[Freeswitch-users] how to use codec g729 on freeswitch ?
Samira Mh
saami_mh at ymail.com
Tue Jun 12 08:13:10 MSD 2012
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
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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
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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