[Freeswitch-users] gsm file record/playback

Fri Jun 6 14:02:30 PDT 2008

Thanks a lot Anthony !!!

It helped when I loaded mod_native_file before mod_voipcodecs.

Many thanks again,

Miroslav

On Fri, Jun 6, 2008 at 3:53 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try reversing the order that you load mod_voipcodecs and mod_sndfile in
> your modules.conf.xml
> iirc there is some symbol collision between the 2 that we need to resolve
> somehow.
>
>
> On Fri, Jun 6, 2008 at 1:29 PM, Miroslav Mostic <nuvovoip at gmail.com>
> wrote:
>
>> It is not inbound codec issue. When I enable only GSM codec in my x-lite I
>> can see that GSM is negotiated inbound codec. Even in that case recording in
>> gsm is not working. However recording in uncompressed PCM wav format is
>> working properly.
>> I have the same behaviour with rc3 and some snapshot release after that,
>> but for the sake of this test I have default Freeswitch 1.0 configuration on
>> the separate box.
>>
>> Could it be that compressed gsm encoded recordings are not supported
>> anymore?
>>
>> Regards,
>>
>> Miroslav
>>
>>   On Tue, Jun 3, 2008 at 1:49 PM, Miroslav Mostic <nuvovoip at gmail.com>
>> wrote:
>>
>>>
>>> G711 u-Law. I was calling from aastra 9133i and X-lite v3.0 and I have
>>> default codec configuration of Freeswitch 1.0. After installation of
>>> Freeswitch 1.0 the only thing that are changed from default config are sip
>>> port 5090 on internal sip profile and additional extensions 9991 and 9992 in
>>> default.xml dialplan. In both scenarios I have the same result.
>>>
>>> Console output when I dial from aastra 9133i :
>>>
>>> 2008-06-03 12:52:18 [DEBUG] sofia.c:1695 sofia_handle_sip_i_state()
>>> Remote SDP:
>>> v=0
>>> o=MxSIP 0 1966242209 IN IP4 172.22.2.102
>>> s=SIP Call
>>> c=IN IP4 172.22.2.102
>>> t=0 0
>>> m=audio 3000 RTP/AVP 0 8 18 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:18 G729/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=ptime:30
>>> a=silenceSupp:on - - - -
>>> 2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp()
>>> Audio Codec Compare [PCMU:0:8000]/[G722:9:8000]
>>> 2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp()
>>> Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000]
>>> 2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp()
>>> Audio Codec Compare [PCMU:0:8000]/[PCMA:8:8000]
>>> 2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp()
>>> Audio Codec Compare [PCMU:0:8000]/[GSM:3:8000]
>>> 2008-06-03 12:52:18 [DEBUG] sofia_glue.c:2137 sofia_glue_negotiate_sdp()
>>> Substituting codec PCMU at 30i@8000h
>>> 2008-06-03 12:52:18 [DEBUG] sofia_glue.c:1392 sofia_glue_tech_set_codec()
>>> Set Codec sofia/internal/1006 at 10.47.40.56:5090 PCMU/8000 30 ms 240
>>> samples
>>> Console output when I dial from X-Lite Version 3.0:
>>>
>>> 2008-06-03 12:57:03 [DEBUG] sofia.c:1695 sofia_handle_sip_i_state()
>>> Remote SDP:
>>> v=0
>>> o=- 4 2 IN IP4 172.22.2.141
>>> s=CounterPath X-Lite 3.0
>>> c=IN IP4 172.22.2.141
>>> t=0 0
>>> m=audio 51672 RTP/AVP 0 8 3 101
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=alt:1 1 : uY3hci3U 9llPveYf 172.22.2.141 51672
>>> 2008-06-03 12:57:03 [DEBUG] switch_core_state_machine.c:365
>>> switch_core_session_run() sofia/internal/1004 at 10.47.40.56:5090 Running
>>> State Change CS_NEW
>>> 2008-06-03 12:57:03 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp()
>>> Audio Codec Compare [PCMU:0:8000]/[G722:9:8000]
>>> 2008-06-03 12:57:03 [DEBUG] sofia_glue.c:2098 sofia_glue_negotiate_sdp()
>>> Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000]
>>> 2008-06-03 12:57:03 [DEBUG] sofia_glue.c:1392 sofia_glue_tech_set_codec()
>>> Set Codec sofia/internal/1004 at 10.47.40.56:5090 PCMU/8000 20 ms 160
>>> samples
>>>
>>> Regards,
>>>
>>> Miroslav
>>>   On Mon, Jun 2, 2008 at 8:52 PM, Brian West <brian at freeswitch.org>
>>> wrote:
>>>
>>>> What codec are you using on the inbound side?
>>>>
>>>> /b
>>>>
>>>>
>>>>
>>>>
>>>> On 6/2/08 4:27 PM, "Miroslav Mostic" <nuvovoip at gmail.com> wrote:
>>>>
>>>>   Hi everyone!
>>>>
>>>> I am probably missing something very basic, but I have problems with
>>>> simple file record and playback in gsm file format.
>>>>
>>>> I have just installed 1.0 release system and I changed only default
>>>> dialplan with following two extensions:
>>>>
>>>> <extension name="gsm_record">
>>>>       <condition field="destination_number" expression="^9991$">
>>>>         <action application="answer"/>
>>>>         <action application="record" data="/tmp/rec.gsm"/>
>>>>       </condition>
>>>>     </extension>
>>>>     <extension name="gsm_playback">
>>>>       <condition field="destination_number" expression="^9992$">
>>>>         <action application="answer"/>
>>>>         <action application="playback" data="/tmp/rec.gsm"/>
>>>>       </condition>
>>>>     </extension>
>>>>
>>>> When I place call to 9991extension I see INFO message about opening
>>>> /tmp/rec.gsm, no error messages and when I hangup I can see that file
>>>> /tmp/rec.gsm exists. However when I try to play this file using 9992
>>>> extension the only thing I can hear is noise.
>>>>
>>>> When I just change name to /tmp/rec.wav in both extensions everything is
>>>> working fine.
>>>>
>>>> What am I missing?
>>>>
>>>> Many regards,
>>>>
>>>> Miroslav
>>>>
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>>>
>>
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>
>
> --
> Anthony Minessale II
>
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