Great. It works. Thank you so much. Now i can restore my production deployment plan back to FS.
Thank you.
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./bOn 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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