Hello, I am a new user of FreeSwitch and I try to switch from Asterisk to FreeSwitch. For now, I just registered a SIP line and I made an extension in the dialplan to playback music on hold when I call : <context name="default"> <extension name="test"> <condition> <!--<condition field="destination_number" expression="^9000$">--> <action application="answer"/> <action application="playback" data="$${hold_music}"/> </condition> </extension> </context> When I call I have a problem with the sound, it is very bad, I hear very fast crackle. On the console I can see a warning during my call and that's all: 2010-11-12 13:02:45.119517 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous@anonymous.invalid [c27034e2-ee54-11df-ad91-7fc19b304698] 2010-11-12 13:02:45.122548 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:02:45.125555 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous@anonymous.invalid] has been answered 2010-11-12 13:02:45.508790 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20 If I add <X-PRE-PROCESS cmd="set" data="timer_name=soft"/> in vars.xml, when I call the sound is good but I still get the warning : 2010-11-12 13:05:42.789161 [NOTICE] switch_channel.c:784 New Channel sofia/external/anonymous@anonymous.invalid [2c565760-ee55-11df-a0d9-0f91e325cd76] 2010-11-12 13:05:42.791179 [INFO] mod_dialplan_xml.c:331 Processing Anonymous <anonymous>->test in context default 2010-11-12 13:05:42.794179 [NOTICE] mod_dptools.c:920 Channel [sofia/external/anonymous@anonymous.invalid] has been answered 2010-11-12 13:05:43.174443 [WARNING] mod_sofia.c:1036 Asynchronous PTIME not supported, changing our end from 30 to 20 2010-11-12 13:05:43.175461 [INFO] sofia.c:709 sofia/external/anonymous@anonymous.invalid Update Callee ID to "anonymous" <anonymous> Is this normal? Why should I add this line in vars.xml for it to work ? What does this warning? How to solve it? Thank you in advance for your help. Loïc