similar to: different callerid for channels

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

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

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

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

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

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

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

-- 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?

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. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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"

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

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

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

Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL:

<div>Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this info, maybe do some action.</div><div>Yes, we can parse CDRs or execute AGI script but we do not want inmplement this logic on Asterisk because it can affectš<span>performance.</span></div><div>š</div><div>02.07.2015, 15:31, "jg"

Hi, I've been trying to use the set_callerid function in the AGI. It sets the CallerIDname properly but I can't figure out how to set the CallerIDnumber. Is it at at possible ? Cheers. SL

<div>Is there any chance to create feature request for that useful functionality?</div><div>š</div><div>02.07.2015, 14:03, "Rusty Newton" <rnewton@digium.com>:</div><blockquote type="cite"><div><div><div>On Wed, Jul 1, 2015 at 4:46 AM, <span><<a href="mailto:royj@yandex.ru"

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

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

Hi, all Is there someway ability to insert custom Header to "SIP 486" message, when HANGUP application is invoked? Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HANGUP application because this can confuse a calling equipment.