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 <[email protected]> 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. > > > > _______________________________________________ > Freeswitch-dev mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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