Thx alot. ;-)
On Wed, Apr 23, 2014 at 6:02 PM, Joshua Colp wrote:
> Gholamreza Sabery wrote:
>
>> Hello,
>>
>
> Kia ora,
>
>
> I have an Asterisk server with a public IP address and a bunch of
>> clients. Most of my clients are behind NATs (sometimes two clients are
>> behind the same NAT i.e
I can't reach digium.com or asterisk.org. Did I miss the memo?
sean
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Sean Darcy wrote:
I can't reach digium.com or asterisk.org. Did I miss the memo?
I have opened a ticket with IT. I'll keep the list apprised when the
problem is isolated and resolved.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 358
Both resolve fine for me :)
William Hetherington
w - www.willwh.com
t - @wmwh
On 26 Apr 2014 13:36, "Sean Darcy" wrote:
> I can't reach digium.com or asterisk.org. Did I miss the memo?
>
> sean
>
>
> --
> _
> -- Bandwidth and Co
William Hetherington wrote:
Both resolve fine for me :)
It seems to be sporadic. I suspect one of the DNS servers is having
issues, so it resolves fine for some and gets cached - for others not so
much.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsvi
When I run ./configure, it aborts with:
checking for uuid_generate_random in -luuid... no
checking for uuid_generate_random in -le2fs-uuid... no
checking for uuid_generate_random... no
configure: error: *** uuid support not found (this typically means the uuid
development package is missing)
But
> "JColp" == Joshua Colp writes:
JColp> The media is not carried over the SIP signaling,
Please give some credit, eh?
Given the sdp-negotiated srtp is not secure unless the sip is carried
over tls, the Best Practice is to require tls (or even sips: uris) to
agree to srtp.
Are you saying th
James Cloos wrote:
"JColp" == Joshua Colp writes:
JColp> The media is not carried over the SIP signaling,
Please give some credit, eh?
Given the sdp-negotiated srtp is not secure unless the sip is carried
over tls, the Best Practice is to require tls (or even sips: uris) to
agree to srtp.
And related thereto:
What needs to be done on kama and ast to ensure that all incoming calls
which route through a given kama box always matches a sip.conf [section]
based on the socket(7)'s remote address, w/o any consideration of the
INVITE's sip headers or body?
I tried a several variations, b
On Sat, Apr 26, 2014 at 3:37 PM, Richard Kenner wrote:
> When I run ./configure, it aborts with:
>
> checking for uuid_generate_random in -luuid... no
> checking for uuid_generate_random in -le2fs-uuid... no
> checking for uuid_generate_random... no
> configure: error: *** uuid support not found
> I think you need the libuuid and libuuid-devel packages.
"yum list available" was not showing any such package.
I installed a few other packages, including "uuid-dce-devel" and one of them
did the trick, but the install-prereq script wasn't good enough.
--
On Sat, Apr 26, 2014 at 6:12 PM, Richard Kenner wrote:
>> I think you need the libuuid and libuuid-devel packages.
>
> "yum list available" was not showing any such package.
>
> I installed a few other packages, including "uuid-dce-devel" and one of them
> did the trick, but the install-prereq scr
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