<div>Hi,</div>
<div> </div>
<div>I am getting problem when one UA is xlite and another UA is another sip application. </div>
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<div>When I call from xlite to a sip application, I am getting noise:</div>
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<div>I have tried these:</div>
<div> <extension name="redial"><br> <condition field="destination_number" expression="^3000"><br> <action application="bridge" data="{absolute_codec_string='GSM,PCMU'}sofia/<a href="http://192.168.1.191/4540">192.168.1.191/4540</a>"/><br>
</condition><br> </extension><br></div>
<div> <extension name="redial"><br> <condition field="destination_number" expression="^3000"><br> <action application="bridge" data="sofia/<a href="http://192.168.1.191/4540">192.168.1.191/4540</a>"/><br>
</condition><br> </extension><br></div>
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<div>show channels give me the following:</div>
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<div>c5f42dec-646a-4675-af40-c4d173c8a7c7,inbound,2009-05-23 10:36:30,1243089390,sofia/internal/<a href="mailto:1000@192.168.1.191">1000@192.168.1.191</a>,CS_EXECUTE,1000,1000,192.168.1.193,3000,bridge,sofia/<a href="http://192.168.1.191/4540,XML,public,GSM,8000,GSM,8000">192.168.1.191/4540,XML,public,GSM,8000,GSM,8000</a><br>
790d9b2a-88b9-4521-8934-31b059e04e7b,outbound,2009-05-23 10:36:30,1243089390,sofia/internal/4540,CS_CONSUME_MEDIA,1000,1000,192.168.1.193,4540,,,XML,public,,,,</div>
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<div>The sip application and xlite is working fine ( voice is clear ) if I use Asterisk with the following line in sip.conf:</div>
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<div>[4540]<br>canreinvite=no<br>type=friend<br>context=sip-external<br><font color="#ff6666">allow=gsm</font><br>host=dynamic</div>
<div><br>[1000]<br>canreinvite=no<br>type=friend <br>context=sip-external<br>allow=gsm <br>host=dynamic </div>
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<div>Does anyone know how to mimic the same behavior in Freeswitch?</div>
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<div>Thanks,</div>
<div>JB</div>
<div> </div>