Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?
Ok. Thank you very much! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users em nome de Ben Newlin Enviado: terça-feira, 27 de março de 2018 17:10 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? Yes, when the ACK is lost there will be retransmissions of the 200 OK. But if the ACK is being misrouted or the connectivity issue persists for too long then the ACK will never be received. Now the endpoint that did not receive the ACK *should* then send a BYE to disconnect. However, not all endpoints operate as they should at all times and we have seen this sometimes does not occur. Also, if the network connectivity issue affected both sides of the call, then the BYE will not be received either. So you are right that the problem scenario requires both the ACK and BYE to be lost/misrouted/not sent. But as I said, it doesn’t happen often and even if it does many times the “stuck” calls cause no issues. But if billing or some other reporting/analytics are being done, the stuck calls can negatively affect those results. The INVITE refresh mechanism is part of the Dialog module and can be enabled when the dialog is created [1]. [1] http://www.opensips.org/html/docs/modules/2.3.x/dialog.html#idp5828384 dialog Module - opensips.org<http://www.opensips.org/html/docs/modules/2.3.x/dialog.html#idp5828384> www.opensips.org The dialog module provides dialog awareness to the OpenSIPS proxy. Its functionality is to keep trace of the current dialogs, to offer information about them (like ... Thanks, Ben Newlin From: Users on behalf of Rodrigo Pimenta Carvalho Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 1:55 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? Hi Ben. Thank you very much! I didn't realized such problems, until you explain that. I will check if my project will need the same procedure. In that case, I will study about INVITE refreshes. What I have observed in my OpenSIPS is that when a ACK is lost for a SIP OK, the callee sends SIP OK again and again. Could you point the OpenSIPS web page (from OpenSIPS documentation) that explain about INVITE refresh, please? Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users em nome de Ben Newlin Enviado: terça-feira, 27 de março de 2018 14:15 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? You don’t have to read very far back in the mailing list archives to see that misrouted ACKs are a fairly common problem when implementing SIP proxies. ☺ Mishandling of the Record-Route headers is the common problem, but loss of connectivity with the far end server can occur as well. Because the INVITE transaction is completed, the TM timers will not catch this and the dialog will stay in the CONFIRMED but not ACKed state until the $DLG_timeout expires. It doesn’t happen very often at all, but if it does and the timeout is set very high then you end up with a stuck call until the timer pops. If you are doing billing on the same endpoint then you potentially end up with a very long call being billed. There are also other ways to accomplish similar safeguards as this, including OPTIONS or INVITE refreshes using the Dialog module. We are still running 1.11 in production so the INVITE refreshes were not available to us and some of our partners do not accept OPTIONS refreshes. We plan to implement the INVITE refreshes once we have completed the upgrade to 2.X. Thanks, Ben Newlin From: Users on behalf of Rodrigo Pimenta Carvalho Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 12:57 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? 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 em nome de Ben Newlin 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_timeou
Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?
Yes, when the ACK is lost there will be retransmissions of the 200 OK. But if the ACK is being misrouted or the connectivity issue persists for too long then the ACK will never be received. Now the endpoint that did not receive the ACK *should* then send a BYE to disconnect. However, not all endpoints operate as they should at all times and we have seen this sometimes does not occur. Also, if the network connectivity issue affected both sides of the call, then the BYE will not be received either. So you are right that the problem scenario requires both the ACK and BYE to be lost/misrouted/not sent. But as I said, it doesn’t happen often and even if it does many times the “stuck” calls cause no issues. But if billing or some other reporting/analytics are being done, the stuck calls can negatively affect those results. The INVITE refresh mechanism is part of the Dialog module and can be enabled when the dialog is created [1]. [1] http://www.opensips.org/html/docs/modules/2.3.x/dialog.html#idp5828384 Thanks, Ben Newlin From: Users on behalf of Rodrigo Pimenta Carvalho Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 1:55 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? Hi Ben. Thank you very much! I didn't realized such problems, until you explain that. I will check if my project will need the same procedure. In that case, I will study about INVITE refreshes. What I have observed in my OpenSIPS is that when a ACK is lost for a SIP OK, the callee sends SIP OK again and again. Could you point the OpenSIPS web page (from OpenSIPS documentation) that explain about INVITE refresh, please? Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users em nome de Ben Newlin Enviado: terça-feira, 27 de março de 2018 14:15 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? You don’t have to read very far back in the mailing list archives to see that misrouted ACKs are a fairly common problem when implementing SIP proxies. ☺ Mishandling of the Record-Route headers is the common problem, but loss of connectivity with the far end server can occur as well. Because the INVITE transaction is completed, the TM timers will not catch this and the dialog will stay in the CONFIRMED but not ACKed state until the $DLG_timeout expires. It doesn’t happen very often at all, but if it does and the timeout is set very high then you end up with a stuck call until the timer pops. If you are doing billing on the same endpoint then you potentially end up with a very long call being billed. There are also other ways to accomplish similar safeguards as this, including OPTIONS or INVITE refreshes using the Dialog module. We are still running 1.11 in production so the INVITE refreshes were not available to us and some of our partners do not accept OPTIONS refreshes. We plan to implement the INVITE refreshes once we have completed the upgrade to 2.X. Thanks, Ben Newlin From: Users on behalf of Rodrigo Pimenta Carvalho Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 12:57 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? 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 em nome de Ben Newlin 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 on behalf of Rodrigo Pimenta Carvalho Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 10:08 AM To: OpenSIPS users mailling list 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
Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?
Hi Ben. Thank you very much! I didn't realized such problems, until you explain that. I will check if my project will need the same procedure. In that case, I will study about INVITE refreshes. What I have observed in my OpenSIPS is that when a ACK is lost for a SIP OK, the callee sends SIP OK again and again. Could you point the OpenSIPS web page (from OpenSIPS documentation) that explain about INVITE refresh, please? Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users em nome de Ben Newlin Enviado: terça-feira, 27 de março de 2018 14:15 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? You don’t have to read very far back in the mailing list archives to see that misrouted ACKs are a fairly common problem when implementing SIP proxies. ☺ Mishandling of the Record-Route headers is the common problem, but loss of connectivity with the far end server can occur as well. Because the INVITE transaction is completed, the TM timers will not catch this and the dialog will stay in the CONFIRMED but not ACKed state until the $DLG_timeout expires. It doesn’t happen very often at all, but if it does and the timeout is set very high then you end up with a stuck call until the timer pops. If you are doing billing on the same endpoint then you potentially end up with a very long call being billed. There are also other ways to accomplish similar safeguards as this, including OPTIONS or INVITE refreshes using the Dialog module. We are still running 1.11 in production so the INVITE refreshes were not available to us and some of our partners do not accept OPTIONS refreshes. We plan to implement the INVITE refreshes once we have completed the upgrade to 2.X. Thanks, Ben Newlin From: Users on behalf of Rodrigo Pimenta Carvalho Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 12:57 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? 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 em nome de Ben Newlin 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 on behalf of Rodrigo Pimenta Carvalho Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 10:08 AM To: OpenSIPS users mailling list 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 em nome de Ben Newlin 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 on behalf of Laszlo Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 9:40 AM To: OpenSIPS users mailling list 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 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
Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?
You don’t have to read very far back in the mailing list archives to see that misrouted ACKs are a fairly common problem when implementing SIP proxies. ☺ Mishandling of the Record-Route headers is the common problem, but loss of connectivity with the far end server can occur as well. Because the INVITE transaction is completed, the TM timers will not catch this and the dialog will stay in the CONFIRMED but not ACKed state until the $DLG_timeout expires. It doesn’t happen very often at all, but if it does and the timeout is set very high then you end up with a stuck call until the timer pops. If you are doing billing on the same endpoint then you potentially end up with a very long call being billed. There are also other ways to accomplish similar safeguards as this, including OPTIONS or INVITE refreshes using the Dialog module. We are still running 1.11 in production so the INVITE refreshes were not available to us and some of our partners do not accept OPTIONS refreshes. We plan to implement the INVITE refreshes once we have completed the upgrade to 2.X. Thanks, Ben Newlin From: Users on behalf of Rodrigo Pimenta Carvalho Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 12:57 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered? 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 em nome de Ben Newlin 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 on behalf of Rodrigo Pimenta Carvalho Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 10:08 AM To: OpenSIPS users mailling list 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 em nome de Ben Newlin 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<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 on behalf of Laszlo Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 9:40 AM To: OpenSIPS users mailling list 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 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<http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#timeout-pvar-id> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?
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 em nome de Ben Newlin 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 on behalf of Rodrigo Pimenta Carvalho Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 10:08 AM To: OpenSIPS users mailling list 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 em nome de Ben Newlin 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 on behalf of Laszlo Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 9:40 AM To: OpenSIPS users mailling list 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 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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 on behalf of Rodrigo Pimenta Carvalho Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 10:08 AM To: OpenSIPS users mailling list 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 em nome de Ben Newlin 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<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 on behalf of Laszlo Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 9:40 AM To: OpenSIPS users mailling list 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 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<http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#timeout-pvar-id> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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 em nome de Ben Newlin 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 on behalf of Laszlo Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 9:40 AM To: OpenSIPS users mailling list 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 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?
Hi Laszlo. Thank you for the reply. I'm using $DLG_timeout to configure how long a call will be, after answered. If I change $DLG_timeout, the duration of an answered call will change too. I have to avoid changing this way. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: Users em nome de Laszlo Enviado: terça-feira, 27 de março de 2018 10:39 Para: OpenSIPS users mailling list Assunto: 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 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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 Thanks, Ben Newlin From: Users on behalf of Laszlo Reply-To: OpenSIPS users mailling list Date: Tuesday, March 27, 2018 at 9:40 AM To: OpenSIPS users mailling list 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 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<http://www.opensips.org/html/docs/modules/2.4.x/dialog.html#timeout-pvar-id> ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to terminate a call, via Opensips, before it being answered?
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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
