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@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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