fairly low (~5s) in order to disconnect the dialog if the ACK is lost. Once we receive the ACK, we then extend it to a much longer value. Thanks, Ben Newlin From: Users <users-boun...@lists.opensips.org> on behalf of Rodrigo Pimenta Carvalho <pime...@inatel.br> Reply-To: OpenSIPS users mailling list <users@lists.opensips.org> Date: Tuesday, March 27, 2018 at 10:08 AM To: OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? Ok Ben. I will check this possibility and see if reply times will change the duration of a not answered call. Thank you. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ________________________________ De: Users <users-boun...@lists.opensips.org> em nome de Ben Newlin <ben.new...@genesys.com> Enviado: terça-feira, 27 de março de 2018 10:43 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? You can also use the reply timers in TM to do this: http://www.opensips.org/html/docs/modules/2.3.x/tm.html#fr_timeout tm Module - openSIPS<http://www.opensips.org/html/docs/modules/2.3.x/tm.html#fr_timeout> www.opensips.org TM module enables stateful processing of SIP transactions. The main use of stateful logic, which is costly in terms of memory and CPU, is some services inherently ... Thanks, Ben Newlin From: Users <users-boun...@lists.opensips.org> on behalf of Laszlo <las...@voipfreak.net> Reply-To: OpenSIPS users mailling list <users@lists.opensips.org> Date: Tuesday, March 27, 2018 at 9:40 AM To: OpenSIPS users mailling list <users@lists.opensips.org> Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? On Tue, Mar 27, 2018 at 3:31 PM, Rodrigo Pimenta Carvalho <pime...@inatel.br<mailto:pime...@inatel.br>> wrote: Hi. When a peer invites another one to a call, there are calling and ringing tones for these peers. My SIP agents let these tones execute during 2 minutes. After this, the call is terminated, if no one answers the call. How to configure OpenSIPS, if possible, so that any call will be terminated after 1 minute? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 Maybe you can play with $DLG_timeout, see http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#timeout-pvar-id
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