I'm attempting to call from a SIP phone (g.729) to the SIP gateway that connects g.711a to the PSTN. It's resulting in "SERVICE UNAVAILABLE" .... If I switch my SIP phone to use g.711a, everything works fine. Any sample on how to transcode 9.729 from my SIP phone to the freeswitch server that would then transcode from g.729 to g.711a to connect to the SIP gateway? I've tried g.729 from my SIP phone to call 1000 and 5000 on the freeswitch server and that worked. And I've bought license for g.729. Thanks, Jeff. ------------------------------------------------------------------------ INVITE sip:+6568465283@155.161.200.188 SIP/2.0 Via: SIP/2.0/TCP 155.161.11.194:54927;rport;branch=z9hG4bKPjDNkdtN2t3.xjdgQ-78gpmFRuBhnjA9Qd;alias Max-Forwards: 70 From: "1001" <sip:1001@155.161.200.188>;tag=x0m7o1ogjzCoro6dBITkX2dB5U.4DX-y To: <sip:+6568465283@155.161.200.188> Contact: <sip:1001@155.161.11.194:54927;transport=TCP;ob> Call-ID: kQZCHnkGqshj5lmFItexEB8zNG1MA4W6 CSeq: 21273 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Bria iOS 3.3.4 Content-Type: application/sdp Content-Length: 234 v=0 o=- 3643948404 3643948404 IN IP4 155.161.11.194 s=cpc_med c=IN IP4 155.161.11.194 t=0 0 m=audio 4004 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ INVITE sip:68465283@155.161.200.188:5070 SIP/2.0 Via: SIP/2.0/UDP 155.161.200.188:5080;rport;branch=z9hG4bKKX98pj6UB2crp Max-Forwards: 69 From: "Extension 1001" <sip:FreeSWITCH@155.161.200.188:5070>;tag=KKS78e6eNpUpN To: <sip:68465283@155.161.200.188:5070> Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33 CSeq: 77143746 INVITE Contact: <sip:gw+outbound1@155.161.200.188:5080;transport=udp;gw=outbound1> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 231 X-FS-Support: update_display,send_info Remote-Party-ID: "Extension 1001" <sip:1001@155.161.200.188:5070>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1434929593 1434929594 IN IP4 155.161.200.188 s=FreeSWITCH c=IN IP4 155.161.200.188 t=0 0 m=audio 30028 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 155.161.200.188:5080;received=155.161.200.188;rport=5080;branch=z9hG4bKKX98pj6UB2crp From: "Extension 1001" <sip:FreeSWITCH@155.161.200.188:5070>;tag=KKS78e6eNpUpN To: <sip:68465283@155.161.200.188:5070>;tag=sipcontrol_2457501727-2035324 Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33 CSeq: 77143746 INVITE Contact: <sip:68465283@155.161.200.188:5070> Content-Length: 0 ------------------------------------------------------------------------ SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 155.161.200.188:5080;received=155.161.200.188;rport=5080;branch=z9hG4bKKX98pj6UB2crp From: "Extension 1001" <sip:FreeSWITCH@155.161.200.188:5070>;tag=KKS78e6eNpUpN To: <sip:68465283@155.161.200.188:5070>;tag=sipcontrol_2457501727-2035324 Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33 CSeq: 77143746 INVITE Contact: <sip:68465283@155.161.200.188:5070> Retry-After: 1 Content-Length: 0 ------------------------------------------------------------------------ My config ... # vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA"/> # dialplan/default.xml <extension name="oubout_pstn"> <condition field="destination_number" expression="^\+65(.+)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/outbound1/$1"/> <action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=true"/> </condition> </extension> # sip_profiles/external.xml <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <settings> <param name="disable-transcoding" value="false"/> ... </settings> # sip_profiless/external/example.xml <gateway name="outbound1"> <param name="realm" value="155.161.200.188:5070"/> <param name="register" value="false"/> <variables> <variable name="absolute_codec_string" value="PCMA" direction="outbound"/> </variables> <param name="reg-id" value="1"/> </gateway>