[Freeswitch-dev] [Freeswitch-users] G723 timer problem

Anthony Minessale anthony.minessale at gmail.com
Wed Jul 1 14:30:12 PDT 2009

try revision 14095 or higher.
This adds the ability to use g723 at 60ms,

If this does not work,

set the param rtp-autofix-timing to true in your sip profile.

On Wed, Jul 1, 2009 at 4:13 PM, Brian West <brian at freeswitch.org> wrote:

> 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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Anthony Minessale II

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