[asterisk-dev] Weird transcoding error (SIP, local channels): sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/256)
Hi. I have the same problem, it only occurs when using SIP LOCAL channels. Even if you join a meetme on a local SIP channel you don't hear the voice saying 'you are the only person in this conference' asterisk just go mad piping out Asked to transmit frame type 64, while native formats is 256 (read/write = 64/256) in the console. I'm using SVN-branch-r38420. The previous guy is using version 1.2.9.1. I alos have 1.2.5 version on which the SIP LOCAL channels work fine. To what version must be upgrade/downgrade. Thanks. Regards. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Bug 2519 back again?
Hi. I'm refering to bug 2519. http://bugs.digium.com/view.php?id=2519 I have the following version of asterisk SVN-branch-1.2-r38420. If I dial SIP local channels then the console goes mad with: Aug 24 05:23:19 WARNING[19893]: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/64) and I cant hear anything. I looked into codecs issues first but everything is fine. If I dial by not using Local channels then everything is fine. Bug 2519 spoke about similar problems but that's more than a year back. Should I just upgrade to a stable version, eg 1.2.11? Sorry, if this is the wron mailing list. Thanks, Regards. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Configuration for recording quality?
Hi. I have recently posted a mail on the users mailing list, asking around how to change the quality setting of files that asterisk record for you. For instance change the 8kHz for meetme recordings to 32kHz. The reply came that you can not configure the recording settings/quality. Is this true? I was just wondering if something like this is in the pipeline? Or was thought about? I was suprised to see that asterisk, which I regard as a functionality rich product, does not allow you to do this. Surely different poeple/companies/customers using asterisk in their voice/software solutions will have different quality requirements. Thanks. Regards, Jan. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Problems loadin the res_ and cdr_odbc modules.
Hi. I am having troubles loading the res_ and cdr_odbc modules, they fail because they cannot find libodbc.so.1 I have unixODBC properly installed and the needed DNS setup correctly. Any ideas why I am having this troubles? Where is asterisk looking for the libodbc.so.1 file? And were can I configure this path? Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Place were Meetme recordings is stored.
Hi. Can anybody tell me why the meetme recordings are stored in "/var/lib/asterisk/sounds" and not in the a more obvious place like "var/spool/asterisk/meetme"? What is the "var/spool/asterisk/meetme" directory for then if meetme recordings are not stored there? The purpose of "/var/lib/asterisk/sounds" is to contain simple sounds which can be playedback to users, isn't it? Thank you. Regards. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
