similar to: MWI issue

I am trying to compile the certified-asterisk-11.6-cert5 code and when I try to start it and then go into the console I am getting the error message "asterisk dead but subsys locked". Can anyone help with why this is happening? I have never seen this before. This is a fresh install on a new server CentOS 6.5. Thanks, Scott Haley IS Voice Projects Team Edward Jones Investments Phone:

I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place calls inbound, everything works fine. If I place calls outbound, originating from the Asterisk box, everything works fine (I have done this with the use of the .call files). If I setup an extension with the findme-followme feature and have it try to hair-pin a call back out the same trunk to the Avaya, I get a

I have setup a trunk on Asterisk 11.7 to an Avaya Session Manager. Every time I try to send a call over it, the call gets rejected. Here is the sip debug trace. Could anyone tell me what may be going wrong? nxdasterisk-2*CLI> [Apr 23 08:20:59] WARNING[19047]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/scott.call: Operation not permitted Audio is at 18380

I have a need to issue a dial command to a number: same => n,Dial(${DIALGROUP1},${TIMER1},t) After a number of seconds, let's say 10 seconds. I want to dial another set of numbers while continuing to ring, or interrupting the first group of numbers. same => n,Dial(${DIALGROUP2},${TIMER1},t) Is there a way to do this without interrupting the first call? Thanks, Scott Haley If you

I am trying to transition an application over from a FreePbx box to a Standard build Asterisk 11.6 box. I have a job that creates a call file and plays a sound file. If it detects a voicemail, then it plays it, waits 1 second and replays it. The FreePbx box works fine but the Standard Asterisk build is dropping the call during the first Voicemail playback and it does not leave the voicemail.

You could use MTR command. Its a trace route improved. Marlon Araujo > On Jan 20, 2015, at 08:59, asterisk-users-request at lists.digium.com wrote: > > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or,

I'm new to the CentOS arena and I'm curious as to what virtualization products are preferred? The two main ones that I know of are Xen and VMWare. VMWare seems to be the most capable inre: number and size of vm's allowed. When looking at Xen it appears that since Citrix acquired them the free version is cripple-ware and is very restrictive. Thanks in advance for any information.

I just registered an Avaya 9620 set to my Astlinux system (0.47 - Asterisk 1.2.22), using Avaya SIP Firmware version 2.0.1.34. Set mailbox=ext at default in the sip.conf Found MWI worked immediately. Turned off as expected. Have Fun! Tom -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm new to the list, so I apologize in advance if I'm beating a dead horse by asking this, but I read somewhere that asterisk 1.4 has MWI working for Avaya and their rather troublesome SIP firmware. Can anyone verify this before I go changing phone systems around? Thanks Derek -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey everybody, I've just spent the last two hours Googling and searching the Wiki. I'm trying to find if there are any listings of codes for the Avaya Definity G3R, to allow for an Asterisk system to turn on/off a phones MWI that is attached to a G3. We are looking to use an Asterisk system as a voice mail server. I'm not having any luck, anybody have such information? Thanks,

Has anyone tried to set Message Waiting Indicators up when public network access and voicemail service are managed by an Asterisk server TDM-connected to a legacy PBX serving analog and digital phones ? For instance: Location 1: - 200 users on a legacy PBX - among those users, 50 have access to voicemail service - TDM trunk to Location 2 Location 2: - 100 users on Asterisk - PSTN access - TDM

I have an Avaya 4602 IP phone that was previously working with Asterisk. It was being used elsewhere for several months, and I recently set it up again to work with Asterisk. Everything works fine for several minutes -- I am able to receive and make calls as expected. However, after a few minutes, and every few minutes thereafter, I get the following message on the console: -- Got SIP

i was reading posts on wiki and noticed lots of posts about Avaya 4610 handset having issue with MWI, Anyone has any more updates? Is this still the case? Any good tutorial for configuring these phones and Asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080318/3d397ca7/attachment.htm

Hi Guys, I was wondering what our next steps should be to solve the slow write times. Recently I was debugging a large code and writing a lot of output at every time step. When I tried writing to our gluster disks, it was taking over a day to do a single time step whereas if I had the same program (same hardware, network) write to our nfs disk the time per time-step was about 45 minutes.

Has anyone had any luck getting the Avaya IP SoftPhone to work in wine? I get these errors on Ubuntu 8 Code: preloader: Warning: failed to reserve range 00000000-60000000 wine: creating configuration directory '/home/microchp/.wine'... preloader: Warning: failed to reserve range 00000000-60000000 preloader: Warning: failed to reserve range 00000000-60000000 err:dosmem:setup_dos_mem

On Mon, Jun 26, 2017 at 7:40 PM, Pat Haley <phaley at mit.edu> wrote: > > Hi All, > > Decided to try another tests of gluster mounted via FUSE vs gluster > mounted via NFS, this time using the software we run in production (i.e. > our ocean model writing a netCDF file). > > gluster mounted via NFS the run took 2.3 hr > > gluster mounted via FUSE: the run took

Hi, Today we experimented with some of the FUSE options that we found in the list. Changing these options had no effect: gluster volume set test-volume performance.cache-max-file-size 2MB gluster volume set test-volume performance.cache-refresh-timeout 4 gluster volume set test-volume performance.cache-size 256MB gluster volume set test-volume performance.write-behind-window-size 4MB gluster

On Fri, Jun 23, 2017 at 9:10 AM, Pranith Kumar Karampuri < pkarampu at redhat.com> wrote: > > > On Fri, Jun 23, 2017 at 2:23 AM, Pat Haley <phaley at mit.edu> wrote: > >> >> Hi, >> >> Today we experimented with some of the FUSE options that we found in the >> list. >> >> Changing these options had no effect: >> >>

Hello, I'm trying to create a H.323 trunk between Asterisk 11 and Avaya. I have done this before between Asterisk 1.6 and Avaya but had some issues placing external calls from the Asterisk to the Public network which is connected to Avaya. I'm trying to create that trunk on Asterisk 11 because the 1.6 is outdated and has no support. On the Asterisk side I have Aastra 6731i SIP phones

Hi ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --