Having some issues with a TE131 apparently causing kernel panics. This is on
CentOS 5.10 with DAHDI 2.9.1 (also tried 2.9.2-rc1, no difference.) There’s
also an
Tracebacks are very similar to this one:
https://pbs.twimg.com/media/BrdrQf3CAAE3RXQ.jpg
It’s configured with FXSKS signaling and
On May 12, 2014, at 5:02 AM, Jan Gaida wrote:
> We are using Asterisk 1.4 as call distribution system with simple queues for
> SIP calls.
>
> With high load (4000 calls/hour) some calls remain in queue forever (until
> queue's max wait time) in spite of being hung up already by the caller. It
On May 5, 2014, at 10:15 AM, motty cruz wrote:
> Hello All,
>
> one of the extensions fall into a loop, I don't know how to hangup that
> channel
>
> -- Executing [i@autoatten:2] Goto("Local/100@sipphones-01b2;2",
> "s,2") in new stack
> -- Goto (autoatten,s,2)
> -- Sent into
On Apr 14, 2014, at 10:19 AM, Eric Wieling wrote:
> So few people use Asteisk on OSX that I doubt anyone will answer.
Yup. I’d love for someone to say otherwise, but the state of Asterisk on OS X
appears to be “can be configured to barely work for development”, and if that’s
your goal, it’s m
On Apr 11, 2014, at 3:01 AM, Manu wrote:
> Hi,
> I used asterisk on Debian7 and it was good experience.
> Now, i'm using osx on mac mini.
> I'd like to install asterisk 12.
> I tried to compile it and after lot of searches, I got it.
> All sip accounts log in. I can call but I haven't any sounds.
Is it possible to get .spec or .src.rpms for packages on packages.digium.com?
I specifically need to rebuild kmod-dahdi-linux-fwload-vpmadt032 for the
kernels available in CentOS 5.10. I see there’s source at
downloads.digium.com/pub/telephony/firmware/releases, but .spec files are not
include
I made myself look a little silly recently in a talk regarding
asteriskdocs.org. I didn’t realize the 4th ed. of the Definitive Guide was
apparently actually out (http://shop.oreilly.com/product/0636920025894.do),
because I went by asteriskdocs.org’s claim that it was being worked on in OFPS
(
On Oct 25, 2013, at 8:02 AM, virendra bhati wrote:
> For me that's matter bcoz i was working with events programming and face
> issue then I notice ,..
If you’re dealing with any interface or protocol that presents case-insensitive
strings, you should always normalize those strings you receive
On Aug 30, 2012, at 5:15 PM, Asterisk Development Team
wrote:
> * The Asterisk RESTful Interface (ARI) has been added. This interface lets
> external systems harness the telephony primitives within Asterisk to develop
> their own communications applications. Communication with Asterisk is done
On Jul 28, 2013, at 2:59 PM, Andrew Colin wrote:
> if you say allow=all it will work but thats not secure at all.
How is allow=all insecure? I can see inefficient, but what would make that
insecure eludes me.
smime.p7s
Description: S/MIME cryptographic signature
--
_
On Jul 15, 2013, at 3:35 PM, Richard Kenner wrote:
> How does one do this? We have a particular SIP phone that needs a large
> jitterbuffer, but all I can see is how to put it on the *read* side of
> the channel.
At the risk of being a little tangential, what is a write-side jitterbuffer?
sm
I'm interested in using the testing a non-DAHDI timing source to have some
assurance I'm on a system that's not likely to give me grief over
timing-related issues.
I'm familiar with dahdi_test and the guideline of needing 99.975% accuracy for
reliable conferencing and such. (Is that an accurat
12 matches
Mail list logo