sorry i forgot to paste codec on internal and external:

freeswitch@internal> sofia status profile internal
=================================================================================================
Name                    internal
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_internal
Pres Hosts              192.168.10.70,192.168.10.70
Dialplan                XML
Context                 public
Challenge Realm         auto_from
RTP-IP                  192.168.10.70
SIP-IP                  192.168.10.70
URL                     sip:mod_sofia@192.168.10.70:5060
BIND-URL                sip:mod_sofia@192.168.10.70:5060
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               G729
CODECS OUT              G729
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                true
PROXY-MEDIA             false
ZRTP-PASSTHRU           false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               0
FAILED-CALLS-OUT        0
REGISTRATIONS           3

---------------------------------------------------------------------------------
freeswitch@internal> sofia status profile external
=================================================================================================
Name                    external
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_external
Pres Hosts
Dialplan                XML
Context                 public
Challenge Realm         auto_to
RTP-IP                  192.168.10.70
Ext-RTP-IP              192.168.10.70
SIP-IP                  192.168.10.70
Ext-SIP-IP              192.168.10.70
URL                     sip:mod_sofia@192.168.10.70:5080
BIND-URL                sip:mod_sofia@192.168.10.70:5080;maddr=192.168.10.70
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               G729
CODECS OUT              PCMU
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
ZRTP-PASSTHRU           false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               0
FAILED-CALLS-OUT        0
REGISTRATIONS           0


From: Samira Mh <saami_mh@ymail.com>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Sent: Tuesday, June 12, 2012 8:36 AM
Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?

thansk for your reply,
it is kind of you to help me..
please let me paste myconfigurations files here;
1-the configuration  file /usr/local/freeswitch/conf/dialplan/default/001_luacallduration.xml  is like this:

<include>
  <extension name="mainmenuvodsl">

        <condition field="destination_number" expression="^(00|\+)?(\d{5}.*)$" break="never">
                   <action application="odbc_query" data="select cash as cashvalue from accounts where contractid like '${nibble_account}';"/>
                <action application="log" data="INFO The value of cashvalue is ${cashvalue}" />
                 <action application="lua" data="checkcash.lua ${cashvalue}" />
                <action application="log" data="INFO The value of nibble_account is  ${nibble_account}"/>
                <action application="log" data="INFO The value of nibble_rate [before] is  ${nibble_rate}"/>
                <!-- RateList Context -->
                <action application="lua" data="checkzeroplus.lua ${destination_number:0:2} ${destination_number:0:1}" />
                <action application="execute_extension" data="${destination_number} XML ratelist"/>
                <action application="log" data="INFO The value of nibble_rate [after] is ${nibble_rate}"/>
                 <!-- Check Nibble_rate -->
                <action application="lua" data="checknibblerate.lua ${nibble_rate}" />
                <action application="set" data="divvalue=${expr(floor((${cashvalue}/${nibble_rate}))}" />
                <action application="set" data="modvalue=${expr(mod(${cashvalue},${nibble_rate}))}" />
                <action application="lua" data="checktime.lua ${divvalue} ${modvalue}" />
                    <!--  Check ZeroZero , Plus  -->
                <action application="lua" data="checkzeroplus.lua ${destination_number:0:2} ${destination_number:0:1}" /> 
                <!-- Making Calls   -->
                <action application="odbc_query" data="select callerid  as effective_caller_id_number from accounts where contractid like '${nibble_account}';"/>
                <action application="log" data="INFO  callerid for Outbound calls ${effective_caller_id_number}"/>
                <!-- <action application="set" data="ignore_early_media=true"/>
                <action application="answer"/>  -->
                <action application="enable_heartbeat"/>
                 
<!-- <param name="disable-transcoding" value="true"/> -->
        <!--    <action application="export" data="nolocal:absolute_codec_string=G729,PCMU"/> -->
<!--  <action application="set" data="bridge_early_media=true"/>  -->
        <!-- <action application="set" data="proxy_media=true"/> -->
                <action application="bridge" data="sofia/gateway/cisco/140112${destination_number}"/>
                <!-- <action application="bridge" data="sofia/gateway/mainasterisk/${destination_number}"/>  -->
<!--  <action application="bridge" data="sofia/gateway/test/${destination_number}"/>  -->
        </condition>

 </extension>
</include>

2-yes, i have enabled  "inbound-late-negotiation" in the (/usr/local/freeswitch/conf/sip_profiles/internal.xml) as follow:
 <param name="inbound-late-negotiation" value="true"/>


3-the issue of sofia status:
 external::cisco       gateway             sip:register:false@85.15.0.154      NOREG


4-also , the configuration file for codecs are as follow
:/usr/local/freeswitch/conf/vars.xml

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA,G7221@32000h,G7221@16000h,G722,GSM"/> 

<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>

5- the mod_g729 was loaded 

6-i have enabled the siptrace:
 sofia profile external siptrace on:
the siptrace outpout as follow:

send 1042 bytes to udp/[85.15.0.154]:5060 at 03:53:07.448136:
   ------------------------------------------------------------------------
   INVITE sip:140112971507247227@85.15.0.154 SIP/2.0
   Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
   Max-Forwards: 69
   From: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvD
   To: <sip:140112971507247227@85.15.0.154>
   Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
   CSeq: 29400529 INVITE
   Contact: <sip:gw+cisco@192.168.10.70:5080;transport=udp;gw=cisco>
   User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 234
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "1000" <sip:1000@85.15.0.154>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1339446571 1339446572 IN IP4 192.168.10.70
   s=FreeSWITCH
   c=IN IP4 192.168.10.70
   t=0 0
   m=audio 26616 RTP/AVP 9 0 8 18 3 101 13
   a=fmtp:18 annexb=yes
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 410 bytes from udp/[85.15.0.154]:5060 at 03:53:07.463921:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
   From: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvD
   To: <sip:140112971507247227@85.15.0.154>;tag=45785134-1BDE
   Date: Tue, 12 Jun 2012 03:53:15 GMT
   Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
   Server: Cisco-SIPGateway/IOS-12.x
   CSeq: 29400529 INVITE
   Allow-Events: telephone-event
   Content-Length: 0

   ------------------------------------------------------------------------
recv 927 bytes from udp/[85.15.0.154]:5060 at 03:53:11.004804:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
   From: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvD
   To: <sip:140112971507247227@85.15.0.154>;tag=45785134-1BDE
   Date: Tue, 12 Jun 2012 03:53:15 GMT
   Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
   Server: Cisco-SIPGateway/IOS-12.x
   CSeq: 29400529 INVITE
   Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
   Allow-Events: telephone-event
   Contact: <sip:140112971507247227@85.15.0.154:5060>
   Content-Disposition: session;handling=required
   Content-Type: application/sdp
   Content-Length: 268

   v=0
   o=CiscoSystemsSIP-GW-UserAgent 7252 5649 IN IP4 85.15.0.154
   s=SIP Call
   c=IN IP4 85.15.0.154
   t=0 0
   m=audio 18218 RTP/AVP 0 13 101
   c=IN IP4 85.15.0.154
   a=rtpmap:0 PCMU/8000
   a=rtpmap:13 CN/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:20
   ------------------------------------------------------------------------
recv 425 bytes from udp/[85.15.0.154]:5060 at 03:53:11.005144:
   ------------------------------------------------------------------------
   SIP/2.0 500 Internal Server Error
   Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
   From: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvD
   To: <sip:140112971507247227@85.15.0.154>;tag=45785134-1BDE
   Date: Tue, 12 Jun 2012 03:53:15 GMT
   Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
   Server: Cisco-SIPGateway/IOS-12.x
   CSeq: 29400529 INVITE
   Allow-Events: telephone-event
   Content-Length: 0

   ------------------------------------------------------------------------
send 350 bytes to udp/[85.15.0.154]:5060 at 03:53:11.005333:
   ------------------------------------------------------------------------
   ACK sip:140112971507247227@85.15.0.154 SIP/2.0
   Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
   Max-Forwards: 69
   From: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvD
   To: <sip:140112971507247227@85.15.0.154>;tag=45785134-1BDE
   Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
   CSeq: 29400529 ACK
   Content-Length: 0


------------------------------------------------------------------------------------------------------------------
when change the configuration file the below:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/> 

the siptrace is like this:

send 1034 bytes to udp/[85.15.0.154]:5060 at 04:01:16.202342:
   ------------------------------------------------------------------------
   INVITE sip:140112971507247227@85.15.0.154 SIP/2.0
   Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
   Max-Forwards: 69
   From: "1000" <sip:register:false@85.15.0.154>;tag=Na0S1Q9mNmS1r
   To: <sip:140112971507247227@85.15.0.154>
   Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
   CSeq: 29400774 INVITE
   Contact: <sip:gw+cisco@192.168.10.70:5080;transport=udp;gw=cisco>
   User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 226
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "1000" <sip:1000@85.15.0.154>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1339447862 1339447863 IN IP4 192.168.10.70
   s=FreeSWITCH
   c=IN IP4 192.168.10.70
   t=0 0
   m=audio 25814 RTP/AVP 18 101 13
   a=fmtp:18 annexb=yes
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
recv 423 bytes from udp/[85.15.0.154]:5060 at 04:01:16.208118:
   ------------------------------------------------------------------------
   SIP/2.0 488 Not Acceptable Media
   Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
   From: "1000" <sip:register:false@85.15.0.154>;tag=Na0S1Q9mNmS1r
   To: <sip:140112971507247227@85.15.0.154>;tag=457FC664-6A6
   Date: Tue, 12 Jun 2012 04:01:24 GMT
   Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
   Server: Cisco-SIPGateway/IOS-12.x
   CSeq: 29400774 INVITE
   Allow-Events: telephone-event
   Content-Length: 0

   ------------------------------------------------------------------------
send 349 bytes to udp/[85.15.0.154]:5060 at 04:01:16.208201:
   ------------------------------------------------------------------------
   ACK sip:140112971507247227@85.15.0.154 SIP/2.0
   Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
   Max-Forwards: 69
   From: "1000" <sip:register:false@85.15.0.154>;tag=Na0S1Q9mNmS1r
   To: <sip:140112971507247227@85.15.0.154>;tag=457FC664-6A6
   Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
   CSeq: 29400774 ACK
   Content-Length: 0



plz help,thanks so much


From: Paul Cupis <paul@cupis.co.uk>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Sent: Monday, June 11, 2012 10:51 PM
Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?

On 11/06/12 17:54, Samira Mh wrote:
> i want to bridge call using my VOIPgateway so that making calls to
> another countries..
> but the carrier only support G729 codec and the FS send G722 (set in
> vars.xml) to myVoipGateway that is set as an gateway in
> /usr/local/freeswitch/sip-profile/external/
> and when FS send media to Gateway(using bridge application) the error
> occure:unacceptable media,then check VOIPGW and find out the only codec
> that
> can be pass through VOIPgw is G729, but FS only send G711,G722,... not G729

Can you provide a SIP or FreeSWITCH trace of a call, please?

Do you have the following enabled in your SIP profile?

  <param name="inbound-late-negotiation" value="true"/>

Do you have mod_g729 loaded and codec G729 enabled in your vars.xml?

Regards,

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
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Join Us At ClueCon - Aug 7-9, 2012

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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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