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naf edited this page Aug 14, 2018
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- strange behavior with lack of ringback tone on early media
- dialing some numbers results in GV playing a ringback tone over the early media, whereas dialing other numbers results in GV playing a silent audio stream instead
- possibly the numbers that dont ring are other google voice / project fi numbers?
- using the Dial 'r' option ignores the early media and replaces it with an out-of-band 180 Ringing message (which should cause the endpoint device to ring), but this results in GV terminating the call after ~20s with a 603 media inactivity
- dialing some numbers results in GV playing a ringback tone over the early media, whereas dialing other numbers results in GV playing a silent audio stream instead
- can't access voicemail if caller id is set to actual GV number (credit to DSLreports user RonR)
- GV typically ignores the caller id specified in From field of outbound INVITEs, but if you (or the gui that configures asterisk for you) sets the caller id in extensions.conf for the call [i.e.
exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=##########)] to your own GV number then GV rejects calls to your own GV number with a 603 Decline - any other caller id string other than your gv number works fine
- GV typically ignores the caller id specified in From field of outbound INVITEs, but if you (or the gui that configures asterisk for you) sets the caller id in extensions.conf for the call [i.e.
- WARNING[22501]: pjproject:0 <?>: SSL STATUS_FROM_SSL_ERR (connecting): Level: 0 err: <336151598> [SSL routines-ssl3_read_bytes-tlsv1 alert protocol version] len: 0
- Need to upgrade openssl. verified working against openssl 1.0.2o and 1.1.0h.
- ERROR[13113]: pjproject:0 <?>: icess0x369fba4 ...Error sending STUN request: Network is unreachable
- In order to work behind firewalls, SIP uses a protocol named ICE to try to detect your NAT settings automatically by connecting to stun.l.google.com using every local interface it can find (and asking what its external IP is so it can populate the ICE candidate list in the sdp it uses when making an outbound call). Most of the interfaces probably work, but for any one that fails it claims it to be an error. On my machine, its just the ipv6 address being unable to connect.
- ERROR[7894] ari/config.c: No configured users for ARI
- Happens when using the samples configuration out of the box.
- ERROR[8149] pjproject: sip_msg Header with no vptr encountered!! Current buffer: ...
- asterisk patches pjproject to show this error when a printing an in-memory message header results in no text being printed, in theory to detect bad in-memory headers. However, a 'hide' route header is hidden by design, so it results in no text and you see this 'error' for almost every message being sent.
- now fixed in upstream