Re: [asterisk-users] SIP call control via RTCP
Hello, I have found here http://www.voip-info.org/wiki/view/Asterisk+RTCP that there has been a patch for RTCP of Asterisk 1.4. Does this mean that starting with Asterisk version 1.6 RTCP call control is working correctly? Kind regards Jan Gaida On Mon, May 12, 2014 at 2:40 PM, Jan Gaida jan.ga...@grupoamper.com wrote: Thank you. Yes, that should work. But if I understand it correctly, only if there's no silence detection activated. Otherwise, when silence is detected no RTP would be send, so that rtptimeout would hang up a still active call. I there no option to use RTCP? Not even in Asterisk 11? Regards On Mon, May 12, 2014 at 2:12 PM, Matt Behrens m...@zigg.com wrote: On May 12, 2014, at 5:02 AM, Jan Gaida jan.ga...@grupoamper.com wrote: We are using Asterisk 1.4 as call distribution system with simple queues for SIP calls. With high load (4000 calls/hour) some calls remain in queue forever (until queue's max wait time) in spite of being hung up already by the caller. It seems that when a BYE is lost, Asterisk has no mechanism to check whether a call is still active. Is there a way to activate a RTCP call control, e.g. Asterisk should hang up when he stops receiving RTCP messages? Have you looked at the rtptimeout and rtpholdtimeout options? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.ga...@grupoamper.com | www.grupoamper.com -- *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.ga...@grupoamper.com | www.grupoamper.com -- This message and any attachments are intended only for the use of the individual to whom they are addressed and it may contain information that is privileged or confidential. If you have received this communication by mistake, please notify us immediately by e-mail or telephone.The storage, recording, use or disclosure of this e-mail and its attachments by anyone other than the intended recipient is strictly prohibited. This message has been verified using antivirus software; however, the sender is not responsible for any damage to hardware or software resulting from the presence of any virus. Este mensaje y cualquier anexo son exclusivamente para la persona a quien van dirigidos y pueden contener información privilegiada o confidencial. Si usted ha recibido esta comunicación por error, le agradecemos notificarlo de inmediato por esta misma vía o por teléfono. Está prohibida su retención, grabación, utilización o divulgación con cualquier propósito. Este mensaje ha sido verificado con software antivirus; sin embargo, el remitente no se hace responsable en caso de que en éste o en los archivos adjuntos haya presencia de algún virus que pueda generar daños en los equipos o programas del destinatario. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP call control via RTCP
Hello, We are using Asterisk 1.4 as call distribution system with simple queues for SIP calls. With high load (4000 calls/hour) some calls remain in queue forever (until queue's max wait time) in spite of being hung up already by the caller. It seems that when a BYE is lost, Asterisk has no mechanism to check whether a call is still active. Is there a way to activate a RTCP call control, e.g. Asterisk should hang up when he stops receiving RTCP messages? Kind regards -- *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.ga...@grupoamper.com | www.grupoamper.com -- This message and any attachments are intended only for the use of the individual to whom they are addressed and it may contain information that is privileged or confidential. If you have received this communication by mistake, please notify us immediately by e-mail or telephone.The storage, recording, use or disclosure of this e-mail and its attachments by anyone other than the intended recipient is strictly prohibited. This message has been verified using antivirus software; however, the sender is not responsible for any damage to hardware or software resulting from the presence of any virus. Este mensaje y cualquier anexo son exclusivamente para la persona a quien van dirigidos y pueden contener información privilegiada o confidencial. Si usted ha recibido esta comunicación por error, le agradecemos notificarlo de inmediato por esta misma vía o por teléfono. Está prohibida su retención, grabación, utilización o divulgación con cualquier propósito. Este mensaje ha sido verificado con software antivirus; sin embargo, el remitente no se hace responsable en caso de que en éste o en los archivos adjuntos haya presencia de algún virus que pueda generar daños en los equipos o programas del destinatario. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP call control via RTCP
Thank you. Yes, that should work. But if I understand it correctly, only if there's no silence detection activated. Otherwise, when silence is detected no RTP would be send, so that rtptimeout would hang up a still active call. I there no option to use RTCP? Not even in Asterisk 11? Regards On Mon, May 12, 2014 at 2:12 PM, Matt Behrens m...@zigg.com wrote: On May 12, 2014, at 5:02 AM, Jan Gaida jan.ga...@grupoamper.com wrote: We are using Asterisk 1.4 as call distribution system with simple queues for SIP calls. With high load (4000 calls/hour) some calls remain in queue forever (until queue's max wait time) in spite of being hung up already by the caller. It seems that when a BYE is lost, Asterisk has no mechanism to check whether a call is still active. Is there a way to activate a RTCP call control, e.g. Asterisk should hang up when he stops receiving RTCP messages? Have you looked at the rtptimeout and rtpholdtimeout options? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.ga...@grupoamper.com | www.grupoamper.com -- This message and any attachments are intended only for the use of the individual to whom they are addressed and it may contain information that is privileged or confidential. If you have received this communication by mistake, please notify us immediately by e-mail or telephone.The storage, recording, use or disclosure of this e-mail and its attachments by anyone other than the intended recipient is strictly prohibited. This message has been verified using antivirus software; however, the sender is not responsible for any damage to hardware or software resulting from the presence of any virus. Este mensaje y cualquier anexo son exclusivamente para la persona a quien van dirigidos y pueden contener información privilegiada o confidencial. Si usted ha recibido esta comunicación por error, le agradecemos notificarlo de inmediato por esta misma vía o por teléfono. Está prohibida su retención, grabación, utilización o divulgación con cualquier propósito. Este mensaje ha sido verificado con software antivirus; sin embargo, el remitente no se hace responsable en caso de que en éste o en los archivos adjuntos haya presencia de algún virus que pueda generar daños en los equipos o programas del destinatario. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
