If I call my extension from another extension on my asterisk server, the call goes straight to voicemail:
Jul 27 01:25:49] VERBOSE[13549] logger.c: -- Executing [s@macro-dial:7] Dial("SIP/marc10-b73c8c50", "SIP/200|22|rM(auto-blkvm)") in new stack
[Jul 27 01:25:49] WARNING[13549] rtp.c: Unable to set TOS to 184
[Jul 27 01:25:49] VERBOSE[13549] logger.c: -- Called 200
[Jul 27 01:25:49] VERBOSE[28246] logger.c: -- Got SIP response 486 "Busy Here" back from xx.xxx.xxx.xxx
[Jul 27 01:25:49] VERBOSE[13549] logger.c: -- SIP/200-08b1ba50 is busy
However if I call it via a DID, it rings:
[Jul 27 01:29:22] VERBOSE[15902] logger.c: -- Executing [s@macro-dial:7] Dial("SIP/marc10-b7377940", "SIP/200|22|rM(auto-blkvm)") in new stack
[Jul 27 01:29:22] WARNING[15902] rtp.c: Unable to set TOS to 184
[Jul 27 01:29:22] VERBOSE[15902] logger.c: -- Called 200
[Jul 27 01:29:22] VERBOSE[15902] logger.c: -- SIP/200-08b1ba50 is ringing
Any ideas why? The extension is using a SPA2102.
And a bonus question. Should the warning ‘Unable to set TOS to 184’ concern me?