Negotiated G.722 in call, then hangup.

GoogleCodeExporter commented 9 years ago

What steps will reproduce the problem?
1. While using an asterisk server, with g722 defined as the prefered codec.
The phone registers correctly but disconnects any call as soon it is started.

The following is the asterisk information.

-- Executing [5000@XXXX:1] MeetMe("SIP/XXXX-028feeb0", ",sicMp") in new stack
[Feb 13 13:02:31] WARNING[17340]: file.c:664 ast_readaudio_callback: Failed
to write frame
    -- <SIP/XXXX-028feeb0> Playing 'conf-getconfno.g722' (language 'en')
  == Spawn extension (remotephone, 5000, 1) exited non-zero on
'SIP/XXXX-028feeb0'

2. When disabling the G.722 codec on the asterisk box, the phone uses g711a
(which is the next codec in the list) and the call goes through.

What is the expected output? What do you see instead?

What version of the product are you using? On what operating system?
iPhone 2.2 with Siphon 2.0.5

Asterisk 1.6.0.5

Please provide any additional information below.

Original issue reported on code.google.com by jo...@asia.com on 13 Feb 2009 at 12:35

GoogleCodeExporter commented 9 years ago

It seems the problem comes from server, isn't it? ;-)
Could you define log level to detailed and post here ?

Original comment by samuelv0...@gmail.com on 13 Feb 2009 at 7:41

GoogleCodeExporter commented 9 years ago

The next version allows to select your codec.
By default, all codecs can be used.

Original comment by samuelv0...@gmail.com on 15 Feb 2009 at 9:51

Changed state: Fixed

BineTech / siphon

Negotiated G.722 in call, then hangup. #154