Also in your first example you say you set the codec preferences in vars.conf.xml to G729, but only PCMU was offered. CODECS OUT only lists PCMU on your external profile. That means you've either not reloaded mod_sofia since editing the file, or you have PCMU listed in the sip_profiles/external.xml<div>
<br></div><div>Steve</div><div><br><br><div class="gmail_quote">On 12 June 2012 05:13, 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"><span>sorry i forgot to paste codec on internal and external:</span></div>
<div style="font-family:'times new roman','new york',times,serif;font-size:12pt"><span><br></span></div><div><div>freeswitch@internal> sofia status profile internal</div><div>=================================================================================================</div>
<div>Name internal</div><div>Domain Name N/A</div><div>Auto-NAT false</div><div>DBName sofia_reg_internal</div><div>Pres Hosts
192.168.10.70,192.168.10.70</div><div>Dialplan XML</div><div>Context public</div><div>Challenge Realm auto_from</div><div>RTP-IP 192.168.10.70</div><div>SIP-IP 192.168.10.70</div>
<div>URL <a href="http://sip:mod_sofia@192.168.10.70:5060" target="_blank">sip:mod_sofia@192.168.10.70:5060</a></div><div>BIND-URL <a href="http://sip:mod_sofia@192.168.10.70:5060" target="_blank">sip:mod_sofia@192.168.10.70:5060</a></div>
<div>HOLD-MUSIC local_stream://moh</div><div>OUTBOUND-PROXY N/A</div><div><span style="background-color:rgb(255,255,0)">CODECS IN
G729</span></div><div><span style="background-color:rgb(255,255,0)">CODECS OUT G729</span></div><div>TEL-EVENT 101</div><div>DTMF-MODE rfc2833</div><div>CNG 13</div>
<div>SESSION-TO 0</div><div>MAX-DIALOG 0</div><div>NOMEDIA false</div><div>LATE-NEG true</div><div>PROXY-MEDIA false</div><div>ZRTP-PASSTHRU false</div>
<div>AGGRESSIVENAT false</div><div>STUN-ENABLED
true</div><div>STUN-AUTO-DISABLE false</div><div>CALLS-IN 0</div><div>FAILED-CALLS-IN 0</div><div>CALLS-OUT 0</div><div>FAILED-CALLS-OUT 0</div><div>REGISTRATIONS 3</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">---------------------------------------------------------------------------------</div>
<div><div>freeswitch@internal> sofia status profile external</div><div>=================================================================================================</div><div>Name external</div>
<div>
Domain Name
N/A</div><div>Auto-NAT false</div><div>DBName sofia_reg_external</div><div>Pres Hosts</div><div>Dialplan XML</div><div>Context public</div><div>Challenge Realm auto_to</div>
<div>RTP-IP 192.168.10.70</div><div>Ext-RTP-IP 192.168.10.70</div><div>SIP-IP 192.168.10.70</div><div>Ext-SIP-IP 192.168.10.70</div><div>URL <a href="http://sip:mod_sofia@192.168.10.70:5080" target="_blank">sip:mod_sofia@192.168.10.70:5080</a></div>
<div>BIND-URL
sip:mod_sofia@192.168.10.70:5080;maddr=192.168.10.70</div><div>HOLD-MUSIC local_stream://moh</div><div>OUTBOUND-PROXY N/A</div><div><span style="background-color:rgb(255,255,0)">CODECS IN G729</span></div>
<div><span style="background-color:rgb(255,255,0)">CODECS OUT PCMU</span></div><div>TEL-EVENT 101</div><div>DTMF-MODE rfc2833</div><div>CNG 13</div><div>SESSION-TO 0</div>
<div>MAX-DIALOG 0</div><div>NOMEDIA
false</div><div>LATE-NEG false</div><div>PROXY-MEDIA false</div><div>ZRTP-PASSTHRU false</div><div>AGGRESSIVENAT false</div><div>STUN-ENABLED true</div><div>STUN-AUTO-DISABLE false</div>
<div>CALLS-IN 0</div><div>FAILED-CALLS-IN 0</div><div>CALLS-OUT 0</div><div>FAILED-CALLS-OUT 0</div><div>REGISTRATIONS 0</div><div style="font-family:'times new roman','new york',times,serif;font-size:12pt">
<br></div></div></div><div style="font-family:'times new roman','new york',times,serif;font-size:12pt"><br></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> Samira Mh <<a href="mailto:saami_mh@ymail.com" target="_blank">saami_mh@ymail.com</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> Tuesday, June 12, 2012 8:36 AM<div>
<div class="h5"><br> <b><span style="font-weight:bold">Subject:</span></b> Re: [Freeswitch-users] how to use codec g729 on freeswitch ?<br> </div></div></font> </div><div><div class="h5"> <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,serif">thansk for your reply,</div><div style="font-size:12pt;font-family:times,serif">
it is kind of you to help me..</div><div style="font-size:12pt;font-family:times,serif">please let me paste myconfigurations files here;</div><div style="font-size:12pt;font-family:times,serif">1-the configuration file /usr/local/freeswitch/conf/dialplan/default/001_luacallduration.xml is like this:</div>
<div style="font-size:12pt;font-family:times,serif"><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-size:12pt;font-family:times,serif"><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> <param name="inbound-late-negotiation" value="true"/><br></div><div style="font-size:12pt;font-family:times,serif"><br>
</div><div style="font-size:12pt;font-family:times,serif"><br></div><div style="font-size:12pt;font-family:times,serif">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,serif"> <div style="font-size:12pt;font-family: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>><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> <b><span style="font-weight:bold">Sent:</span></b> Monday, June 11, 2012 10:51 PM<br> <b><span style="font-weight:bold">Subject:</span></b> Re: [Freeswitch-users] how to use codec g729 on freeswitch ?<br>
</font> </div> <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 rel="nofollow" href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br>
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