[Freeswitch-users] change the remote RTP port after sample rate doesnot match
Erwin Davis
davis.erwin at gmail.com
Wed Dec 2 12:08:37 PST 2009
Hi, Anthony,
Thanks for your reply.
When I type the command below, I got the error,
Unknown target hd-sound-install
make[1]: *** [hd-sound-install] Error 1
make: *** [hd-sound-install] Error 2
I found out that under /usr/local/freeswitch/sounds/en/us/callie/voicemail,
there are directories, 8000, 16000, 32000, 48000 for recorded voicemail
greetings. It should explain why at first FS played in right sample rate.
But after playing serveral time, FS complained about sample rate not
matching. Any clue? Thanks,
On 12/2/09, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
> you must only have 8k sounds so the resample is when it's playing files
>
> try make hd-sounds-install to install 16k sounds too
>
>
>
>
>
> On Wed, Dec 2, 2009 at 12:41 PM, Erwin Davis <davis.erwin at gmail.com>wrote:
>
>> Hi, I got a weird issue when I dialed an extension and listen to a
>> recorded voice mail greeting message.
>> After playing a couple of time of the greeting, the FS printed the warning
>> of "sample rate not matching", then
>> send the audio to a different remote RTP port. See the log below,
>>
>>
>> 2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec
>> Activated L16 at 16000hz 1 channels 20ms
>> 2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649
>> sofia/internal/1003 at xxx.yyy.zzz.31 receive message
>> [TRANSCODING_NECESSARY]
>> 2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing
>> file
>> 2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language
>> specified - Using [en]
>> 2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle
>> play-file:[voicemail/vm-record_message.wav] (en:en)
>> 2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec
>> Activated L16 at 16000hz 1 channels 20ms
>> 2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649
>> sofia/internal/1003 at xxx.yyy.zzz.31 receive message
>> [TRANSCODING_NECESSARY]
>> 2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done playing
>> file
>> 2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649
>> sofia/internal/1003 at xxx.yyy.zzz.31 receive message
>> [TRANSCODING_NECESSARY]
>> 2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate
>> doesn't match
>> 2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec
>> Activated
>> 2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port
>> from xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748
>> 2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore
>> original codec.
>> 2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less
>> than minimum record length: 3, discarding it.
>> 2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language
>> specified - Using [en]
>> 2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle
>> play-file:[voicemail/vm-too-small.wav] (en:en)
>> 2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec
>> Activated L16 at 16000hz 1 channels 20ms
>> 2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649
>> sofia/internal/1003 at xxx.yyy.zzz.31 receive message [
>>
>>
>> the original codec is wideband 16kHz Speex and the wireshark shows that
>> the FS used the same codec. I used FS 1.04 in fedora 8.
>> I have two questions here,
>> (1) why does FS report "Sample rate doesn't match"? is it a bug or
>> configuration issue?
>> (2) Why does FS change the RTP port ? how to fix it?
>>
>> Thanks,
>>
>> Regards,
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/9f8ee643/attachment-0002.html
More information about the FreeSWITCH-users
mailing list