I'm sorry about my response... I had overlooked that I only did one
30ms implementation in mod_g723_1.c, Anthony added some more to the
list so it might actually work correctly.
Thanks,
Brian
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.
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
