similar to: Voice mail: MWI problem / pjsip (13.24.0)

Here is my post issued again from the beginning in some sort of logical order I hope, with additional information as suggested by George Dunlap. I am having trouble getting PCI Passthrough to work from Dom0 to DomU I am using Xen 4.6 with CentOS kernel 3.18.34-20.el7.x86_64 on a Dell Poweredge T430. When I plug in a device to the USB port, nothing happens. I am Watching /var/log/messages in

I am having trouble getting PCI Passthrough to work from Dom0 running CentOS 7 to DomU runnning Debian 8 I am using Xen 4.6 with CentOS kernel 3.18.34-20.el7.x86_64 on a Dell Poweredge T430. I think I have set it all up correctly, but I see no message when putting a USB device into any of the USB slots on the DomU There are three other DomUs running, but I have no need of PCI Passthrough set up

The Asterisk Development Team would like to announce the release of Asterisk 13.24.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.24.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I

On 27.12.18 at 18:14 Joshua C. Colp wrote: > On Thu, Dec 27, 2018, at 1:07 PM, Michael Maier wrote: >> Hi! >> >> I just want to say, that 13.24.1 doesn't fix the problem described in >> the posts above. > > You're going to need to file an issue[1] with traces and actual configuration. > > [1] https://issues.asterisk.org/jira > Before I'm

Hi! I just want to say, that 13.24.1 doesn't fix the problem described in the posts above. Regards, Michael

What procedure did you follow to revert back to the old version? It sounds like your binary has been revereted, but the modules it needs to load are still the 13.24.0-rc1 modules... --- Hi. I am trying to upgrade my asterisk from 13.15 to the latest of asterisk 13 which seems to be 13.24.0-rc1. At the same time I want to go from Debian 8 to DEbian 9 to get a more recent operating system and

Many months ago, my machine ran WoW 3.x flawlessly. Now however, with Linux, it wont run at all. I am using Wine 1.2.1 and WoW 4.0.1 installed fine with the downloader, and the launcher runs without a hitch (although the "play" button won't appear once in a while). But once the Launcher program attempts to run the wow.exe, I get nothing. No game, no error message. Here's my

Hi all, When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has resulted in a MWI clearing delay of around 5 minutes. After listening to a voicemail and deleting it, the Polycom VVX 601's MWI light is left on for around five minutes, before clearing. Installing Asterisk 13.24.1 did not fix this. Moving back to 13.23.1 allows the MWI to clear immediately. I see a note in

I would like to code the following in R: a1(b1+b2+b3) + a2(b1+b3+b4) + a3(b1+b2+b4) + a4(b1+b2+b3) or in summation notation: sum_{i=1, j\neq i}^{4} a_i * b_i I realise this is the same as: sum_{i=1, j=1}^{4} a_i * b_i - sum_{i=j} a_i * b_i would appreciate some help. Thank you. -- View this message in context: http://r.789695.n4.nabble.com/summation-coding-tp4646678.html Sent from the R

Hi, I'm using libvirt and qemu on Debian Wheezy. I'm having a strange behavior. Guests can't connect to each other when they're on the same host. On the host I'm using bonding (in active / backup mode) and vlan. It looks like this : eth0 \ / macvtap0 bond0 --- vlan222 eth1 / \ macvtap1 So I've got two guests, let's say A and B. When

On Wed, Jun 22, 2016 at 11:49 AM, Francis Greaves <francis at choughs.net> wrote: > More information... > I have pcifront showing as a module in the DomU and the usb shows in dmesg > as: > [ 3.167543] usbcore: registered new interface driver usbfs > [ 3.167563] usbcore: registered new interface driver hub > [ 3.167585] usbcore: registered new device driver usb >

ESTIMADA COMUNIDAD R, Tengo un data frame de datos de salud sobre Enfermedades de Notificacion Obligatoria. Algunas variables tienen una codificacion 99, 999, y 9999 para asiganr los valores perdidos. LAs variables que tienen esta codificacion son la EDAD, COMUNA_RESIDENCIA y la REGION_RESIDENCIA, respectivamente. Me gustaria poder editar esos valores a NA, sin tener que hacerlo uno por uno con

I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great. However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. The call stays connected and the inbound audio works fine. The thing is, it happens on such an irregular basis (once or twice per day) that I can't get

Hello all, I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5 from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that blindpreferred=1 (all transfers default as blind transfers). If a customer calls in & we answer & transfer, everything works fine. But if we call out to a customer & then transfer to another internal extension, that

hi, let me explain in detail, what i have configured and what is happening now: 1st router w724v (Deutsche Telekom AG): - port forwarding, everything to destination port 51000-55999 to device with ip 192.168.2.50 (interface of 2nd router) 2nd router Bintec RS353j): - configured NAT, everything to port 51000-55999 to device 192.168.3.99 (same ports) other direction is totally open. I

I did a clean install to freepbx 2.6.1 and now when i do *76 i get a 1 second flash on the screen then the phone hangs up. the FOP says it is on DND but some ext are still getting calls. once i do a *76 FOP still says I am on dnd. I am running asterisk 1.6.0.21. before i was getting a message like dnd activated and dnd deactivated. i posted this on the freepbx site and here is what i got

ping times are fine as well: [root at freepbx asterisk]# ping sipgate.de PING sipgate.de (217.10.79.9) 56(84) bytes of data. 64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57

Trying to make this work, and Office 365 support is useless, giving me the following response when I asked them for help troubleshooting a 488 Not Acceptable Here. Regarding your service request about configuring your PBX system with Office 365, we do not support specific setups for PBX systems for Unified Messaging. Please contact the vendor for more specific instructions and configurations.

Arch x86_64 OS CentOS-6.4 (freepbx) Asterisk 11.4 FreePBX 2.11.0.4 Snom870 with FW-8.7.4.8 What I am attempting to do is to set a different background colour for the BLF vkeys when a station is set to DND. This is supposedly accomplished through this setting in the phones provisioning file: <vkey_blue perm="RW"> DND Blue.on Blue.pickup Blue.park Blue.message </vkey_blue>