similar to: asterisk tries reinvite when incompatible codecs on call legs

I hope this doesn't already exist, but I couldn't find anything to help. I am installing a brand new Asterisk server, and want to use the Polycom IP331 phones. Does anyone have any steps on how to configure these? I have softphones working just fine, but for some reason I can't find a clear step by step on provisioning the Polycoms. Any help is greatly appreciated! Mark J.

I recently sent this email to a user in response to a problem report of phone calls going to voicemail without the phone ringing. I'm wondering if I've covered all bases, or whether there is some logical explanation I haven't considered, and generally what others' opinions/experiences are that relate. This is an Asterisk system, of course. ------- I looked at the server logs

We are having an issue with Asterisk 1.6.1 and the MWI turning on when a user doesn't have voicemail. We see random MWI lights come on and the phone indicates a random number of messages (its been anywhere from 1-14) when a server reload is done. I just checked one user, they have no messages old or new and the phone (Polycom IP330) indicates that they have 2 messages. The user will check for

Hi, just a question: Let say I have 2 phones with G729 onboard, but no 729 licence for Asterisk. Preferred codec set up in phones is G729, followed by ULAW, in Asterisk I have allow=ULAW deny=ALL. When call is reinvited by Asterisk will the two phones use G729 between each other or they will stick to ULAW they used for first part of the call ? A quick test showed that they will use ULAW ...

On map: only flush when old PTE was valid or invalid PTE may be cached. On unmap: always flush old entry, but skip flush for unaffected IOMMUs. Signed-off-by: Espen Skoglund <espen.skoglund@netronome.com> -- iommu.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) -- diff -r 8187fd8113f9 xen/drivers/passthrough/vtd/iommu.c ---

The memory relating guest domain should be allocated from NUMA node on which the guest runs. Because the latency of the same NUMA node is faster than that of a different one. This patch fixes memory allocation for Address Translation Structure of VT-d. VT-d uses two types of Structures for DMA address translation. The one is Device Assignment Structure. The other is Address Translation

Hi, We're routing SIP calls through Asterisk and we want to be able to reinvite calls without Asterisk performing codec conversion. We've performed the following test: Asterisk has license for G.729 installed sip.conf [general] context=default autocreatepeer=yes disallow=all allow=alaw allow=g729 canreinvite=yes nat=no We have configured two endpoints: EP1, preferred codec order

I've been using the NVidia drivers and recently managed to get Nouveau working. Well almost working. Running the command: xrandr -s 640x480 results in a desktop that spans the monitor but leaves black bands of about a third of the screen on the top and bottom. This results in an unreadable display playing games like Lbrickbuster2. I'd like the display to fill the screen. To confirm

Hello, trying out btrfs on my linux installation. I am running Funtoo with Linux 3.0 kernel. After a reboot kernel panicked (no access to error log since it is my root volume that failed). I get this using a rescue cd (2.6.38, btrfs v 0.19) and then trying to mount : [  752.129118] btrfs bad tree block start 0 131072 [  752.129152] btrfs: failed to read chunk root on sda5 [  752.132190] btrfs:

Ilia Mirkin <imirkin at alum.mit.edu> writes: > On Wed, Feb 6, 2019 at 3:28 PM Dan Espen <dan1espen at gmail.com> wrote: >> >> Ilia Mirkin <imirkin at alum.mit.edu> writes: >> >> > It would be useful to know how the screen is connected. Also please >> > grab the monitor's EDID from /sys/class/drm/cardN-connector/edid and >> >

Hi, I am faced with this dilema of asterisk not sending an ACK after it receives 200 OK from OpenSER (which is a response to a reinvite request sent by asterisk. Here is my setup Carrier<->OpenSER<->Asterisk1<->Asterisk2 A user is connected with Asterisk1 (through the carrier and OpenSER). On certain dtmf events the call is forwarded to Asterisk2 using the Dial command.

On Xen-3.3.0, domain0 Megaraid SAS (SAS 1068 controller) driver is not loading correctly if vtd support in Xen is enabled. It fails at the point of initializing firmware. I wasn''t seeing this error with Xen-3.2.2-rc3 (Unstable version), though with vtd disabled in Xen-3.3.0, it is working. Looks like a degrade problem. Any clues? Thx, Venkat

Hi all, I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk through OpenVPN seems to have the problem. From CDR, I see for 3 calls from this morning I'm aware of, that asterisk hangup after respectively 899s 894s 898s In logs I see WARNING[8213] chan_sip.c: Retransmission timeout reached on

Hi, I''ve added some preliminary support for VT-d for paravirtualized guests. This must be enabled using an ''iommu_pv'' boot parameter (disabled by default). I''ve added some python bindigs to allow xend to assign PCI devices to IOMMU for PV guests. For HVM guests this is handled in ioemu. Not sure if it makes sense to handle both cases in one place. The

On Sat, Feb 16, 2019 at 12:49 PM Dan Espen <dan1espen at gmail.com> wrote: > > Dan Espen <dan1espen at gmail.com> writes: > > > Ilia Mirkin <imirkin at alum.mit.edu> writes: > > > >> On Wed, Feb 6, 2019 at 3:28 PM Dan Espen <dan1espen at gmail.com> wrote: > >>> > >>> Ilia Mirkin <imirkin at alum.mit.edu> writes:

Hi, Keir, These patches are rebased version of Yunhong''s original patches, which were sent out before XEN 3.2 was released. These patches enable MSI support and limited MSI-X support in XEN. Here is the original description of the patches from Yunhong''s mail. The basic idea including: 1) Keep vector global resource owned by xen, while split pirq into per-domain

My Booking model has: serialize :custom_data, Hash From the console it works as expected saving values to the custom_data attribute. But when having a form with parameters like this: booking[custom_data][hello] and creating a new object in the controller like this: Booking.new( params[:booking] ), values are saved in the database with added metadata like this: ---

Fixes made in version 2: 1) Use PML4[257] for ioremap of PCI mmcfg. As full 16-bit segment support would require 44-bits. Since each slot only has 39-bits, we support 2048 PCI segments for now. This can be easily expanded if deemed necessary in the future. 2) Integrated PCI mmcfg access with existing PCI config interface for x86_64. Use MMCFG interface if offset is greater than 256.

Ilia Mirkin <imirkin at alum.mit.edu> writes: > It would be useful to know how the screen is connected. Also please > grab the monitor's EDID from /sys/class/drm/cardN-connector/edid and > attach it here. It would also be interesting to get a boot with > "drm.debug=0x1e nouveau.debug=disp=trace" which has the modeswitch in > question. The screen is connected

Hi, I put reinvite=yes in my sip.conf. For testing, I restricted the codecs to alaw. I have no modifiers in my dial command. Thus, there should be no reason not to reinvite. Call (sip, authenticated) comes in and is forward via SIP (not authenticated) to another asterisk box. Unfortunately, media path still passes through the asterisk box in the middle. Using sip debug I even can't find