_________________________________________________________________________Hi,
I’ve purchased 6 g729 codec license and activated it successfully.
When I establish a call from my Service Provider ( using G729 ), call is received through the SIP Trunk.
In FreeSwitch I have routed the call to an extension.
When a sip client ( MircoSIP) logged into that extension ( configured for G.711 u-law and G.711 A-law)
Call is not getting transferred and it returns currently not available.
But When the sip client ( MircoSIP) is configured for G.711 u-law and G.711 A-law and G729 8kHz, then the call is going through.
My concern is even after purchasing G729 license, Transcoding is not happening.
If anyone has come across this problem and found a solution, your inputs are appreciated. Thank you.
My Configuration values I the respective files are
/usr/local/freeswitch/ conf/sip_profiles/external/EXTSIPGW.xml :
<include>
<gateway name=" EXTSIPGW ">
<param name="proxy" value="192.168.168.34:5060"/>
<param name="register" value="false"/>
<param name="caller-id-in-from" value="true"/>
<param name="username" value="not-used"/>
<param name="password" value="not-used"/>
<param name="contact-params" value="tport=tcp"/>
<param name="register-transport" value="tcp"/>
</gateway>
</include>
/usr/local/freeswitch/ conf/vars.xml :
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,G729,H264,VP8"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G722,PCMU,PCMA,G729,H264,VP8"/>
/usr/local/freeswitch/ conf/autoload_configs/modules.conf.xml :
<!-- Codec Interfaces -->
<load module="mod_spandsp"/>
<load module="mod_g723_1"/>
<!--<load module="mod_com_g729"/>-->
<load module="mod_com_g729"/>
<load module="mod_amr"/>
<!--<load module="mod_ilbc"/>-->
<!--<load module="mod_h26x"/>-->
<load module="mod_b64"/>
<!--<load module="mod_siren"/>-->
<!--<load module="mod_isac"/>-->
<load module="mod_opus"/>
fs_cli: Log
freeswitch@debian> g729_info
G.729A license available => 6
G.729A license allocated => 0
Regards,
Anand Sivaraj S
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