[Asterisk-Users] g726 Codec w/ GrandStream 1.0.4.50 Firmware

[Greg Boehnlein]

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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