Hello since upgrading from 15648 to "FreeSWITCH Version 1.0.head (svn-17188)" we see that - despite only allowing PMCA and PCMU codecs in vars.xml - Freeswitch offers all available codec to the called phone. Any idea how to solve this? Best regards Peter See Invite example U 10.xx.xx.1:5060 -> 10.xx.xx.172:2048 INVITE sip:31@10.xx.xx.172:2048;line=hxbudrul SIP/2.0. Via: SIP/2.0/UDP 10.xx.xx.1;rport;branch=z9hG4bKZD66c84339SHH. Max-Forwards: 70. From: "069xxxxxxxx" <sip:069xxxxxxxx@10.xx.xx.1>;tag=5evr6508K9S3K. To: <sip:31@10.xx.xx.172:2048;line=hxbudrul>. Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5. CSeq: 129803060 INVITE. Contact: <sip:mod_sofia@10.xx.xx.1:5060>. User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 920. X-FS-Support: update_display. Remote-Party-ID: "069xxxxxxxx" <sip:069xxxxxxxx@10.xx.xx.1>;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.1. s=FreeSWITCH. c=IN IP4 10.xx.xx.1. t=0 0. m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98 122 9 121 3 101 13. a=rtpmap:115 G7221/32000. a=fmtp:115 bitrate=48000. a=rtpmap:96 AMR/8000. a=fmtp:96 octet-align=0. a=rtpmap:99 SPEEX/32000. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:7 LPC/8000. a=rtpmap:124 G726-16/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:6 DVI4/16000. a=rtpmap:123 G726-24/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:10 L16/22050. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode #