Hi When using an extension to my android gingerbread nexus one, calls drop after a n minutes of call due as per the following [Jun 21 09:34:37] == Begin MixMonitor Recording SIP/nexusone-0a39 [Jun 21 09:34:37] -- Executing [0031xxxxxxxx0@default:4] Dial(" ... [Jun 21 09:34:37] -- Called 45xx:xxxxxx...@iax.voixxxxe.xl/00x ... [Jun 21 09:34:37] -- Call accepted by 84.243.247.100 (format al ... [Jun 21 09:34:37] -- Format for call is alaw [Jun 21 09:34:38] -- IAX2/4506-8090 is making progress passing [Jun 21 09:34:42] -- IAX2/4506-8090 answered SIP/nexusone-0a39a .c: Context 'macro-notifymobile' for macro 'notifymobile' lacks 's' logger.c: [Jun 21 09:35:44] -- Saved useragent "SIPAUA/0.1.001" for peer logger.c: [Jun 21 09:42:39] -- Unregistered SIP 'nexusone' logger.c: [Jun 21 09:42:39] -- Unregistered SIP 'nexusone' logger.c: [Jun 21 09:42:39] -- Registered SIP 'nexusone' at 77.165.58.113 logger.c: [Jun 21 09:42:39] -- Saved useragent "SIPAUA/0.1.001" for peer logger.c: [Jun 21 09:45:57] -- Unregistered SIP 'nexusone' logger.c: [Jun 21 09:45:58] -- Registered SIP 'nexusone' at 77.165.58.113 logger.c: [Jun 21 09:45:58] -- Saved useragent "SIPAUA/0.1.001" for peer logger.c: [Jun 21 10:13:23] -- Unregistered SIP 'nexusone' logger.c: [Jun 21 10:13:23] -- Unregistered SIP 'nexusone'
ex sip.conf; [nexusone] type=friend context=default username=nexusone secret=Iainttelling nat=no disallow=all allow=gsm *CLI> sip show peers Name/username Host Dyn Nat ACL Port Status nexusone/nexusone (Unspecified) D 0 Unmonitored How could I prevent these calls from dropping? --- Eric Smith -- _____________________________________________________________________ -- 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