[Freeswitch-users] Bad sound with crackle

David Ponzone david.ponzone at ipeva.fr
Sat Nov 13 02:58:03 PST 2010


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
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.




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 at 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 at 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 at 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 at 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 at 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 at 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 at 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
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> 
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> 
>> 
>> --
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20101113/706a705d/attachment-0001.html 


More information about the FreeSWITCH-users mailing list