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