Great. It works. Thank you so much. Now i can restore my production deployment plan back to FS.
Thank you. On Wed, Jul 1, 2009 at 4:30 PM, Anthony Minessale < [email protected]> wrote: > 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 > > _______________________________________________ > 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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: [email protected] Email: [email protected]
_______________________________________________ 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
