Displaying 20 results from an estimated 1000 matches similar to: "Slightly more advanced dialling.."
2014 Aug 06
1
different callerid for channels
Hi, all.
Is there any chance to set individual CALLERID(num) for channels SIP/peer1, SIP/peer2 in a call Dial(SIP/peer1&SIP/peer2). There is an option to use Dial(SIP/peer1&SIP/peer2,,M(set_callerid)), but the macro will be launched after the channel answered. Not really want to use local channel because of not quite usable cdr.
Thanks.
2017 Sep 15
3
0-client_t: null client [Invalid argument] & high CPU usage (Gluster 3.12)
Howdy,
I'm setting up several gluster 3.12 clusters running on CentOS 7 and have having issues with glusterd.log and glustershd.log both being filled with errors relating to null client errors and client-callback functions.
They seem to be related to high CPU usage across the nodes although I don't have a way of confirming that (suggestions welcomed!).
in
2017 May 29
2
Best way to know a call is being transfered
Hello
using Asterisk 1.8.32.3.
What is the best way of knowing a call is being transfered (attended and
unattended) ? And also knowing whereto (sip user) the call is being
transfered and who is the transferer ?
So I can log this information.
Kind regards.
J.
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2007 Jan 04
1
asterisk sip peer/user matching methodsforauthentication backwards?
I have considered opening a bug report on this, but wanted to get some
feedback and make sure I am not missing something in the way of a simple
work around. What is the scenario in which this impacts your
implementation?
Ours is the desire to use the same realtime SIP database for many
asterisk servers, and route the call based on a "home server" value in
the realtime database. The
2017 Sep 18
2
0-client_t: null client [Invalid argument] & high CPU usage (Gluster 3.12)
Thanks Milind,
Yes I?m hanging out for CentOS?s Storage / Gluster SIG to release the packages for 3.12.1, I can see the packages were built a week ago but they?re still not on the repo :(
--
Sam
> On 18 Sep 2017, at 9:57 pm, Milind Changire <mchangir at redhat.com> wrote:
>
> Sam,
> You might want to give glusterfs-3.12.1 a try instead.
>
>
>
>> On Fri, Sep
2010 Mar 02
1
sem package and growth curves
I have been working through the book "Applied longitudinal data analysis: modeling change and event occurrence" by Judith D. Singer and John B. Willett. I have been working examples using SAS and also using it as an opportunity for learning to use R for statistical analysis.
I ran into some difficulties in chapter 8 which deals with using structural equation modeling. I have tried to
2017 Sep 18
0
0-client_t: null client [Invalid argument] & high CPU usage (Gluster 3.12)
Sam,
You might want to give glusterfs-3.12.1 a try instead.
On Fri, Sep 15, 2017 at 6:42 AM, Sam McLeod <mailinglists at smcleod.net>
wrote:
> Howdy,
>
> I'm setting up several gluster 3.12 clusters running on CentOS 7 and have
> having issues with glusterd.log and glustershd.log both being filled with
> errors relating to null client errors and client-callback
2017 Sep 25
0
0-client_t: null client [Invalid argument] & high CPU usage (Gluster 3.12)
FYI - I've been testing the Gluster 3.12.1 packages with the help of the SIG maintainer and I can confirm that the logs are no longer being filled with NFS or null client errors after the upgrade.
--
Sam McLeod
@s_mcleod
https://smcleod.net
> On 18 Sep 2017, at 10:14 pm, Sam McLeod <mailinglists at smcleod.net> wrote:
>
> Thanks Milind,
>
> Yes I?m hanging out for
2008 Mar 05
2
Passing variables between two DUNDi/IAX2 peers
Hi.
I am trying to pass a variable from one Asterisk PBX
to another.
I'm using DUNDi with IAX2. Is there a way to do it?
I tried the following but it fails.
On peer1:
[dundi-outgoing]
switch => DUNDI/priv
exten => s,1,Set(CDR(userfield)=test)
exten => s,2,Set(DUNDIVAR=${ARG1}#TEST)
exten => s,3,NoOp(Passing ${DUNDIVAR} to DUNDi peer.)
exten => s,4,Goto(${DUNDIVAR},1)
On
2012 Oct 10
1
Change transport type on volume from tcp to rdma
Hello
I have two peers setup and working with x2 bricks each. They have been
working via tcp for the last 4-5 months.
I just got two Infiniband cards and put the on the peers. I want to
change the transport type to rdma instead of tcp but I don't see an easy
way to do this.
Can you please help me with proper instructions.
Best Regards
Ivan Dimitrov
2007 Nov 30
3
How to setup redundant SIP peers
Hello list,
I try to setup an asterisk-server with different SIP-Peers to PSTN.
The Peer are working and configured in sip.conf:
[peer1]
type=peer
host=10.10.10.1
[peer2]
type=peer
host=10.10.10.2
Now dialout is no problem. Extensions.conf says:
exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30)
But how can I setup a failure-route if the SIP-Proxy "peer1" ist not
2006 Mar 16
4
New one on me: How to UN-transfer
I'm using a Snom 320 in a CAP position and the receptionist wants to do
blind transfers. OK, no problem so far. Now she has asked me how to
UN-transfer a call, as in, she transfers a call and wants to hook the call
back before it connects (she wanted to tell the caller additional
information for example)
I don't think that this is possible as once my dialplan starts using Dial()
2009 Feb 12
4
Asterisk Queue and URL Calling
Dear All
I want to integrate sugarcrm and asterisk , so when customer call the call
center the agent or extension which answers the call , before pickup the
phone and talk to customer , view his/her information if it is available.
I do this as described below :
1-Setup login username for sugarcrm for each extension
2-Extension Users will login to the sugarcrm.
3-Develop php script to handle new
2005 Mar 22
4
Quick Newbie Question - Auto Call Routing
Hello all,
Quick description of scenario:
Would like to be able to plug in an analog line to a Digium Wildcard
X100P FXO card in an * server. Whenever that line receives a call, we
would like it to automatically ring a hunt group of VoIP soft phones
that are connected to the * server. Is this possible, and if it is,
what's the easiest way to go about it?
My thanks in advance.
Gordon
2009 Jan 19
1
Suggestions on how to create a hunt or hunt like (rollover, multi-line) group or where to get one?
I have about 5 incoming USA SIP lines, but my provider does not have any sort of roll-over or huntgroup feature. Does anybody have an idea on how I can create a general number that will ring to the next available, non-busy SIP line that I have? Is there a provider out there that would do this?
Any suggestions would be greatly welcome.
Thank you.
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2007 Jan 22
2
Samba members in NT4.0 to AD upgrade
Hi,
I've got an NT4.0 domain that I'm planning to upgrade
to Active Directory.
I've got a couple of Samba member servers in the
domain, various versions above 3.0.
So I have two questions:
1. Are there any gotchas with this scenario or will
the Samba clients
just keep working as NT4-esque clients.
2. Is it possible to "upgrade" a Samba member server
to be an Active
2016 May 22
3
Shader bytecode JIT
Hi all. I'm writing a remote 2D graphics API for embedded devices.
It includes a little GLSL 1.0-esque shader language, which is compiled
to executable SSE code at runtime.
Is LLVM a good fit for this sort of thing? At the moment I'm
compiling NASM into my code, but it doesn't do any register allocation
(which is why I'm considering LLVM). I don't want to add too much
bloat
2006 Nov 07
1
How do I make this stop? (Bridging of IAX channels?)
-- Attempting native bridge of IAX2/peer1-iax-7 and IAX2/peer2-21
I want everything to stay in the VoIP server rather then briding. I
have notransfer=yes on, but it still seems to bridge the call
natively.. can I keep the RTP stream on the asterisk server some how?
2011 Aug 17
4
[LLVMdev] Style question: NULL or 0?
Hi, LLVM. I have a question I'd like to get put into the official style guidelines: do we prefer NULL or 0 for C++ objects? I've seen both throughout the code.
Personally I prefer NULL, since it establishes that something is a pointer and not an integer (or integer-constructed object, but thankfully we avoid implicit conversions in LLVM/Clang). But I think I read somewhere that 0 is more
2008 Aug 18
5
MT9 Capabilities
Sorry, but i wasn't sure the best place to post this. I was looking for some
forums but it doesn't look like Xiph has any official ones. Anyways, i heard
about this MT9 think a while ago and i thought it would be awesome if FLAC,
Vorbis, and all could implement something like this.
http://www.guardian.co.uk/music/2008/may/27/news.seanmichaels
http://en.wikipedia.org/wiki/MT9
In