[Freeswitch-users] how to use codec g729 on freeswitch ?
Steven Ayre
steveayre at gmail.com
Tue Jun 12 18:03:11 MSD 2012
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
Steve
On 12 June 2012 05:13, Samira Mh <saami_mh at ymail.com> wrote:
> sorry i forgot to paste codec on internal and external:
>
> freeswitch at internal> sofia status profile internal
>
> =================================================================================================
> Name internal
> Domain Name N/A
> Auto-NAT false
> DBName sofia_reg_internal
> Pres Hosts 192.168.10.70,192.168.10.70
> Dialplan XML
> Context public
> Challenge Realm auto_from
> RTP-IP 192.168.10.70
> SIP-IP 192.168.10.70
> URL sip:mod_sofia at 192.168.10.70:5060
> BIND-URL sip:mod_sofia at 192.168.10.70:5060
> HOLD-MUSIC local_stream://moh
> OUTBOUND-PROXY N/A
> CODECS IN G729
> CODECS OUT G729
> TEL-EVENT 101
> DTMF-MODE rfc2833
> CNG 13
> SESSION-TO 0
> MAX-DIALOG 0
> NOMEDIA false
> LATE-NEG true
> PROXY-MEDIA false
> ZRTP-PASSTHRU false
> AGGRESSIVENAT false
> STUN-ENABLED true
> STUN-AUTO-DISABLE false
> CALLS-IN 0
> FAILED-CALLS-IN 0
> CALLS-OUT 0
> FAILED-CALLS-OUT 0
> REGISTRATIONS 3
>
>
> ---------------------------------------------------------------------------------
> freeswitch at internal> sofia status profile external
>
> =================================================================================================
> Name external
> Domain Name N/A
> Auto-NAT false
> DBName sofia_reg_external
> Pres Hosts
> Dialplan XML
> Context public
> Challenge Realm auto_to
> RTP-IP 192.168.10.70
> Ext-RTP-IP 192.168.10.70
> SIP-IP 192.168.10.70
> Ext-SIP-IP 192.168.10.70
> URL sip:mod_sofia at 192.168.10.70:5080
> BIND-URL sip:mod_sofia at 192.168.10.70:5080
> ;maddr=192.168.10.70
> HOLD-MUSIC local_stream://moh
> OUTBOUND-PROXY N/A
> CODECS IN G729
> CODECS OUT PCMU
> TEL-EVENT 101
> DTMF-MODE rfc2833
> CNG 13
> SESSION-TO 0
> MAX-DIALOG 0
> NOMEDIA false
> LATE-NEG false
> PROXY-MEDIA false
> ZRTP-PASSTHRU false
> AGGRESSIVENAT false
> STUN-ENABLED true
> STUN-AUTO-DISABLE false
> CALLS-IN 0
> FAILED-CALLS-IN 0
> CALLS-OUT 0
> FAILED-CALLS-OUT 0
> REGISTRATIONS 0
>
>
> ------------------------------
> *From:* Samira Mh <saami_mh at ymail.com>
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *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 at 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 at 32000h,G7221 at 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:
> ------------------------------------------------------------------------
> INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
> Max-Forwards: 69
> From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
> To: <sip:140112971507247227 at 85.15.0.154>
> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9
> CSeq: 29400529 INVITE
> Contact: <sip:gw+cisco at 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 at 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 at 85.15.0.154>;tag=62QN1XNSF6rvD
> To: <sip:140112971507247227 at 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 at 85.15.0.154>;tag=62QN1XNSF6rvD
> To: <sip:140112971507247227 at 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
> Contact: <sip:140112971507247227 at 85.15.0.154:5060>
> 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 at 85.15.0.154>;tag=62QN1XNSF6rvD
> To: <sip:140112971507247227 at 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:
> ------------------------------------------------------------------------
> ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bK7vFUZrQK2r8NS
> Max-Forwards: 69
> From: "1000" <sip:register:false at 85.15.0.154>;tag=62QN1XNSF6rvD
> To: <sip:140112971507247227 at 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:
> ------------------------------------------------------------------------
> INVITE sip:140112971507247227 at 85.15.0.154 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
> Max-Forwards: 69
> From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
> To: <sip:140112971507247227 at 85.15.0.154>
> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9
> CSeq: 29400774 INVITE
> Contact: <sip:gw+cisco at 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 at 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 at 85.15.0.154>;tag=Na0S1Q9mNmS1r
> To: <sip:140112971507247227 at 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:
> ------------------------------------------------------------------------
> ACK sip:140112971507247227 at 85.15.0.154 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.70:5080;rport;branch=z9hG4bKpBa558yUmp0UK
> Max-Forwards: 69
> From: "1000" <sip:register:false at 85.15.0.154>;tag=Na0S1Q9mNmS1r
> To: <sip:140112971507247227 at 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 at cupis.co.uk>
> *To:* FreeSWITCH Users Help <freeswitch-users at 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 at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
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>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.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
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.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 at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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