Loïc,

I would suspect the issue to be on the leg B, because I use  a Siemens C470IP on FS regularly for testing, and I never had a such warning.

David Ponzone  Direction Technique
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26

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Le 12/11/2010 à 18:45, Loïc a écrit :

Hi Anthony,

I have this warning as well with my Siemens C470IP as my Linksys
SPA942.
And my Linksys has been configured with a RTP Packet Size to 0.020. If
I change to 0.20 I have the same warning :-/

Loïc



On Fri, 12 Nov 2010 11:23:46 -0600, Anthony Minessale
<anthony.minessale@gmail.com> wrote:
What is your device?

is it a cisco/linksys/sipura ?

you can change the packet size from 0.30 to 0.20 in the UI of your
phone.




On Fri, Nov 12, 2010 at 6:09 AM, Loïc <loic.latreille@ovh.net> wrote:
 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



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Anthony Minessale II

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