similar to: How to setup redundant SIP peers

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

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:

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 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

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

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

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

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

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.

-- 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?

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

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

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

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?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U)

Hi... I've been fighting this for a while now, trying clean builds of Asterisk 1.14.18, 1.14.19rc3, and then 1.6 Beta 6 today. No workee. :-( Here's the results for various calls made off-hook (push the blue Speakerphone button on the Polycom 430): 988852700 - Phone waits for me to either hit the soft-key "Send" or "EndCall". If I hit "Send",

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

As an example....if we have a call that: 1. originates via PSTN line to one of our local DID's in Seattle 2. comes into our Asterisk server in Los Angeles or Denver 3. is routed by Asterisk for termination back to a different Seattle PSTN ....and if our VOIP call termination provider requires (in order to get their best rate) all calls to go through their Nextone