<div style>The problem is on your Cisco device, not FreeSWITCH.</div><div style><br></div><div style>You're sending an invite with G729:</div><div style> m=audio 25814 RTP/AVP 18 101 13
</div><div style>18=G729 codec 101=RFC2833 DTMF 13=Comfort Noise</div><div style>This gets the 488 Not Acceptable response from the Cisco device.</div><div style><br></div><div style>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.</div>
<div style><br></div><div><br></div><div><br></div><br><div class="gmail_quote">On 12 June 2012 05:06, Samira Mh <span dir="ltr"><<a href="mailto:saami_mh@ymail.com" target="_blank">saami_mh@ymail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div style="font-size:12pt;font-family:times new roman,new york,times,serif"><div style="font-family:'times new roman','new york',times,serif;font-size:12pt">
thansk for your reply,</div><div style="font-family:'times new roman','new york',times,serif;font-size:12pt">it is kind of you to help me..</div><div style="font-family:'times new roman','new york',times,serif;font-size:12pt">
please let me paste myconfigurations files here;</div><div style="font-family:'times new roman','new york',times,serif;font-size:12pt">1-the configuration file /usr/local/freeswitch/conf/dialplan/default/001_luacallduration.xml is like this:</div>
<div style="font-family:'times new roman','new york',times,serif;font-size:12pt"><br></div><div><div><include></div><div> <extension name="mainmenuvodsl"></div><div><br></div><div>
<condition field="destination_number" expression="^(00|\+)?(\d{5}.*)$" break="never"></div><div> <action application="odbc_query" data="select cash as cashvalue from accounts where contractid like '${nibble_account}';"/></div>
<div> <action application="log" data="INFO The value of cashvalue is ${cashvalue}" /><br></div><div> <action application="lua" data="checkcash.lua ${cashvalue}" /><br>
</div><div> <action application="log" data="INFO The value of nibble_account is ${nibble_account}"/></div><div> <action application="log" data="INFO The value of nibble_rate [before] is ${nibble_rate}"/></div>
<div>
<!-- RateList Context --></div><div> <action application="lua" data="checkzeroplus.lua ${destination_number:0:2} ${destination_number:0:1}" /></div><div> <action application="execute_extension" data="${destination_number} XML ratelist"/></div>
<div> <action application="log" data="INFO The value of nibble_rate [after] is ${nibble_rate}"/></div><div> <!-- Check Nibble_rate --></div><div> <action application="lua" data="checknibblerate.lua ${nibble_rate}" /></div>
<div> <action application="set"
data="divvalue=${expr(floor((${cashvalue}/${nibble_rate}))}" /></div><div> <action application="set" data="modvalue=${expr(mod(${cashvalue},${nibble_rate}))}" /></div>
<div> <action application="lua" data="checktime.lua ${divvalue} ${modvalue}" /></div><div> <!-- Check ZeroZero , Plus --></div><div> <action application="lua" data="checkzeroplus.lua ${destination_number:0:2} ${destination_number:0:1}" /> </div>
<div> <!-- Making Calls --></div><div> <action application="odbc_query" data="select callerid as effective_caller_id_number from
accounts where contractid like '${nibble_account}';"/></div><div> <action application="log" data="INFO callerid for Outbound calls ${effective_caller_id_number}"/></div>
<div> <!-- <action application="set" data="ignore_early_media=true"/></div><div> <action application="answer"/> --></div><div> <action application="enable_heartbeat"/></div>
<div> </div><div><!-- <param name="disable-transcoding" value="true"/> --></div><div> <!-- <action application="export" data="nolocal:absolute_codec_string=G729,PCMU"/> --></div>
<div><!-- <action application="set"
data="bridge_early_media=true"/> --></div><div> <!-- <action application="set" data="proxy_media=true"/> --></div><div> <action application="bridge" data="sofia/gateway/cisco/140112${destination_number}"/><br>
</div><div> <!-- <action application="bridge" data="sofia/gateway/mainasterisk/${destination_number}"/> --></div><div><!-- <action application="bridge" data="sofia/gateway/test/${destination_number}"/> --></div>
<div> </condition></div><div><br></div><div> </extension></div><div></include></div><div style="font-family:'times new roman','new york',times,serif;font-size:12pt"><br></div></div>
<div>2-yes, i have enabled "inbound-late-negotiation" in the
(/usr/local/freeswitch/conf/sip_profiles/internal.xml) as follow:</div><div class="im"><div> <param name="inbound-late-negotiation" value="true"/><br></div><div style="font-family:'times new roman','new york',times,serif;font-size:12pt">
<br></div><div style="font-family:'times new roman','new york',times,serif;font-size:12pt"><br></div></div><div style="font-family:'times new roman','new york',times,serif;font-size:12pt">
3-the issue of sofia status:</div>
<div> external::cisco gateway <a href="mailto:sip%3Aregister%3Afalse@85.15.0.154" target="_blank">sip:register:false@85.15.0.154</a> NOREG<br></div><div><br></div><div><br></div><div>4-also , the configuration file for codecs are as follow</div>
<div>:/usr/local/freeswitch/conf/vars.xml</div><div><br></div><div><div><X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G729,PCMU,PCMA,G7221@32000h,G7221@16000h,G722,GSM"/> </div><div><br></div></div><div><div><X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/></div>
<div><br></div><div>5- the mod_g729 was loaded </div><div><br></div><div>6-i have enabled the siptrace:</div><div> sofia profile external siptrace on:<br></div><div><span style="background-color:rgb(255,255,0)">the siptrace outpout as follow:</span></div>
<div><br></div><div><div><div>send 1042 bytes to udp/[85.15.0.154]:5060 at 03:53:07.448136:</div><div> ------------------------------------------------------------------------</div><div> INVITE <a href="mailto:sip%3A140112971507247227@85.15.0.154" target="_blank">sip:140112971507247227@85.15.0.154</a> SIP/2.0</div>
<div> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS</div><div> Max-Forwards: 69</div><div> From: "1000" <<a href="mailto:sip%3Aregister%3Afalse@85.15.0.154" target="_blank">sip:register:false@85.15.0.154</a>>;tag=62QN1XNSF6rvD</div>
<div>
To: <<a href="mailto:sip%3A140112971507247227@85.15.0.154" target="_blank">sip:140112971507247227@85.15.0.154</a>></div><div> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9</div><div> CSeq: 29400529 INVITE</div>
<div> Contact: <sip:gw+cisco@192.168.10.70:5080;transport=udp;gw=cisco></div><div> User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2</div><div> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY</div>
<div> Supported: timer, precondition, path, replaces</div><div> Allow-Events: talk, hold, refer</div><div> Content-Type: application/sdp</div><div> Content-Disposition: session</div><div> Content-Length: 234</div>
<div> X-FS-Support: update_display,send_info</div><div> Remote-Party-ID: "1000" <<a href="mailto:sip%3A1000@85.15.0.154" target="_blank">sip:1000@85.15.0.154</a>>;party=calling;screen=yes;privacy=off</div>
<div><br></div><div> v=0</div><div> o=FreeSWITCH
1339446571 1339446572 IN IP4 192.168.10.70</div><div> s=FreeSWITCH</div><div> c=IN IP4 192.168.10.70</div><div> t=0 0</div><div> m=audio 26616 RTP/AVP 9 0 8 18 3 101 13</div><div> a=fmtp:18 annexb=yes</div><div>
a=rtpmap:101 telephone-event/8000</div><div> a=fmtp:101 0-16</div><div> a=ptime:20</div><div> ------------------------------------------------------------------------</div><div>recv 410 bytes from udp/[85.15.0.154]:5060 at 03:53:07.463921:</div>
<div> ------------------------------------------------------------------------</div><div> SIP/2.0 100 Trying</div><div> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS</div><div> From: "1000" <<a href="mailto:sip%3Aregister%3Afalse@85.15.0.154" target="_blank">sip:register:false@85.15.0.154</a>>;tag=62QN1XNSF6rvD</div>
<div> To:
<<a href="mailto:sip%3A140112971507247227@85.15.0.154" target="_blank">sip:140112971507247227@85.15.0.154</a>>;tag=45785134-1BDE</div><div> Date: Tue, 12 Jun 2012 03:53:15 GMT</div><div> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9</div>
<div> Server: Cisco-SIPGateway/IOS-12.x</div><div> CSeq: 29400529 INVITE</div><div> Allow-Events: telephone-event</div><div> Content-Length: 0</div><div><br></div><div> ------------------------------------------------------------------------</div>
<div>recv 927 bytes from udp/[85.15.0.154]:5060 at 03:53:11.004804:</div><div> ------------------------------------------------------------------------</div><div> SIP/2.0 183 Session Progress</div><div> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS</div>
<div> From: "1000" <<a href="mailto:sip%3Aregister%3Afalse@85.15.0.154" target="_blank">sip:register:false@85.15.0.154</a>>;tag=62QN1XNSF6rvD</div><div> To:
<<a href="mailto:sip%3A140112971507247227@85.15.0.154" target="_blank">sip:140112971507247227@85.15.0.154</a>>;tag=45785134-1BDE</div><div> Date: Tue, 12 Jun 2012 03:53:15 GMT</div><div> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9</div>
<div> Server: Cisco-SIPGateway/IOS-12.x</div><div> CSeq: 29400529 INVITE</div><div> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER</div><div> Allow-Events: telephone-event</div>
<div> Contact: <<a href="http://sip:140112971507247227@85.15.0.154:5060" target="_blank">sip:140112971507247227@85.15.0.154:5060</a>></div><div> Content-Disposition: session;handling=required</div><div> Content-Type: application/sdp</div>
<div> Content-Length: 268</div><div><br></div><div> v=0</div><div> o=CiscoSystemsSIP-GW-UserAgent 7252 5649 IN IP4 85.15.0.154</div><div> s=SIP Call</div><div> c=IN IP4 85.15.0.154</div><div> t=0 0</div><div>
m=audio 18218 RTP/AVP 0 13 101</div><div> c=IN IP4 85.15.0.154</div><div> a=rtpmap:0 PCMU/8000</div><div> a=rtpmap:13 CN/8000</div><div> a=rtpmap:101 telephone-event/8000</div><div> a=fmtp:101 0-15</div><div>
a=ptime:20</div><div> ------------------------------------------------------------------------</div><div>recv 425 bytes from udp/[85.15.0.154]:5060 at 03:53:11.005144:</div><div> ------------------------------------------------------------------------</div>
<div> SIP/2.0 500 Internal Server Error</div><div> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS</div><div> From: "1000" <<a href="mailto:sip%3Aregister%3Afalse@85.15.0.154" target="_blank">sip:register:false@85.15.0.154</a>>;tag=62QN1XNSF6rvD</div>
<div> To: <<a href="mailto:sip%3A140112971507247227@85.15.0.154" target="_blank">sip:140112971507247227@85.15.0.154</a>>;tag=45785134-1BDE</div><div> Date: Tue, 12 Jun 2012 03:53:15
GMT</div><div> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9</div><div> Server: Cisco-SIPGateway/IOS-12.x</div><div> CSeq: 29400529 INVITE</div><div> Allow-Events: telephone-event</div><div> Content-Length: 0</div>
<div><br></div><div> ------------------------------------------------------------------------</div><div>send 350 bytes to udp/[85.15.0.154]:5060 at 03:53:11.005333:</div><div> ------------------------------------------------------------------------</div>
<div> ACK <a href="mailto:sip%3A140112971507247227@85.15.0.154" target="_blank">sip:140112971507247227@85.15.0.154</a> SIP/2.0</div><div> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS</div><div>
Max-Forwards: 69</div><div> From: "1000" <<a href="mailto:sip%3Aregister%3Afalse@85.15.0.154" target="_blank">sip:register:false@85.15.0.154</a>>;tag=62QN1XNSF6rvD</div><div> To: <<a href="mailto:sip%3A140112971507247227@85.15.0.154" target="_blank">sip:140112971507247227@85.15.0.154</a>>;tag=45785134-1BDE</div>
<div> Call-ID:
f61cf067-2ee4-1230-0cad-0050569414f9</div><div> CSeq: 29400529 ACK</div><div> Content-Length: 0</div><div><br></div></div></div><div><br></div><div>------------------------------------------------------------------------------------------------------------------</div>
<div><span style="background-color:rgb(255,255,0)">when change the configuration file the below:</span></div><div><div><X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729"/></div><div><X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/> <br>
</div></div><div><br></div><div>the siptrace is like this:</div><div><br></div><div><div>send 1034 bytes to udp/[85.15.0.154]:5060 at 04:01:16.202342:</div><div> ------------------------------------------------------------------------</div>
<div> INVITE <a href="mailto:sip%3A140112971507247227@85.15.0.154" target="_blank">sip:140112971507247227@85.15.0.154</a> SIP/2.0</div><div> Via: SIP/2.0/UDP
192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK</div><div> Max-Forwards: 69</div><div> From: "1000" <<a href="mailto:sip%3Aregister%3Afalse@85.15.0.154" target="_blank">sip:register:false@85.15.0.154</a>>;tag=Na0S1Q9mNmS1r</div>
<div> To: <<a href="mailto:sip%3A140112971507247227@85.15.0.154" target="_blank">sip:140112971507247227@85.15.0.154</a>></div><div> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9</div><div> CSeq: 29400774 INVITE</div>
<div> Contact: <sip:gw+cisco@192.168.10.70:5080;transport=udp;gw=cisco></div><div> User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2</div><div> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY</div>
<div> Supported: timer, precondition, path, replaces</div><div> Allow-Events: talk, hold, refer</div><div> Content-Type: application/sdp</div><div> Content-Disposition: session</div><div> Content-Length: 226</div>
<div> X-FS-Support:
update_display,send_info</div><div> Remote-Party-ID: "1000" <<a href="mailto:sip%3A1000@85.15.0.154" target="_blank">sip:1000@85.15.0.154</a>>;party=calling;screen=yes;privacy=off</div><div><br></div><div>
v=0</div><div> o=FreeSWITCH 1339447862 1339447863 IN IP4 192.168.10.70</div><div> s=FreeSWITCH</div><div> c=IN IP4 192.168.10.70</div><div> t=0 0</div><div> m=audio 25814 RTP/AVP 18 101 13</div><div> a=fmtp:18 annexb=yes</div>
<div> a=rtpmap:101 telephone-event/8000</div><div> a=fmtp:101 0-16</div><div> a=ptime:20</div><div> ------------------------------------------------------------------------</div><div>recv 423 bytes from udp/[85.15.0.154]:5060 at 04:01:16.208118:</div>
<div> ------------------------------------------------------------------------</div><div> SIP/2.0 488 Not Acceptable Media</div><div> Via: SIP/2.0/UDP
192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK</div><div> From: "1000" <<a href="mailto:sip%3Aregister%3Afalse@85.15.0.154" target="_blank">sip:register:false@85.15.0.154</a>>;tag=Na0S1Q9mNmS1r</div>
<div> To: <<a href="mailto:sip%3A140112971507247227@85.15.0.154" target="_blank">sip:140112971507247227@85.15.0.154</a>>;tag=457FC664-6A6</div><div> Date: Tue, 12 Jun 2012 04:01:24 GMT</div><div> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9</div>
<div> Server: Cisco-SIPGateway/IOS-12.x</div><div> CSeq: 29400774 INVITE</div><div> Allow-Events: telephone-event</div><div> Content-Length: 0</div><div><br></div><div> ------------------------------------------------------------------------</div>
<div>send 349 bytes to udp/[85.15.0.154]:5060 at 04:01:16.208201:</div><div> ------------------------------------------------------------------------</div><div> ACK <a href="mailto:sip%3A140112971507247227@85.15.0.154" target="_blank">sip:140112971507247227@85.15.0.154</a> SIP/2.0</div>
<div> Via: SIP/2.0/UDP
192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK</div><div> Max-Forwards: 69</div><div> From: "1000" <<a href="mailto:sip%3Aregister%3Afalse@85.15.0.154" target="_blank">sip:register:false@85.15.0.154</a>>;tag=Na0S1Q9mNmS1r</div>
<div> To: <<a href="mailto:sip%3A140112971507247227@85.15.0.154" target="_blank">sip:140112971507247227@85.15.0.154</a>>;tag=457FC664-6A6</div><div> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9</div><div> CSeq: 29400774 ACK</div>
<div> Content-Length: 0</div><div><br></div></div><div><br></div><div><br></div><div>plz help,thanks so much</div><div><br></div><div><br></div></div> <div style="font-size:12pt;font-family:'times new roman','new york',times,serif">
<div style="font-size:12pt;font-family:'times new roman','new york',times,serif"> <div dir="ltr"> <font face="Arial"> <hr size="1"> <b><span style="font-weight:bold">From:</span></b> Paul Cupis <<a href="mailto:paul@cupis.co.uk" target="_blank">paul@cupis.co.uk</a>><div class="im">
<br> <b><span style="font-weight:bold">To:</span></b> FreeSWITCH Users Help
<<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>> <br> </div><b><span style="font-weight:bold">Sent:</span></b> Monday, June 11, 2012 10:51 PM<div class="im">
<br> <b><span style="font-weight:bold">Subject:</span></b> Re: [Freeswitch-users] how to use codec g729 on freeswitch ?<br> </div></font> </div><div><div class="h5"> <br>
On 11/06/12 17:54, Samira Mh wrote:<br>> i want to bridge call using my VOIPgateway so that making calls to<br>> another countries..<br>> but the carrier only support G729 codec and the FS send G722 (set in<br>> vars.xml) to myVoipGateway that is set as an gateway in<br>
> /usr/local/freeswitch/sip-profile/external/<br>> and when FS send media to Gateway(using bridge application) the error<br>> occure:unacceptable media,then check VOIPGW and find out the only codec<br>> that<br>
> can be pass through VOIPgw is G729, but FS only send G711,G722,... not G729<br><br>Can you provide a SIP or FreeSWITCH trace of a call, please?<br><br>Do you have the following enabled in your SIP profile?<br><br> <param name="inbound-late-negotiation" value="true"/><br>
<br>Do you have mod_g729 loaded and codec G729 enabled in your
vars.xml?<br><br>Regards,<br><br>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br>
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<br><br> </div></div></div> </div> </div></div><br>_________________________________________________________________________<br>
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<br></blockquote></div><br>