>From what I have seen if your sip provider does not take g722 then you will have problems with outgoing calls. When I tried this, the same way you did, I could make calles externally but had no audio each way reguardless of what I tried to pass to the 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 <[email protected]> wrote: > Try this: http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch > > Regards, > Mindaugas Kezys > > Kolmisoft UAB > VoIP Billing Solutions > e-mail: [email protected] > URL: http://www.kolmisoft.com > > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Steve Davies > Sent: Tuesday, June 29, 2010 7:51 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Codec negotiation > > On 26 June 2010 22:08, Ryan Wagoner <[email protected]> wrote: > > I have Polycom phones that support the g722 codec. Adding allow=g722 > > to the [general] section of sip.conf works great and I can make calls > > between the phones using g722. However Asterisk is negotiating g722 > > for calls going out my voip provider and transcoding these to ulaw. In > > sip.conf for the provider I have deny=all and allow=ulaw. This can > > cause potential audio degrading and wastes cpu cycles. If Asterisk > > knows the trunk only supports ulaw why would it offer g722 to the > > phone. > > > > Ryan > > Because the codec is already chosen before the call is made, and you > told it that g722 is permitted. > > There are all sorts of discussions in play about codec negotiation, > but at this point in time, if you want different behaviour you'll need > to go and code it yourself, and cross-channeltype this is not going to > be trivial :) > > Cheers, > Steve > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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