Displaying 20 results from an estimated 3000 matches similar to: "asterisk sip peer/user matching methods for authentication backwards?"
2007 Jan 04
0
asterisk sip peer/user matching methods forauthentication backwards?
Hi,
I too have found this matching to be frustrating. I would like it to
behave as you describe.
Doug
--
Doug Meredith
506-854-7997 ext. 801
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Damon
Estep
Sent: Thursday, January 04, 2007 1:50 AM
To: Asterisk Users Mailing List - Non-Commercial
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
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
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
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
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.
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
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?
2013 Mar 31
1
Feature request: Need to INVITE to peer with other domain without peer domain addition
Hi, asterisk admin and users.
I need to SIP INVITE uri with domain via peer. And uri domain differ
then peer domain.
dialplan:
exten => s,n,Dial(SIP/peer1/number at domain2.com,60,r)
[peer1]
type=friend
host=domain1.com
fromdomain=domain1.com
As a result in SIP packet uri: number at domain2.com@domain1.com
I need: number at domain2.com
I can't use "SIP uri dial", i need
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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2014 Apr 29
1
IAX2 trunk on IPV6
Hi,
I have installed asterisk-1.8.25.0 on an Ubuntu server which has both an
ipv6 ip and ipv4 ip(real ip) assigned. And I have a client ubuntu with only
ipv4 ip(local ip) installed asterisk-1.8.25.0 . I want to configure the
client asterisk with the server asterisk as IAX2 peer and want to connect
to the IPV6 ip. I bind the server with ipv6 and also sending the
registration request from the
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
2018 Jun 05
2
How to execute priorities following a caller hangup in a successful Dial?
We're using Asterisk 14.7.6 and I have a dialplan that ends like this:
same => n,Dial(SIP/${EXTEN:0:4}@peer1)
same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH})
same => n,Hangup()
When peer1 hangsup, the priorities after the Dial are executed fine. But
when the caller hangsup during the Dial, the cleanup steps aren't done. Why?
I did read "Note that on a successful
2018 Jun 05
2
How to execute priorities following a caller hangup in a successful Dial?
Thanks, Anthony.
I added both 'g' and 'F' options. Now, when the caller hangs-up, my cleanup
code is run by both the caller channel and the peer channel, but I only
want the caller channel to do that.
Also, when the peer hangs-up, there is no execution of the priorities
following the Dial.
Finally, is there a way to reset all globals, maybe as a variant of
"dialplan
2004 Dec 18
1
One-way audio with SIP client only on certain calls
Hello.
I have an * server set up on a public IP. I have SIP clients at three
different locations, all behind NATs. I have all the SIP users set up
this way:
[user1]
type=friend
username=user1
secret=password1
callerid="User 1"<101>
host=dynamic
qualify=yes
context=outgoing
All three SIP clients are configured to use STUN (using
stun.fwdnet.net:3478).
Furthermore, I have
2018 Jun 05
2
How to execute priorities following a caller hangup in a successful Dial?
Thanks, Eric. I just tried a hangup handler, but it's showing a similar
problem: When the peer hangs-up, the hangup handler is not invoked and the
caller channel remains open.
same =>
n(callPeer),Set(GLOBAL(Peer${IndexIntoPeers}CurrentCallsCount)=$[${PeerCurrentCallsCount}
+ 1])
same =>
n,Set(CHANNEL(hangup_handler_push)=handleHangupByCallerOrPeer,doesntMatter,1(args))
same =>
2012 Sep 04
1
ifcpu64.c32 not working properly when used in a menu include file
The following is a pxelinux problem, specifically to do with
including config files with the menu include directive and the
ifcpu64.c32 com module.
I have a working ifcpu64.c32 setup that jumps to the label rescue64
in the case of a 64-bit CPU. The label "rescue64" defines a 64-bit
kernel and a 64-bit initrd.img. The setup jumps to a label named
"rescue32" in the case of a