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