[Freeswitch-users] how to transcode from g.729 to g.711a ?

[Jeff Chua]

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 at 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 at 155.161.200.188>;tag=x0m7o1ogjzCoro6dBITkX2dB5U.4DX-y
   To: <sip:+6568465283 at 155.161.200.188>
   Contact: <sip:1001 at 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 at 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 at 155.161.200.188:5070>;tag=KKS78e6eNpUpN
   To: <sip:68465283 at 155.161.200.188:5070>
   Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33
   CSeq: 77143746 INVITE
   Contact: <sip:gw+outbound1 at 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 at 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 at 155.161.200.188:5070>;tag=KKS78e6eNpUpN
   To: <sip:68465283 at 155.161.200.188:5070>;tag=sipcontrol_2457501727-2035324
   Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33
   CSeq: 77143746 INVITE
   Contact: <sip:68465283 at 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 at 155.161.200.188:5070>;tag=KKS78e6eNpUpN
   To: <sip:68465283 at 155.161.200.188:5070>;tag=sipcontrol_2457501727-2035324
   Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33
   CSeq: 77143746 INVITE
   Contact: <sip:68465283 at 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>