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
sorry i forgot to paste codec on internal and external:freeswitch@internal> sofia status profile internal=================================================================================================Name internalDomain Name N/AAuto-NAT falseDBName sofia_reg_internalPres Hosts 192.168.10.70,192.168.10.70Dialplan XMLContext publicChallenge Realm auto_fromRTP-IP 192.168.10.70SIP-IP 192.168.10.70BIND-URL sip:mod_sofia@192.168.10.70:5060HOLD-MUSIC local_stream://mohOUTBOUND-PROXY N/ACODECS IN G729CODECS OUT G729TEL-EVENT 101DTMF-MODE rfc2833CNG 13SESSION-TO 0MAX-DIALOG 0NOMEDIA falseLATE-NEG truePROXY-MEDIA falseZRTP-PASSTHRU falseAGGRESSIVENAT falseSTUN-ENABLED trueSTUN-AUTO-DISABLE falseCALLS-IN 0FAILED-CALLS-IN 0CALLS-OUT 0FAILED-CALLS-OUT 0REGISTRATIONS 3---------------------------------------------------------------------------------freeswitch@internal> sofia status profile external=================================================================================================Name externalDomain Name N/AAuto-NAT falseDBName sofia_reg_externalPres HostsDialplan XMLContext publicChallenge Realm auto_toRTP-IP 192.168.10.70Ext-RTP-IP 192.168.10.70SIP-IP 192.168.10.70Ext-SIP-IP 192.168.10.70BIND-URL sip:mod_sofia@192.168.10.70:5080;maddr=192.168.10.70HOLD-MUSIC local_stream://mohOUTBOUND-PROXY N/ACODECS IN G729CODECS OUT PCMUTEL-EVENT 101DTMF-MODE rfc2833CNG 13SESSION-TO 0MAX-DIALOG 0NOMEDIA falseLATE-NEG falsePROXY-MEDIA falseZRTP-PASSTHRU falseAGGRESSIVENAT falseSTUN-ENABLED trueSTUN-AUTO-DISABLE falseCALLS-IN 0FAILED-CALLS-IN 0CALLS-OUT 0FAILED-CALLS-OUT 0REGISTRATIONS 0
From: Samira Mh <saami_mh@ymail.com>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 loaded6-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/2.0Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NSMax-Forwards: 69From: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvDCall-ID: f61cf067-2ee4-1230-0cad-0050569414f9CSeq: 29400529 INVITEContact: <sip:gw+cisco@192.168.10.70:5080;transport=udp;gw=cisco>User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFYSupported: timer, precondition, path, replacesAllow-Events: talk, hold, referContent-Type: application/sdpContent-Disposition: sessionContent-Length: 234X-FS-Support: update_display,send_infoRemote-Party-ID: "1000" <sip:1000@85.15.0.154>;party=calling;screen=yes;privacy=offv=0o=FreeSWITCH 1339446571 1339446572 IN IP4 192.168.10.70s=FreeSWITCHc=IN IP4 192.168.10.70t=0 0m=audio 26616 RTP/AVP 9 0 8 18 3 101 13a=fmtp:18 annexb=yesa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20------------------------------------------------------------------------recv 410 bytes from udp/[85.15.0.154]:5060 at 03:53:07.463921:------------------------------------------------------------------------SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NSFrom: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvDTo: <sip:140112971507247227@85.15.0.154>;tag=45785134-1BDEDate: Tue, 12 Jun 2012 03:53:15 GMTCall-ID: f61cf067-2ee4-1230-0cad-0050569414f9Server: Cisco-SIPGateway/IOS-12.xCSeq: 29400529 INVITEAllow-Events: telephone-eventContent-Length: 0------------------------------------------------------------------------recv 927 bytes from udp/[85.15.0.154]:5060 at 03:53:11.004804:------------------------------------------------------------------------SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NSFrom: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvDTo: <sip:140112971507247227@85.15.0.154>;tag=45785134-1BDEDate: Tue, 12 Jun 2012 03:53:15 GMTCall-ID: f61cf067-2ee4-1230-0cad-0050569414f9Server: Cisco-SIPGateway/IOS-12.xCSeq: 29400529 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTERAllow-Events: telephone-eventContact: <sip:140112971507247227@85.15.0.154:5060>Content-Disposition: session;handling=requiredContent-Type: application/sdpContent-Length: 268v=0o=CiscoSystemsSIP-GW-UserAgent 7252 5649 IN IP4 85.15.0.154s=SIP Callc=IN IP4 85.15.0.154t=0 0m=audio 18218 RTP/AVP 0 13 101c=IN IP4 85.15.0.154a=rtpmap:0 PCMU/8000a=rtpmap:13 CN/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=ptime:20------------------------------------------------------------------------recv 425 bytes from udp/[85.15.0.154]:5060 at 03:53:11.005144:------------------------------------------------------------------------SIP/2.0 500 Internal Server ErrorVia: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NSFrom: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvDTo: <sip:140112971507247227@85.15.0.154>;tag=45785134-1BDEDate: Tue, 12 Jun 2012 03:53:15 GMTCall-ID: f61cf067-2ee4-1230-0cad-0050569414f9Server: Cisco-SIPGateway/IOS-12.xCSeq: 29400529 INVITEAllow-Events: telephone-eventContent-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/2.0Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NSMax-Forwards: 69From: "1000" <sip:register:false@85.15.0.154>;tag=62QN1XNSF6rvDTo: <sip:140112971507247227@85.15.0.154>;tag=45785134-1BDECall-ID: f61cf067-2ee4-1230-0cad-0050569414f9CSeq: 29400529 ACKContent-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/2.0Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UKMax-Forwards: 69From: "1000" <sip:register:false@85.15.0.154>;tag=Na0S1Q9mNmS1rCall-ID: 196eea77-2ee6-1230-789e-0050569414f9CSeq: 29400774 INVITEContact: <sip:gw+cisco@192.168.10.70:5080;transport=udp;gw=cisco>User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFYSupported: timer, precondition, path, replacesAllow-Events: talk, hold, referContent-Type: application/sdpContent-Disposition: sessionContent-Length: 226X-FS-Support: update_display,send_infoRemote-Party-ID: "1000" <sip:1000@85.15.0.154>;party=calling;screen=yes;privacy=offv=0o=FreeSWITCH 1339447862 1339447863 IN IP4 192.168.10.70s=FreeSWITCHc=IN IP4 192.168.10.70t=0 0m=audio 25814 RTP/AVP 18 101 13a=fmtp:18 annexb=yesa=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20------------------------------------------------------------------------recv 423 bytes from udp/[85.15.0.154]:5060 at 04:01:16.208118:------------------------------------------------------------------------SIP/2.0 488 Not Acceptable MediaVia: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UKFrom: "1000" <sip:register:false@85.15.0.154>;tag=Na0S1Q9mNmS1rTo: <sip:140112971507247227@85.15.0.154>;tag=457FC664-6A6Date: Tue, 12 Jun 2012 04:01:24 GMTCall-ID: 196eea77-2ee6-1230-789e-0050569414f9Server: Cisco-SIPGateway/IOS-12.xCSeq: 29400774 INVITEAllow-Events: telephone-eventContent-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/2.0Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UKMax-Forwards: 69From: "1000" <sip:register:false@85.15.0.154>;tag=Na0S1Q9mNmS1rTo: <sip:140112971507247227@85.15.0.154>;tag=457FC664-6A6Call-ID: 196eea77-2ee6-1230-789e-0050569414f9CSeq: 29400774 ACKContent-Length: 0plz 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,
_________________________________________________________________________
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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Official FreeSWITCH Sites
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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
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http://wiki.freeswitch.org
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