[Freeswitch-dev] [Freeswitch-users] G723 timer problem
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.
> On Jul 1, 2009, at 4:04 PM, Muhammad Shahzad wrote:
> 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
> Linksys/Sipura aka Cisco
> 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-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
Anthony Minessale II
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the Freeswitch-dev