Re: Freepbx + grandstream GS-HT702 woes

[§ñühwö£f]

Evan Platt wrote:
> I have Freepbx and a grandstream hs-ht702.
>
> I've tried every combination of codecs, to no avail. Initially, I
> configured (by accident) the grandstream to connect directly to
> flowroute. It worked. Then, I reconfigured the grandstream to
> correctly connect to my pbx server. I get a dialtone, and dial, but
> get no audio. People can hear me, but I can't hear anything. I've
> tried codec by codec, only using one codec on both the pbx and on the
> grandstream.
>
> I have in Codecs on the server g723 and lbc enabled, and on the
> grandstream g723 & ilbc. In the extension, I have "g723&ilbc" for
> Allow.
>
> Here's one of the most recent logs.
>
> Any suggestions / ideas?
>
> Thanks. (Not sure if more is needed from the logs- numbers munged.)
>
> [2013-11-15 23:21:18] VERBOSE[29893][C-00000031] netsock2.c: == Using
> SIP RTP TOS bits 184
> [2013-11-15 23:21:18] VERBOSE[29893][C-00000031] netsock2.c: == Using
> SIP RTP CoS mark 5
> [2013-11-15 23:21:18] VERBOSE[29893][C-00000031] app_dial.c: -- Called
> SIP/flowroute/###########
> [2013-11-15 23:21:18] WARNING[29893][C-00000031] channel.c: No path to
> translate from SIP/flowroute-00000032 to SIP/1-00000031
> [2013-11-15 23:21:18] VERBOSE[29893][C-00000031] app_macro.c: == Spawn
> extension (macro-dialout-trunk, s, 22) exited non-zero on
> 'SIP/1-00000031' in macro 'dialout-trunk'
> [2013-11-15 23:21:18] VERBOSE[29893][C-00000031] pbx.c: == Spawn
> extension (from-internal, ###########, 5) exited non-zero on
> 'SIP/1-00000031'
> [2013-11-15 23:21:18] VERBOSE[29893][C-00000031] pbx.c: -- Executing
> [h@from-internal:1] Hangup("SIP/1-00000031", "") in new stack
> [2013-11-15 23:21:18] VERBOSE[29893][C-00000031] pbx.c: == Spawn
> extension (from-internal, h, 1) exited non-zero on 'SIP/1-00000031'

Looks like a klew from their website:

Sometimes people create system audio files using an external sound file
editor, such as Audacity, in order to get better sound quality. What
they don't realize is that Asterisk is very picky about the format of
audio files it will play back. For example, if the file is .wav file
format, Asterisk wants a file recorded at 8000 Hz, 16 bit, monaural
(a.k.a. single channel) format and if you directly upload a file in any
other format, the CLI may show that the file is being played, but
callers hear nothing. If normal system files play correctly but the
files you've created do not, check the format, especially if you've
directly copied it to a particular location on the system instead of
importing it with the System Recordings module.

Sound "file" needs to be exactly 16 bit mono, single channel. Theres a
forum for users, might wanna use it.
Jus Sayin...
http://www.freepbx.org/support/documentation/howtos/howto-resolving-audio-problems

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