The problem is on your Cisco device, not FreeSWITCH.

You're sending an invite with G729:
   m=audio 25814 RTP/AVP 18 101 13
18=G729 codec  101=RFC2833 DTMF  13=Comfort Noise
This gets the 488 Not Acceptable response from the Cisco device.

So you're offering G729 to the Cisco but having it refused. This either means the Cisco device doesn't have G729 enabled or the Cisco SIP implementation doesn't like the lack of the optional a=rtpmap line. If it is the latter use <action application="set" data"verbose_sdp=true"/> before the bridge to workaround that issue.




On 12 June 2012 05:06, Samira Mh <saami_mh@ymail.com> wrote:
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:
   ------------------------------------------------------------------------
   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
   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
   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:
   ------------------------------------------------------------------------
   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:
   ------------------------------------------------------------------------
   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
   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:
   ------------------------------------------------------------------------
   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
http://www.freeswitch.org