[Freeswitch-dev] [Freeswitch-users] G723 timer problem
Brian West
brian at freeswitch.org
Wed Jul 1 14:13:00 PDT 2009
You have two choices... set codec neg. to scrooge or get a provider
that doesn't lie about the ptime in their SDP.
/b
On Jul 1, 2009, at 4:04 PM, Muhammad Shahzad wrote:
> Hi,
>
> I am using FS svn revision 14046 and trying to send call from SIP
> Dialer to a SIP gateway using G723 in passthrough mode. Everything
> works perfect and destination rings but then call drops with
> following error on FS CLI,
>
>
> 2009-07-02 02:39:28.790508 [WARNING] mod_sofia.c:807 We were told to
> use ptime 30 but what they meant to say was 60
> This issue has so far been identified to happen on the following
> broken platforms/devices:
> Linksys/Sipura aka Cisco
> ShoreTel
> Sonus/L3
> We will try to fix it but some of the devices on this list are so
> broken who knows what will happen..
> 2009-07-02 02:39:28.790508 [WARNING] switch_core_codec.c:499 Codec
> G723 Exists but not at the desired implementation. 8000hz 60ms
>
>
> Is there any work around for this or i have downgrade my server back
> to Asterisk. :'-(
>
> Thank you.
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