<html><body><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div><span>bu i have set the codec in dialpeer of VOIP gateway<br></span></div><div><br></div> <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <div dir="ltr"> <font face="Arial" size="2"> <hr size="1"> <b><span style="font-weight:bold;">From:</span></b> Kristian Kielhofner <kris@kriskinc.com><br> <b><span style="font-weight: bold;">To:</span></b> FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <br> <b><span style="font-weight: bold;">Sent:</span></b> Wednesday, June 13, 2012 1:40 AM<br> <b><span style="font-weight: bold;">Subject:</span></b> Re: [Freeswitch-users] how to use codec g729 on freeswitch ?<br> </font> </div> <br>Your Cisco gateway doesn't support G729. Check your
codec-list<br>configuration in IOS.<br><br>On Tue, Jun 12, 2012 at 12:06 AM, Samira Mh <<a ymailto="mailto:saami_mh@ymail.com" href="mailto:saami_mh@ymail.com">saami_mh@ymail.com</a>> wrote:<br>> thansk for your reply,<br>> it is kind of you to help me..<br>> please let me paste myconfigurations files here;<br>> 1-the configuration file<br>> /usr/local/freeswitch/conf/dialplan/default/001_luacallduration.xml is like<br>> this:<br>><br>> <include><br>> <extension name="mainmenuvodsl"><br>><br>> <condition field="destination_number"<br>> expression="^(00|\+)?(\d{5}.*)$" break="never"><br>> <action application="odbc_query" data="select cash as<br>> cashvalue from accounts where contractid like '${nibble_account}';"/><br>>
<action application="log" data="INFO The value of cashvalue<br>> is ${cashvalue}" /><br>> <action application="lua" data="checkcash.lua ${cashvalue}"<br>> /><br>> <action application="log" data="INFO The value of<br>> nibble_account is ${nibble_account}"/><br>> <action application="log" data="INFO The value of<br>> nibble_rate [before] is ${nibble_rate}"/><br>> <!-- RateList Context --><br>> <action application="lua" data="checkzeroplus.lua<br>> ${destination_number:0:2} ${destination_number:0:1}" /><br>> <action
application="execute_extension"<br>> data="${destination_number} XML ratelist"/><br>> <action application="log" data="INFO The value of<br>> nibble_rate [after] is ${nibble_rate}"/><br>> <!-- Check Nibble_rate --><br>> <action application="lua" data="checknibblerate.lua<br>> ${nibble_rate}" /><br>> <action application="set"<br>> data="divvalue=${expr(floor((${cashvalue}/${nibble_rate}))}" /><br>> <action application="set"<br>> data="modvalue=${expr(mod(${cashvalue},${nibble_rate}))}" /><br>> <action application="lua" data="checktime.lua
${divvalue}<br>> ${modvalue}" /><br>> <!-- Check ZeroZero , Plus --><br>> <action application="lua" data="checkzeroplus.lua<br>> ${destination_number:0:2} ${destination_number:0:1}" /><br>> <!-- Making Calls --><br>> <action application="odbc_query" data="select callerid as<br>> effective_caller_id_number from accounts where contractid like<br>> '${nibble_account}';"/><br>> <action application="log" data="INFO callerid for Outbound<br>> calls ${effective_caller_id_number}"/><br>> <!-- <action
application="set"<br>> data="ignore_early_media=true"/><br>> <action application="answer"/> --><br>> <action application="enable_heartbeat"/><br>><br>> <!-- <param name="disable-transcoding" value="true"/> --><br>> <!-- <action application="export"<br>> data="nolocal:absolute_codec_string=G729,PCMU"/> --><br>> <!-- <action application="set" data="bridge_early_media=true"/> --><br>> <!-- <action application="set" data="proxy_media=true"/> --><br>> <action application="bridge"<br>> data="sofia/gateway/cisco/140112${destination_number}"/><br>>
<!-- <action application="bridge"<br>> data="sofia/gateway/mainasterisk/${destination_number}"/> --><br>> <!-- <action application="bridge"<br>> data="sofia/gateway/test/${destination_number}"/> --><br>> </condition><br>><br>> </extension><br>> </include><br>><br>> 2-yes, i have enabled "inbound-late-negotiation" in the<br>> (/usr/local/freeswitch/conf/sip_profiles/internal.xml) as follow:<br>> <param name="inbound-late-negotiation" value="true"/><br>><br>><br>> 3-the issue of sofia status:<br>> external::cisco gateway sip:register:false@85.15.0.154<br>> NOREG<br>><br>><br>> 4-also , the configuration file for codecs are as follow<br>> :/usr/local/freeswitch/conf/vars.xml<br>><br>>
<X-PRE-PROCESS cmd="set"<br>> data="global_codec_prefs=G729,PCMU,PCMA,G7221@32000h,G7221@16000h,G722,GSM"/><br>><br>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/><br>><br>> 5- the mod_g729 was loaded<br>><br>> 6-i have enabled the siptrace:<br>> sofia profile external siptrace on:<br>> the siptrace outpout as follow:<br>><br>> send 1042 bytes to udp/[85.15.0.154]:5060 at 03:53:07.448136:<br>> ------------------------------------------------------------------------<br>> INVITE sip:140112971507247227@85.15.0.154 SIP/2.0<br>> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS<br>> Max-Forwards: 69<br>> From: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvD<br>> To: <sip:140112971507247227@85.15.0.154><br>> Call-ID:
f61cf067-2ee4-1230-0cad-0050569414f9<br>> CSeq: 29400529 INVITE<br>> Contact: <sip:gw+cisco@192.168.10.70:5080;transport=udp;gw=cisco><br>> User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2<br>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,<br>> REGISTER, REFER, NOTIFY<br>> Supported: timer, precondition, path, replaces<br>> Allow-Events: talk, hold, refer<br>> Content-Type: application/sdp<br>> Content-Disposition: session<br>> Content-Length: 234<br>> X-FS-Support: update_display,send_info<br>> Remote-Party-ID: "1000"<br>> <sip:1000@85.15.0.154>;party=calling;screen=yes;privacy=off<br>><br>> v=0<br>> o=FreeSWITCH 1339446571 1339446572 IN IP4 192.168.10.70<br>> s=FreeSWITCH<br>> c=IN
IP4 192.168.10.70<br>> t=0 0<br>> m=audio 26616 RTP/AVP 9 0 8 18 3 101 13<br>> a=fmtp:18 annexb=yes<br>> a=rtpmap:101 telephone-event/8000<br>> a=fmtp:101 0-16<br>> a=ptime:20<br>> ------------------------------------------------------------------------<br>> recv 410 bytes from udp/[85.15.0.154]:5060 at 03:53:07.463921:<br>> ------------------------------------------------------------------------<br>> SIP/2.0 100 Trying<br>> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS<br>> From: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvD<br>> To: <sip:140112971507247227@85.15.0.154>;tag=45785134-1BDE<br>> Date: Tue, 12 Jun 2012 03:53:15 GMT<br>> Call-ID:
f61cf067-2ee4-1230-0cad-0050569414f9<br>> Server: Cisco-SIPGateway/IOS-12.x<br>> CSeq: 29400529 INVITE<br>> Allow-Events: telephone-event<br>> Content-Length: 0<br>><br>> ------------------------------------------------------------------------<br>> recv 927 bytes from udp/[85.15.0.154]:5060 at 03:53:11.004804:<br>> ------------------------------------------------------------------------<br>> SIP/2.0 183 Session Progress<br>> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS<br>> From: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvD<br>> To: <sip:140112971507247227@85.15.0.154>;tag=45785134-1BDE<br>> Date: Tue, 12 Jun 2012 03:53:15 GMT<br>> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9<br>> Server:
Cisco-SIPGateway/IOS-12.x<br>> CSeq: 29400529 INVITE<br>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,<br>> NOTIFY, INFO, UPDATE, REGISTER<br>> Allow-Events: telephone-event<br>> Contact: <sip:140112971507247227@85.15.0.154:5060><br>> Content-Disposition: session;handling=required<br>> Content-Type: application/sdp<br>> Content-Length: 268<br>><br>> v=0<br>> o=CiscoSystemsSIP-GW-UserAgent 7252 5649 IN IP4 85.15.0.154<br>> s=SIP Call<br>> c=IN IP4 85.15.0.154<br>> t=0 0<br>> m=audio 18218 RTP/AVP 0 13 101<br>> c=IN IP4 85.15.0.154<br>> a=rtpmap:0 PCMU/8000<br>> a=rtpmap:13 CN/8000<br>> a=rtpmap:101 telephone-event/8000<br>>
a=fmtp:101 0-15<br>> a=ptime:20<br>> ------------------------------------------------------------------------<br>> recv 425 bytes from udp/[85.15.0.154]:5060 at 03:53:11.005144:<br>> ------------------------------------------------------------------------<br>> SIP/2.0 500 Internal Server Error<br>> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS<br>> From: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvD<br>> To: <sip:140112971507247227@85.15.0.154>;tag=45785134-1BDE<br>> Date: Tue, 12 Jun 2012 03:53:15 GMT<br>> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9<br>> Server: Cisco-SIPGateway/IOS-12.x<br>> CSeq: 29400529 INVITE<br>> Allow-Events: telephone-event<br>> Content-Length: 0<br>><br>>
------------------------------------------------------------------------<br>> send 350 bytes to udp/[85.15.0.154]:5060 at 03:53:11.005333:<br>> ------------------------------------------------------------------------<br>> ACK sip:140112971507247227@85.15.0.154 SIP/2.0<br>> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS<br>> Max-Forwards: 69<br>> From: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvD<br>> To: <sip:140112971507247227@85.15.0.154>;tag=45785134-1BDE<br>> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9<br>> CSeq: 29400529 ACK<br>> Content-Length: 0<br>><br>><br>> ------------------------------------------------------------------------------------------------------------------<br>> when change the configuration file the
below:<br>> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729"/><br>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/><br>><br>> the siptrace is like this:<br>><br>> send 1034 bytes to udp/[85.15.0.154]:5060 at 04:01:16.202342:<br>> ------------------------------------------------------------------------<br>> INVITE sip:140112971507247227@85.15.0.154 SIP/2.0<br>> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK<br>> Max-Forwards: 69<br>> From: "1000" <sip:register:false@85.15.0.154>;tag=Na0S1Q9mNmS1r<br>> To: <sip:140112971507247227@85.15.0.154><br>> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9<br>> CSeq: 29400774 INVITE<br>> Contact: <sip:gw+cisco@192.168.10.70:5080;transport=udp;gw=cisco><br>> User-Agent:
FreeSWITCH-mod_sofia/1.2.0-rc2<br>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,<br>> REGISTER, REFER, NOTIFY<br>> Supported: timer, precondition, path, replaces<br>> Allow-Events: talk, hold, refer<br>> Content-Type: application/sdp<br>> Content-Disposition: session<br>> Content-Length: 226<br>> X-FS-Support: update_display,send_info<br>> Remote-Party-ID: "1000"<br>> <sip:1000@85.15.0.154>;party=calling;screen=yes;privacy=off<br>><br>> v=0<br>> o=FreeSWITCH 1339447862 1339447863 IN IP4 192.168.10.70<br>> s=FreeSWITCH<br>> c=IN IP4 192.168.10.70<br>> t=0 0<br>> m=audio 25814 RTP/AVP 18 101 13<br>> a=fmtp:18 annexb=yes<br>> a=rtpmap:101 telephone-event/8000<br>>
a=fmtp:101 0-16<br>> a=ptime:20<br>> ------------------------------------------------------------------------<br>> recv 423 bytes from udp/[85.15.0.154]:5060 at 04:01:16.208118:<br>> ------------------------------------------------------------------------<br>> SIP/2.0 488 Not Acceptable Media<br>> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK<br>> From: "1000" <sip:register:false@85.15.0.154>;tag=Na0S1Q9mNmS1r<br>> To: <sip:140112971507247227@85.15.0.154>;tag=457FC664-6A6<br>> Date: Tue, 12 Jun 2012 04:01:24 GMT<br>> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9<br>> Server: Cisco-SIPGateway/IOS-12.x<br>> CSeq: 29400774 INVITE<br>> Allow-Events: telephone-event<br>> Content-Length:
0<br>><br>> ------------------------------------------------------------------------<br>> send 349 bytes to udp/[85.15.0.154]:5060 at 04:01:16.208201:<br>> ------------------------------------------------------------------------<br>> ACK sip:140112971507247227@85.15.0.154 SIP/2.0<br>> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK<br>> Max-Forwards: 69<br>> From: "1000" <sip:register:false@85.15.0.154>;tag=Na0S1Q9mNmS1r<br>> To: <sip:140112971507247227@85.15.0.154>;tag=457FC664-6A6<br>> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9<br>> CSeq: 29400774 ACK<br>> Content-Length: 0<br>><br>><br>><br>> plz help,thanks so much<br>><br>><br>> ________________________________<br>> From: Paul Cupis <<a ymailto="mailto:paul@cupis.co.uk"
href="mailto:paul@cupis.co.uk">paul@cupis.co.uk</a>><br>><br>> To: FreeSWITCH Users Help <<a ymailto="mailto:freeswitch-users@lists.freeswitch.org" href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>> Sent: Monday, June 11, 2012 10:51 PM<br>><br>> Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?<br>><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<br>>> 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 ymailto="mailto:consulting@freeswitch.org" href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>><br>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>><br>>
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target="_blank">http://www.freeswitch.org</a><br>><br>><br>><br>> _________________________________________________________________________<br>> Professional FreeSWITCH Consulting Services:<br>> <a ymailto="mailto:consulting@freeswitch.org" href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>><br>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>><br>> Official FreeSWITCH Sites<br>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>> <a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>> <a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>><br>> Join Us At ClueCon - Aug 7-9,
2012<br>><br>> FreeSWITCH-users mailing list<br>> <a ymailto="mailto:FreeSWITCH-users@lists.freeswitch.org" href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>><br><br><br><br>-- <br>Kristian Kielhofner<br><br>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a ymailto="mailto:consulting@freeswitch.org" href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br><a
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