Displaying 20 results from an estimated 2000 matches similar to: "Best way to know a call is being transfered"
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 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
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 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
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${EXTEN:1}@peer1,30)
But how can I setup a failure-route if the SIP-Proxy "peer1" ist not
2014 Aug 06
1
different callerid for channels
Hi, all.
Is there any chance to set individual CALLERID(num) for channels SIP/peer1, SIP/peer2 in a call Dial(SIP/peer1&SIP/peer2). There is an option to use Dial(SIP/peer1&SIP/peer2,,M(set_callerid)), but the macro will be launched after the channel answered. Not really want to use local channel because of not quite usable cdr.
Thanks.
2008 Mar 05
2
Passing variables between two DUNDi/IAX2 peers
Hi.
I am trying to pass a variable from one Asterisk PBX
to another.
I'm using DUNDi with IAX2. Is there a way to do it?
I tried the following but it fails.
On peer1:
[dundi-outgoing]
switch => DUNDI/priv
exten => s,1,Set(CDR(userfield)=test)
exten => s,2,Set(DUNDIVAR=${ARG1}#TEST)
exten => s,3,NoOp(Passing ${DUNDIVAR} to DUNDi peer.)
exten => s,4,Goto(${DUNDIVAR},1)
On
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?
2006 Jan 26
6
Fail over to Pri on VoIP connection failure
I am trying to tweak my dial plan and I am running into a problem.
Sometimes my VoIP out bound calls do not complete on overseas calls(busy
or just a hang-up). Is there a way in the dial plan to automatically
dial out of my PRI when something like this happens. Either by time
limit by a failure event?
Any point in the right direction would be great
Thanks,
CLI output (cleansed to protect the
2010 Jun 30
1
RE How to break pri DID to multiple SIP Trunks
Hey Guys
I have an indial range of 61211118[01234]X being trunked sip to
xxx.yyy.189.65
Now I want to break this down into 612111180x going to xxx.yyy.188.145 and
612111184x going to xxx.yyy.189.199
reminder being used for fax->email etc etc etc
I have created the outbound routes and sip trunks
I can see that all the sip trunks are up
I can see the outbound routes are there and
2011 Mar 11
1
Anyway to monitor SIP debug from originator and terminator separate of each other on two screens?
Hi Everyone,
In order to make life easier and to do debugging easier I want to observe
"sip set debug originator" and "sip set debug terminator" on two different
putty screens. Trick is that originator calls the terminator. I can of
course put two separate calls and get sip debugs at different times but
that's not what I want to do. I want both to spit out on my two
2006 Jan 23
3
canreinvite always =no * no matter what we try :-(
been testing with a rather simple setup.
The mission is to actually get a reinvite to work on the lan.
I am trying with two sipura phones G.711 codec forced on both
both on the lan no nat no fancy options suchs as tT or H
No matter what we do asterisk hangs on to the media path, how
in the world do I get a reinvite to work where the media path
is actually handled by the two phones on the lan?
2015 Jul 15
2
how to return a transfered call to the transferrer?
Hi all
Any of you guys could point me in the right direction?
I need to make that a blind transfer to return to the transferrer when the transferee does not answer.
Scenario:
. Miss Jane Doe, our front desk attendant, picks up an external call to
Mr. Smith;
. Miss Doe flashes, dial Mr. Smith's extension and then hangup;
. Mr Smith's phone rings until timeout;
. At this point, how
2010 Jan 26
2
Attended Transfer with REFER
Hi guys,
I am wondering (and have been unable to find out thus far) whether Asterisk
sets some special channel variables or something when a call is transfered
with the REFER method.
Basically, I'm trying to figure out if it is possible to somehow get a
transferred call back to the transferrer (as it is done with the built-in
atxfer) after X seconds (or an unsuccessful attempt).
Using a
2010 Sep 03
3
How to tell if there is a transfer from CDR?
Is there any way to know if a call was transferred from reading the
CDR? Any relation in fields like UNIQUEID? Something that can be
scripted to make a special report?
--
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director de Tecnolog?a
+52-55-91169161 ext 2001
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2009 Oct 26
1
Cancel attended transfer
Hi folks,
I have a simple question regarding attended transfers. I have some
queues where agents take calls and I have configured attended transfers
between queues. That is, the agent dials the attended transfer extension
that routes it to the aproppiate transfer queue where the second agent
answers and they both talk for a while. Finally the transferrer leaves
the call with *, connecting
2006 Mar 14
3
Attended Transfer - transfer timeout, how to change?
Hi,
We are trying to use attended transfer with Asterisk 1.2.5, but when we
do the transfer and dial the new number, it times out after 3 rings and
then the callee is put back to the original agent.
Where can I adjust the timeout which applies to the number we are
transferring to? I have changed the extension for this number to timeout
at 60 seconds, but that seems to make no difference.
--