Use fisheye.freeswitch.org or do a git checkout and "git log -p" -MC On Wed, Dec 19, 2012 at 4:14 PM, Yungwei Chen <yungwei@resolvity.com> wrote:
Please tell me the list of files affected by this fix or the bug id. Thanks.****
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*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Tuesday, December 18, 2012 8:26 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] voices in the recordings are out of sync ****
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Latest version of FreeSWITCH has some updates that may fix this issue. I would update to 1.2.5.3 ASAP. -MC****
On Tue, Dec 18, 2012 at 1:56 PM, Yungwei Chen <yungwei@resolvity.com> wrote:****
Hi,
I found one issue that voices are always out of sync in the recordings. I am running freeswitch-1.2.5.1-1 on CentOS 5 (i386), which was installed from yum. I am having trouble installing the latest version from source due to an error: Autoconf version 2.62 or higher is required. It would be nice if someone can reproduce this issue against HEAD. Thanks.
Here're the steps to reproduce it. The idea is to call a phone number and then bridge to another phone number while the entire session is being recorded. 1. In dialplan/public.xml, make sure you have a dialplan to handle any 10 digit phone numbers. <extension name="public_extensions"> <condition field="destination_number" expression="^\d{10}$"> <action application="transfer" data="main XML default"/> </condition> </extension>
2. In dialplan/default/main.xml, make sure you have an extension to handle the call in the default context. <extension name="test"> <condition field="${destination_number}" expression="^main$"> <action application="set" data="RECORD_STEREO=false"/> <action application="answer" /> <action application="record_session" data="/tmp/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="bridge" data="{ignore_early_media=false}[leg_timeout=60]sofia/gateway/gw1/1234567890"/> </condition> </extension>
3. In sip_profiles/external/gateways.xml, make sure you have a gateway that allows you to make an outbound call. <include> <gateway name="gw1"> <param name="username" value=""/> <param name="password" value=""/> <param name="realm" value=""/> <param name="from-domain" value=""/> <param name="extension" value=""/> <param name="expire-seconds" value="60"/> <param name="register" value="false"/> <param name="retry-seconds" value="60"/> </gateway> </include>
4. make a call to one of the allowed 10-digit phone numbers in your environment. 5. Once the call is answered, the caller shall start to count from 1 to 60 with some pause after each number. 6. The callee shall repeat each number he/she heard from the caller. 7. You should be able to hear that 2 voices in the recoridng (/tmp/rec.wav) are out of sync.
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-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org