[Freeswitch-users] how to use codec g729 on freeswitch ?
Samira Mh
saami_mh at ymail.com
Wed Jun 13 07:40:40 MSD 2012
bu i have set the codec in dialpeer of VOIP gateway
________________________________
From: Kristian Kielhofner <kris at kriskinc.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, June 13, 2012 1:40 AM
Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
Your Cisco gateway doesn't support G729. Check your codec-list
configuration in IOS.
On Tue, Jun 12, 2012 at 12:06 AM, Samira Mh <saami_mh at ymail.com> wrote:
> 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
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>
> 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
>
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>
--
Kristian Kielhofner
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
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