Displaying 20 results from an estimated 2000 matches similar to: "asterisk sip peer/user matching methodsforauthentication 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 03
0
asterisk sip peer/user matching methods for authentication backwards?
Take an example where there is two sip users defined in sip.conf as
follows;
[peer1]
Host=192.168.1.1
...
[peer2]
Host=dynamic
Secret=password
...
[Peer3]
Config not relevant
...
The intention is to accept calls from peer1 without authentication (ip
address authentication only), but require authentication from peer2
If by chance a SIP invite comes "From"
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 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
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 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
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
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${EXTEN:1}@peer1,30)
But how can I setup a failure-route if the SIP-Proxy "peer1" ist not
2008 Mar 05
2
Passing variables between two DUNDi/IAX2 peers
Hi.
I am trying to pass a variable from one Asterisk PBX
to another.
I'm using DUNDi with IAX2. Is there a way to do it?
I tried the following but it fails.
On peer1:
[dundi-outgoing]
switch => DUNDI/priv
exten => s,1,Set(CDR(userfield)=test)
exten => s,2,Set(DUNDIVAR=${ARG1}#TEST)
exten => s,3,NoOp(Passing ${DUNDIVAR} to DUNDi peer.)
exten => s,4,Goto(${DUNDIVAR},1)
On
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.
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?
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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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
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
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
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
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
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
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