Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR's, voicemail etc, you must purchase commercial licenses for G729. /Peter Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Samira Mh Skickat: den 11 juni 2012 10:16 Till: Free SWITCH Users Help Ämne: [Freeswitch-users] how to use codec g729 on freeswitch ? hi guys, i am going to make call using codec g729, because the carrier support only codec g729 is it possible to get free that? in freeswitch implement only g729 passthrough . but how can i get it ? !DSPAM:4fd5a7d032762000267653!
but for making call from one country to another need to set codec to G729(no passthrough) because the carrier support only G729 and i want the free one not licensing , is it posiible to use media proxy so that use G729 without need to use commercial license? ________________________________ From: Peter Olsson <peter.olsson@visionutveckling.se> To: Free SWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, June 11, 2012 2:21 PM Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ? The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR’s, voicemail etc, you must purchase commercial licenses for G729. /Peter Från:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Samira Mh Skickat: den 11 juni 2012 10:16 Till: Free SWITCH Users Help Ämne: [Freeswitch-users] how to use codec g729 on freeswitch ? hi guys, i am going to make call using codec g729, because the carrier support only codec g729 is it possible to get free that? in freeswitch implement only g729 passthrough . but how can i get it ? !DSPAM:4fd5a7d032762000267653! _________________________________________________________________________ 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 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
samira, if your IP phone, ATA or gateway supports G729, then you can use FS free G729 codec. just make sure the codec is set in the dialplan. -nandy On Mon, Jun 11, 2012 at 6:13 PM, Samira Mh <saami_mh@ymail.com> wrote:
but for making call from one country to another need to set codec to G729(no passthrough) because the carrier support only G729 and i want the free one not licensing , is it posiible to use media proxy so that use G729 without need to use commercial license?
------------------------------ *From:* Peter Olsson <peter.olsson@visionutveckling.se> *To:* Free SWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Sent:* Monday, June 11, 2012 2:21 PM *Subject:* Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR’s, voicemail etc, you must purchase commercial licenses for G729.
/Peter
*Från:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *För *Samira Mh *Skickat:* den 11 juni 2012 10:16 *Till:* Free SWITCH Users Help *Ämne:* [Freeswitch-users] how to use codec g729 on freeswitch ?
hi guys, i am going to make call using codec g729, because the carrier support only codec g729 is it possible to get free that? in freeswitch implement only g729 passthrough . but how can i get it ? !DSPAM:4fd5a7d032762000267653!
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yes, my voipgateway support G729 and i set "G729" on freeswitch but when making call the error occure: G729 only for passthrough !! ________________________________ From: Nandy Dagondon <gcd@i.ph> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, June 11, 2012 3:47 PM Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ? samira, if your IP phone, ATA or gateway supports G729, then you can use FS free G729 codec. just make sure the codec is set in the dialplan. -nandy On Mon, Jun 11, 2012 at 6:13 PM, Samira Mh <saami_mh@ymail.com> wrote: but for making call from one country to another need to set codec to G729(no passthrough) because the carrier support only G729 and i want the free one not licensing ,
is it posiible to use media proxy so that use G729 without need to use commercial license?
________________________________ From: Peter Olsson <peter.olsson@visionutveckling.se> To: Free SWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, June 11, 2012 2:21 PM Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR’s, voicemail etc, you must purchase commercial licenses for G729. /Peter Från:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Samira Mh Skickat: den 11 juni 2012 10:16 Till: Free SWITCH Users Help Ämne: [Freeswitch-users] how to use codec g729 on freeswitch ? hi guys, i am going to make call using codec g729, because the carrier support only codec g729 is it possible to get free that? in freeswitch implement only g729 passthrough . but how can i get it ? !DSPAM:4fd5a7d032762000267653! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Are you bridging a call to this gateway or are you playing a soundfile to it? "passthrough" means no file based stuff is to happen on the FS server - only the codec can pass through the switch. in order to play "files" or record to a file, you will need a g729 license. HTH Chris On 11/06/2012, at 9:33 PM, Samira Mh wrote:
yes, my voipgateway support G729 and i set "G729" on freeswitch but when making call the error occure: G729 only for passthrough !!
From: Nandy Dagondon <gcd@i.ph> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, June 11, 2012 3:47 PM Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
samira,
if your IP phone, ATA or gateway supports G729, then you can use FS free G729 codec. just make sure the codec is set in the dialplan.
-nandy
On Mon, Jun 11, 2012 at 6:13 PM, Samira Mh <saami_mh@ymail.com> wrote: but for making call from one country to another need to set codec to G729(no passthrough) because the carrier support only G729 and i want the free one not licensing , is it posiible to use media proxy so that use G729 without need to use commercial license?
From: Peter Olsson <peter.olsson@visionutveckling.se> To: Free SWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, June 11, 2012 2:21 PM Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR’s, voicemail etc, you must purchase commercial licenses for G729.
/Peter
Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Samira Mh Skickat: den 11 juni 2012 10:16 Till: Free SWITCH Users Help Ämne: [Freeswitch-users] how to use codec g729 on freeswitch ?
hi guys, i am going to make call using codec g729, because the carrier support only codec g729 is it possible to get free that? in freeswitch implement only g729 passthrough . but how can i get it ? !DSPAM:4fd5a7d032762000267653!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Adding to comments by Chris, this includes voicemail, faxing, basically anything media related that freeswitch would have to accomplish on it's own. If your getting that message then you are in fact doing something media relates inside freeswitch and its telling you that you've been a naughty, naughty boy. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jun 11, 2012 7:41 AM, "Chris Mylonas" <chris@opencsta.org> wrote:
Are you bridging a call to this gateway or are you playing a soundfile to it?
"passthrough" means no file based stuff is to happen on the FS server - only the codec can pass through the switch. in order to play "files" or record to a file, you will need a g729 license.
HTH Chris
On 11/06/2012, at 9:33 PM, Samira Mh wrote:
yes, my voipgateway support G729 and i set "G729" on freeswitch but when making call the error occure: G729 only for passthrough !!
------------------------------ *From:* Nandy Dagondon <gcd@i.ph> *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Sent:* Monday, June 11, 2012 3:47 PM *Subject:* Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
samira,
if your IP phone, ATA or gateway supports G729, then you can use FS free G729 codec. just make sure the codec is set in the dialplan.
-nandy
On Mon, Jun 11, 2012 at 6:13 PM, Samira Mh <saami_mh@ymail.com> wrote:
but for making call from one country to another need to set codec to G729(no passthrough) because the carrier support only G729 and i want the free one not licensing , is it posiible to use media proxy so that use G729 without need to use commercial license?
------------------------------ *From:* Peter Olsson <peter.olsson@visionutveckling.se> *To:* Free SWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Sent:* Monday, June 11, 2012 2:21 PM *Subject:* Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR’s, voicemail etc, you must purchase commercial licenses for G729.
/Peter
*Från:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *För *Samira Mh *Skickat:* den 11 juni 2012 10:16 *Till:* Free SWITCH Users Help *Ämne:* [Freeswitch-users] how to use codec g729 on freeswitch ?
hi guys, i am going to make call using codec g729, because the carrier support only codec g729 is it possible to get free that? in freeswitch implement only g729 passthrough . but how can i get it ? !DSPAM:4fd5a7d032762000267653!
_________________________________________________________________________ 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 http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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Of course if you are faxing over g729 gl:) as said if the phone supports G729 then you just need to enable pass through mode for it, check the wiki for pass through. ~Mitch On Mon, Jun 11, 2012 at 5:14 AM, Brian Foster <bdfoster@endigotech.com> wrote:
Adding to comments by Chris, this includes voicemail, faxing, basically anything media related that freeswitch would have to accomplish on it's own. If your getting that message then you are in fact doing something media relates inside freeswitch and its telling you that you've been a naughty, naughty boy.
Brian Foster Endigo Computer LLC
Sent from a mobile device.
On Jun 11, 2012 7:41 AM, "Chris Mylonas" <chris@opencsta.org> wrote:
Are you bridging a call to this gateway or are you playing a soundfile to it?
"passthrough" means no file based stuff is to happen on the FS server - only the codec can pass through the switch. in order to play "files" or record to a file, you will need a g729 license.
HTH Chris
On 11/06/2012, at 9:33 PM, Samira Mh wrote:
yes, my voipgateway support G729 and i set "G729" on freeswitch but when making call the error occure: G729 only for passthrough !!
________________________________ From: Nandy Dagondon <gcd@i.ph> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, June 11, 2012 3:47 PM Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
samira,
if your IP phone, ATA or gateway supports G729, then you can use FS free G729 codec. just make sure the codec is set in the dialplan.
-nandy
On Mon, Jun 11, 2012 at 6:13 PM, Samira Mh <saami_mh@ymail.com> wrote:
but for making call from one country to another need to set codec to G729(no passthrough) because the carrier support only G729 and i want the free one not licensing , is it posiible to use media proxy so that use G729 without need to use commercial license?
________________________________ From: Peter Olsson <peter.olsson@visionutveckling.se> To: Free SWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, June 11, 2012 2:21 PM Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR’s, voicemail etc, you must purchase commercial licenses for G729.
/Peter
Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Samira Mh Skickat: den 11 juni 2012 10:16 Till: Free SWITCH Users Help Ämne: [Freeswitch-users] how to use codec g729 on freeswitch ?
hi guys, i am going to make call using codec g729, because the carrier support only codec g729 is it possible to get free that? in freeswitch implement only g729 passthrough . but how can i get it ? !DSPAM:4fd5a7d032762000267653!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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hi guys, thanks alot for your great replies, yes Chris, 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 ________________________________ From: Chris Mylonas <chris@opencsta.org> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, June 11, 2012 4:10 PM Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ? Are you bridging a call to this gateway or are you playing a soundfile to it? "passthrough" means no file based stuff is to happen on the FS server - only the codec can pass through the switch. in order to play "files" or record to a file, you will need a g729 license. HTH Chris On 11/06/2012, at 9:33 PM, Samira Mh wrote: yes, my voipgateway support G729 and i set "G729" on freeswitch but when making call the error occure:
G729 only for passthrough !!
________________________________ From: Nandy Dagondon <gcd@i.ph> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, June 11, 2012 3:47 PM Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
samira,
if your IP phone, ATA or gateway supports G729, then you can use FS free G729 codec. just make sure the codec is set in the dialplan.
-nandy
On Mon, Jun 11, 2012 at 6:13 PM, Samira Mh <saami_mh@ymail.com> wrote:
but for making call from one country to another need to set codec to G729(no passthrough) because the carrier support only G729 and i want the free one not licensing ,
is it posiible to use media proxy so that use G729 without need to use commercial license?
________________________________ From: Peter Olsson <peter.olsson@visionutveckling.se> To: Free SWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, June 11, 2012 2:21 PM Subject: Re: [Freeswitch-users] how to use codec g729 on freeswitch ?
The passthrough codec for G.729 is built by default, just make sure to load the module mod_g729. If you need transcoding, or use IVR’s, voicemail etc, you must purchase commercial licenses for G729. /Peter Från:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Samira Mh Skickat: den 11 juni 2012 10:16 Till: Free SWITCH Users Help Ämne: [Freeswitch-users] how to use codec g729 on freeswitch ? hi guys, i am going to make call using codec g729, because the carrier support only codec g729 is it possible to get free that? in freeswitch implement only g729 passthrough . but how can i get it ? !DSPAM:4fd5a7d032762000267653! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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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,
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 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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@85.15.0.154> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9 CSeq: 29400529 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@85.15.0.154> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9 CSeq: 29400774 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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, _________________________________________________________________________ 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 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
sorry i forgot to paste codec on internal and external: freeswitch@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@192.168.10.70:5060 BIND-URL sip:mod_sofia@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@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@192.168.10.70:5080 BIND-URL sip:mod_sofia@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@ymail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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 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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@85.15.0.154> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9 CSeq: 29400529 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@85.15.0.154> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9 CSeq: 29400774 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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, _________________________________________________________________________ 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 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@lists.freeswitch.org 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@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@ymail.com> wrote:
sorry i forgot to paste codec on internal and external:
freeswitch@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@192.168.10.70:5060 BIND-URL sip:mod_sofia@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@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@192.168.10.70:5080 BIND-URL sip:mod_sofia@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@ymail.com>
*To:* FreeSWITCH Users Help <freeswitch-users@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@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 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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@85.15.0.154> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9 CSeq: 29400529 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@85.15.0.154> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9 CSeq: 29400774 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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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The problem is on your Cisco device, not FreeSWITCH. You're sending an invite with G729: m=audio 25814 RTP/AVP 18 101 13 18=G729 codec 101=RFC2833 DTMF 13=Comfort Noise This gets the 488 Not Acceptable response from the Cisco device. So you're offering G729 to the Cisco but having it refused. This either means the Cisco device doesn't have G729 enabled or the Cisco SIP implementation doesn't like the lack of the optional a=rtpmap line. If it is the latter use <action application="set" data"verbose_sdp=true"/> before the bridge to workaround that issue. On 12 June 2012 05:06, Samira Mh <saami_mh@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@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 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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@85.15.0.154> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9 CSeq: 29400529 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@85.15.0.154> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9 CSeq: 29400774 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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,
_________________________________________________________________________ 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 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@lists.freeswitch.org 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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
First call: FreeSWITCH 192.168.10.70 send invite to 140112971507247227@85.15.0.154 offers: m=audio 26616 RTP/AVP 9 0 8 18 3 101 13 Cisco 85.15.0.154 replies with 18/SDP: m=audio 18218 RTP/AVP 0 13 101 Cisco 85.15.0.154 replies with 500 Internal Server Error Second call: FreeSWITCH 192.168.10.70 send invite to 140112971507247227@85.15.0.154 offers: m=audio 25814 RTP/AVP 18 101 13 Cisco 85.15.0.154 replied with 488 Not Acceptable Media Your Cisco is either misconfigured or is rejecting the INVITE/SDP from FreeSWITCH - you'll need to check the logs on the Cisco to see why. You should try Stevens suggestion of verbose_sdp=true and do the same two tests, but it looks like your Cisco should be happy to set the call up as G711 (PCMU) or G729 if you can get it to accept the INVITE/SDP. Regards,
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@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@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 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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@85.15.0.154> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9 CSeq: 29400529 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@85.15.0.154> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9 CSeq: 29400774 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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,
_________________________________________________________________________ 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 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@lists.freeswitch.org 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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
bu i have set the codec in dialpeer of VOIP gateway ________________________________ From: Kristian Kielhofner <kris@kriskinc.com> To: FreeSWITCH Users Help <freeswitch-users@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@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@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 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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@85.15.0.154> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9 CSeq: 29400529 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@85.15.0.154> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9 CSeq: 29400774 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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,
_________________________________________________________________________ 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 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@lists.freeswitch.org 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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner _________________________________________________________________________ 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 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
It is very clear the Cisco is responding with a 488 when the only codec offered is G729. When you offer G729 and PCMU it accepts PCMU because that is all it supports. By that time the A leg between your other endpoint and FreeSWITCH has already selected G729 and FreeSWITCH tries to transcode and it can't because you don't have the codec installed. You need to support G729 on your Cisco gateway, install the G729 codec in FreeSWITCH, or use PCMU on all of your devices. On Tue, Jun 12, 2012 at 11:40 PM, Samira Mh <saami_mh@ymail.com> wrote:
bu i have set the codec in dialpeer of VOIP gateway
________________________________ From: Kristian Kielhofner <kris@kriskinc.com>
To: FreeSWITCH Users Help <freeswitch-users@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@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@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 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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@85.15.0.154> Call-ID: f61cf067-2ee4-1230-0cad-0050569414f9 CSeq: 29400529 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=62QN1XNSF6rvD To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@85.15.0.154> Call-ID: 196eea77-2ee6-1230-789e-0050569414f9 CSeq: 29400774 INVITE Contact: <sip:gw+cisco@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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@85.15.0.154>;tag=Na0S1Q9mNmS1r To: <sip:140112971507247227@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@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,
_________________________________________________________________________ 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 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@lists.freeswitch.org 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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
_________________________________________________________________________ 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 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@lists.freeswitch.org 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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
Hi! Sorry for non-technical question on the list, but I just do not know where to ask it. Regarding to the nightly rates at Wyndham. It gets much more cheaper if we'll cooperate and book double or may be triple room. Is there any attendees that want to save some money? Best regards, Kirill
Email marketing@cluecon.com if you don't get a response here. There are some pretty hefty discounts already in place for booking at that hotel (since this years cluecon will be held in the same hotel). I can't remember what the figures are on that bit its safe to say that I don't think you'll get any more $$ discounted than what is currently available. I believe you will have to call the hotel and book your rooms with a slight discount, but you'll end up saving money through the actual cluecon tickets if you book there. I could be wrong on that. We had this discussion a few weeks ago on the community call, so my memory is a little fuzzy. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jun 11, 2012 9:58 AM, "Kirill Sysoev" <iam@onnet.su> wrote:
** Hi!
Sorry for non-technical question on the list, but I just do not know where to ask it. Regarding to the nightly rates at Wyndham. It gets much more cheaper if we'll cooperate and book double or may be triple room.
Is there any attendees that want to save some money?
Best regards, Kirill
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
The community conf call and IRC channels are the best place to get in touch w/ other ClueCon attendees. -MC On Mon, Jun 11, 2012 at 7:29 AM, Brian Foster <bdfoster@endigotech.com>wrote:
Email marketing@cluecon.com if you don't get a response here. There are some pretty hefty discounts already in place for booking at that hotel (since this years cluecon will be held in the same hotel). I can't remember what the figures are on that bit its safe to say that I don't think you'll get any more $$ discounted than what is currently available. I believe you will have to call the hotel and book your rooms with a slight discount, but you'll end up saving money through the actual cluecon tickets if you book there. I could be wrong on that. We had this discussion a few weeks ago on the community call, so my memory is a little fuzzy.
Brian Foster Endigo Computer LLC
Sent from a mobile device. On Jun 11, 2012 9:58 AM, "Kirill Sysoev" <iam@onnet.su> wrote:
** Hi!
Sorry for non-technical question on the list, but I just do not know where to ask it. Regarding to the nightly rates at Wyndham. It gets much more cheaper if we'll cooperate and book double or may be triple room.
Is there any attendees that want to save some money?
Best regards, Kirill
_________________________________________________________________________ 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 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@lists.freeswitch.org 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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (11)
-
Brian Foster
-
Chris Mylonas
-
Kirill Sysoev
-
Kristian Kielhofner
-
Michael Collins
-
Mitch Capper
-
Nandy Dagondon
-
Paul Cupis
-
Peter Olsson
-
Samira Mh
-
Steven Ayre