search for: asteriskcodecnegotiationpatch

...e sip provider. Best bet is to use what your sip provider can use or find another provider that that can do g722. That's what I did when I wanted to use g726. my2cents On Tue, Jun 29, 2010 at 2:42 PM, Mindaugas Kezys <mkezys at gmail.com> wrote: > Try this: http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch > > Regards, > Mindaugas Kezys > > Kolmisoft UAB > VoIP Billing Solutions > e-mail: info at kolmisoft.com > URL: http://www.kolmisoft.com > > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at li...

...original codec negotiation patch ( http://bugs.digium.com/view.php?id=4825 report) just closed yesterday, and as well as my report (http://bugs.digium.com/view.php?id=11998), I had nothing to do but send my patches to the list. It might be good if my patches are placed at http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch, but don't know whom should I contact. Anyway sending here. thanks, Ganbold -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080215/a85363cd/attachment.htm -------------- next part -------------- A...

What is the current best practice to avoid transcoding on an outgoing call to a party whose codec preference is not known in advance? In other words, incoming calls are easy since codecs are negotiated from least-known (the remote party) to most-known (my endpoint) and my codecs can simply be preferred accordingly to match the remote. Outbound calls seem harder. Our endpoints always negotiate