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 <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 lists.digium.com] 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 <rswagoner at gmail.com> 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
>
--001485f45080fce216048a30af12
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable
<div>From what I have seen if your sip provider does not=A0take 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.</div>
<div>=A0</div>
<div>my2cents<br><br></div>
<div class=3D"gmail_quote">On Tue, Jun 29, 2010 at 2:42 PM,
Mindaugas Kezys <span dir=3D"ltr"><<a
href=3D"mailto:mkezys at gmail.com">mkezys at
gmail.com</a>></span> wrote:<br>
<blockquote style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px
0.8ex; PADDING-LEFT: 1ex" class=3D"gmail_quote">Try this:
<a href=3D"http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch"
target=3D"_blank">http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch</a><br>
<br>Regards,<br>Mindaugas Kezys<br><br>Kolmisoft
UAB<br>VoIP Billing Solutions<br>e-mail: <a
href=3D"mailto:info at kolmisoft.com">info at
kolmisoft.com</a><br>URL: <a
href=3D"http://www.kolmisoft.com/"
target=3D"_blank">http://www.kolmisoft.com</a><br>
<div>
<div></div>
<div class=3D"h5"><br><br>-----Original
Message-----<br>From: <a href=3D"mailto:asterisk-users-bounces at
lists.digium.com">asterisk-users-bounces at
lists.digium.com</a><br>[mailto:<a
href=3D"mailto:asterisk-users-bounces at
lists.digium.com">asterisk-users-bounces at lists.digium.com</a>]
On Behalf Of Steve Davies<br>
Sent: Tuesday, June 29, 2010 7:51 PM<br>To: Asterisk Users Mailing List -
Non-Commercial Discussion<br>Subject: Re: [asterisk-users] Codec
negotiation<br><br>On 26 June 2010 22:08, Ryan Wagoner <<a
href=3D"mailto:rswagoner@gmail.com">rswagoner at
gmail.com</a>> wrote:<br>
> I have Polycom phones that support the g722 codec. Adding
allow=3Dg722<br>> to the [general] section of sip.conf works great
and I can make calls<br>> between the phones using g722. However
Asterisk is negotiating g722<br>
> for calls going out my voip provider and transcoding these to ulaw.
In<br>> sip.conf for the provider I have deny=3Dall and
allow=3Dulaw. This can<br>> cause potential audio degrading and
wastes cpu cycles. If Asterisk<br>
> knows the trunk only supports ulaw why would it offer g722 to
the<br>> phone.<br>><br>>
Ryan<br><br>Because the codec is already chosen before the call is
made, and you<br>told it that g722 is permitted.<br>
<br>There are all sorts of discussions in play about codec
negotiation,<br>but at this point in time, if you want different behaviour
you'll need<br>to go and code it yourself, and cross-channeltype
this is not going to<br>
be trivial
:)<br><br>Cheers,<br>Steve<br><br>--<br>_____________________________________________________________________<br>--
Bandwidth and Colocation Provided by <a
href=3D"http://www.api-digital.com/"
target=3D"_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every
Thurs:<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 <a
href=3D"http://www.asterisk.org/hello"
target=3D"_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users
mailing list<br>To UNSUBSCRIBE or update options visit:<br>
=A0 <a
href=3D"http://lists.digium.com/mailman/listinfo/asterisk-users"
target=3D"_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br><br>--<br>_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a
href=3D"http://www.api-digital.com/"
target=3D"_blank">http://www.api-digital.com</a>
--<br>New to Asterisk? Join us for a live introductory webinar every
Thurs:<br>=A0 =A0 =A0 =A0 =A0 =A0 =A0 <a
href=3D"http://www.asterisk.org/hello"
target=3D"_blank">http://www.asterisk.org/hello</a><br>
<br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options
visit:<br>=A0 <a
href=3D"http://lists.digium.com/mailman/listinfo/asterisk-users"
target=3D"_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br>
--001485f45080fce216048a30af12--
