Hello all,
I'm trying to get the g726 codec patch contained in:
http://bugs.digium.com/bug_view_page.php?bug_id=0001104 to work with the
latest GrandStream beta firmware and I am a lot closer than I was a couple
of weeks ago with the 1.0.4.46 firmware. I am now hearing Audio that is
distinguishable with the .50 firmware release, but Asterisk is giving me
the following error messages on the console:
== Spawn extension (default, 8500, 1) exited non-zero on
'SIP/damin-8ebc'
-- Executing VoiceMailMain("SIP/damin-e62c", "") in new stack
Mar 7 12:22:30 WARNING[278542]: rtp.c:1069 ast_rtp_write: Not sure about sending
format G726 packets
Mar 7 12:22:30 WARNING[278542]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp
format for codec format G726
-- Playing 'vm-login' (language 'en')
Mar 7 12:22:30 NOTICE[278542]: rtp.c:484 ast_rtp_read: Unable to calculate samples
for format G726
Mar 7 12:22:30 WARNING[278542]: rtp.c:1069 ast_rtp_write: Not sure about sending
format G726 packets
Mar 7 12:22:30 WARNING[278542]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp
format for codec format G726
Mar 7 12:22:30 NOTICE[278542]: rtp.c:484 ast_rtp_read: Unable to calculate samples
for format G726
Mar 7 12:22:30 NOTICE[278542]: rtp.c:484 ast_rtp_read: Unable to calculate samples
for format G726
Mar 7 12:22:30 WARNING[278542]: rtp.c:1069 ast_rtp_write: Not sure about sending
format G726 packets
Mar 7 12:22:30 WARNING[278542]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp
format for codec format G726
Mar 7 12:22:30 WARNING[278542]: rtp.c:1069 ast_rtp_write: Not sure about sending
format G726 packets
Mar 7 12:22:30 WARNING[278542]: rtp.c:934 ast_rtp_raw_write: Not sure about timestamp
format for codec format G726
Any suggestions on where I should look? Could this possibly be a
configuration issue on my part?
--
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http://www.n2net.net Where everything clicks into place!
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