Dear list, I am playing with Sipura 3000 since last week. Through the wiki pages I could get running it reasonably well. My setup is that of a Sipura, linked with a local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out from the cordless phone trough the Sipura - Asterisk link, using the PTSN line on the other port of the Sipura. So far, so good. BUT: While I can receive a phone call arriving on the PSTN port, it is correctly routed to the cordless phone on the other spa port with the faked callerid trick found in the wiki, the spa does not seems to detect the end of the call. So after the other party ends the call, I end up with an open SIP channel on the asterisk server, and what is way worse, the SPA can not accept or dial out any other call on the PSTN line. I have to manually reset it (and restart the asterisk server to get rid of the zombie SIP channel). The point is in other words how to setup the end of a call detection. I assume that the phone line I am using is set up with italian (or european / etsi) standards. How should I setup the end of call detection for this kind of pstn line? Thanks for any help, Aldo
I don't know if it is a phone like issue or not, but try the SPA-3000 setup at http://geekgazette.com. -Kerry -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Aldo Bergamini Sent: Friday, May 20, 2005 6:35 AM To: Asterisk Users Subject: [Asterisk-Users] Sipura 3000 Question Dear list, I am playing with Sipura 3000 since last week. Through the wiki pages I could get running it reasonably well. My setup is that of a Sipura, linked with a local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out from the cordless phone trough the Sipura - Asterisk link, using the PTSN line on the other port of the Sipura. So far, so good. BUT: While I can receive a phone call arriving on the PSTN port, it is correctly routed to the cordless phone on the other spa port with the faked callerid trick found in the wiki, the spa does not seems to detect the end of the call. So after the other party ends the call, I end up with an open SIP channel on the asterisk server, and what is way worse, the SPA can not accept or dial out any other call on the PSTN line. I have to manually reset it (and restart the asterisk server to get rid of the zombie SIP channel). The point is in other words how to setup the end of a call detection. I assume that the phone line I am using is set up with italian (or european / etsi) standards. How should I setup the end of call detection for this kind of pstn line? Thanks for any help, Aldo _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
In the advanced options there are a few options for hang-up detection including tone detection, and silence detection. They also have parameters to adjust timing and sensitivy. IIRC, they are not enabled by default. -----Original Message----- From: Aldo Bergamini [mailto:aab.lists@nb-a.com] Sent: Friday, May 20, 2005 8:35 AM To: Asterisk Users Subject: [Asterisk-Users] Sipura 3000 Question Dear list, I am playing with Sipura 3000 since last week. Through the wiki pages I could get running it reasonably well. My setup is that of a Sipura, linked with a local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out from the cordless phone trough the Sipura - Asterisk link, using the PTSN line on the other port of the Sipura. So far, so good. BUT: While I can receive a phone call arriving on the PSTN port, it is correctly routed to the cordless phone on the other spa port with the faked callerid trick found in the wiki, the spa does not seems to detect the end of the call. So after the other party ends the call, I end up with an open SIP channel on the asterisk server, and what is way worse, the SPA can not accept or dial out any other call on the PSTN line. I have to manually reset it (and restart the asterisk server to get rid of the zombie SIP channel). The point is in other words how to setup the end of a call detection. I assume that the phone line I am using is set up with italian (or european / etsi) standards. How should I setup the end of call detection for this kind of pstn line? Thanks for any help, Aldo _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
asterisk-users-request@lists.digium.com is believed to have said:> >I don't know if it is a phone like issue or not, but try the SPA-3000 setup >at http://geekgazette.com. >-Kerry >Kerry, thanks for the hint. A first try did not get better results, but I was doing it very quickly.. Aldo
asterisk-users-request@lists.digium.com is believed to have said:> >In the advanced options there are a few options for hang-up detection >including tone detection, and silence detection. They also have parameters >to adjust timing and sensitivy. IIRC, they are not enabled by default. >Nathan, thanks: this is something I still have to try systematically. Has anybody hints to give as where to find understandable (by the uninitiated..) documentation of what the telecoms do in Europe with regards to signaling the hangup condition? Aldo