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/[email protected]
,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
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