[Freeswitch-users] need hear solving a noise problem

[Juan Backson]

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 at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090523/4578466c/attachment-0002.html