Hi! I’m getting really strange things, or I’m just missed something. My phone is dials to freeswitch with this this line in log 2016-03-06 08:31:56.681036 [DEBUG] sofia.c:6770 Remote SDP: v=0 o=root 1697549695 1697549695 IN IP4 <EXTERNAL IP HERE> s=call c=IN IP4 <EXTERNAL IP HERE> t=0 0 m=audio 26894 RTP/AVP 9 0 8 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=nortpproxy:yes 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4161 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:4216 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2016-03-06 08:31:56.681036 [DEBUG] switch_core_media.c:2906 Set Codec sofia/internal/10@consertis.securenetvox.net G722/8000 20 ms 160 samples 64000 bits 1 channels And when switches to external profile, I see 2016-03-06 08:31:56.721010 [DEBUG] sofia_glue.c:1257 sofia/external/00972543279009 sending invite version: 1.6.6 git d2d0b32 2016-01-11 20:16:12Z 64bit Local SDP: v=0 o=FreeSWITCH 1457219712 1457219713 IN IP4 10.0.20.71 s=FreeSWITCH c=IN IP4 10.0.20.71 t=0 0 m=audio 29804 RTP/AVP 9 101 13 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 a=sendrecv The question is - why only G722 left? Across dialplan there is no things like inherit_codec, bypass media or other codec-related stuff Profiles external ext-sip-ip [10.0.20.71] rtp-timeout-sec [300] rtp-hold-timeout-sec [1800] tls [false] tls-only [false] tls-bind-params [transport=tls] tls-sip-port [5081] tls-cert-dir [/usr/local/freeswitch/conf/ssl] tls-passphrase [] tls-verify-date [true] tls-verify-depth [2] tls-verify-in-subjects [] tls-version [tlsv1] tls-verify-policy [all] odbc-dsn [pgsql://hostaddr=127.0.0.1 port=5432 dbname=freeswitch user=fusionpbx password=pass options='' application_name='freeswitch'] track-calls [true] inbound-codec-negotiation [greedy] debug [0] user-agent-string [FreeSWITCH] sip-trace [no] sip-capture [no] rfc2833-pt [101] sip-port [5080] dialplan [XML] context [public] dtmf-duration [2000] inbound-codec-prefs [G7221@32000h,G7221@16000h,G722,PCMU,PCMA,OPUS,SILK] outbound-codec-prefs [PCMU,PCMA,GSM] hold-music [local_stream://default] zrtp-passthru [true] rtp-timer-name [soft] local-network-acl [localnet.auto] manage-presence [false] nonce-ttl [60] auth-calls [false] rtp-ip [10.0.20.71] sip-ip [10.0.20.71] ext-rtp-ip [10.0.20.71] internal tls-cert-dir [/usr/local/freeswitch/conf/ssl] tls-passphrase [] tls-verify-date [true] tls-verify-depth [2] tls-verify-in-subjects [] tls-version [tlsv1] nonce-ttl [60] auth-calls [true] inbound-reg-force-matching-username [true] auth-all-packets [false] ext-rtp-ip [10.0.20.71] ext-sip-ip [10.0.20.71] rtp-timeout-sec [300] rtp-hold-timeout-sec [1800] tls-verify-policy [all] multiple-registrations [contact] enable-timer [false] dbname [share_presence] send-presence-on-register [true] inbound-codec-negotiation [greedy] NDLB-force-rport [safe] challenge-realm [auto_to] outbound-proxy [10.0.20.70] track-calls [true] odbc-dsn [pgsql://hostaddr=127.0.0.1 port=5432 dbname=freeswitch user=fusionpbx password=btgJek49 options='' application_name='freeswitch'] nat-options-ping [true] liberal-dtmf [true] all-reg-options-ping [true] force-publish-expires [true] unregister-on-options-fail [true] user-agent-string [FreeSWITCH] debug [0] sip-trace [no] sip-capture [no] watchdog-enabled [no] watchdog-step-timeout [30000] watchdog-event-timeout [30000] log-auth-failures [true] forward-unsolicited-mwi-notify [false] context [public] rfc2833-pt [101] sip-port [5060] dialplan [XML] dtmf-duration [2000] inbound-codec-prefs [G7221@32000h,G7221@16000h,G722,PCMU,PCMA,OPUS,SILK] outbound-codec-prefs [G7221@32000h,G7221@16000h,G722,PCMU,PCMA,OPUS,SILK] rtp-timer-name [soft] rtp-ip [10.0.20.71] sip-ip [10.0.20.71] hold-music [local_stream://default] apply-nat-acl [nat.auto] aggressive-nat-detection [true] apply-inbound-acl [domains] local-network-acl [localnet.auto] record-path [/usr/local/freeswitch/recordings] record-template [${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}] manage-presence [true] presence-probe-on-register [true] manage-shared-appearance [true] tls [false] tls-only [false] tls-bind-params [transport=tls] tls-sip-port [5061] Tried with indbound-late-negotiation=false, also not helps… Can you please, point, what is missing? Thanks -- Best regards, Igor