similar to: Polycom 4.1.6 BLF/URL Dialing

On Mon, Mar 9, 2015 at 9:40 AM, David Wessell <david at ringfree.biz> wrote: > Welcome to our hell. > > We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally > got Polycom to issue a hotfix firmware version. I'll be happy to share it > with you offlist, just email me. > > Officially Polycom will fix the issue in 5.3 in a few months.. > >

Welcome to our hell. We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally got Polycom to issue a hotfix firmware version. I'll be happy to share it with you offlist, just email me. Officially Polycom will fix the issue in 5.3 in a few months.. Thanks David On Mon, Mar 9, 2015 at 9:34 AM, Andrew Colin <andrew at convergedgroup.net> wrote: > Hi Guys, > >

Hi Guys, We are getting a strange issue on certain polycom phones, sometimes when a call comes in it just "flashes" to show there is a call but does not play any sound. This problem is very intermittent and happens to maybe 2 out of 10 calls. Has any else experienced this before? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello I have a setup where i have 2 asterisk servers connected over the public internet with plenty of bandwidth, NAT on one side only. If i use IAX between the two *'s dtmf is flawless. If I use SIP, DTMF detection is around 30% or less. I have an exten to dial into and check DTMF: exten => NPANXXxxxx,1,Answer(); (actual number blanked for privacy) exten =>

I am having some difficulty getting a T100P card to work with my PRI. When I attempt to make an outbound call via: exten => 1004,1,Dial(Zap/g1/NPANXXXXXX) I see the following on the asterisk console: -- Executing Dial("SIP/sbruton-b8ce", "Zap/g1/NPANXXXXXX") in new stack Apr 21 08:18:48 NOTICE[16401]: app_dial.c:554 dial_exec: Unable to create channel of type

We fairly recently switched service providers for our 4 PRI circuits. Since that time, we started to notice some failed inbound calls. These calls terminate with an ISDN cause code 47 'resource unavailble'. Most of the time I see this error on the first or second channel on the second span in a trunk group (This is the providers trunk group for hunting, not an Asterisk trunk group). All

Make all the static constant strings as char arrays, so they can be fully stored in read-only memory. --- align/scan.c | 2 +- builder/index-validate.c | 2 +- cat/cat.c | 2 +- cat/filesystems.c | 2 +- cat/log.c | 2 +- cat/ls.c | 2 +-

Hi ! We curently have a centos 5 / asterisk 1.4 server that we have some DTMF problems with. It has a Sangoma A104d card and only port one is used to connect to the PSTN. Port 2 is conencted via a cross-over cable to a RAS for modem access and port 3 is connected for data communication via PPP. Now, I want to freshen this setup to something newer. So I installed a Scientific Linux 6 server,

Everybody, I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone. He said, "Our old Analog phone system could do it, how hard can it be?" I've gone down the path of trying

Hi - Just wondering if anyone has gotten a Polycom VVX phone to successfully do an Auto Answer with asterisk. I have an older generation of Polycom phones that do this just fine, but I can't seem to make the VVX phones work. I tried the guide here: http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167 And I have this in my diaplan:

On 2020-05-28 10:15, Doug Lytle wrote: > Everybody, > > I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone. > > He said, "Our old Analog phone system could do it,

Hi Sjoerd We (at Huawei) also have a pass for this. Originally we implemented this back in 2018 and meant to upstream it, but there were some issues with the implementation that required some changes in the code. We started revising it,a few weeks ago. I thought now that there are multiple options, maybe we can discuss our approaches, and see if there is a preference in the community for one

+ Evgeny We have a jump threading pass downstream for this that we would love to upstream. I believe Evgeny was working on exactly this, i.e. preparing it for upstreaming. ________________________________ From: llvm-dev <llvm-dev-bounces at lists.llvm.org> on behalf of Eli Friedman via llvm-dev <llvm-dev at lists.llvm.org> Sent: 23 September 2020 19:16 To: Phipps, Alan <a-phipps

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 have an asterisk 1.8.x box that intermittently returns a 401. Calls come through the same peer all the time, from the same carrier. However intermittently the asterisk box returns a 401. Below is the output of a failed call (1st) and a successful call (2nd). I can't see any difference until we get to these lines. Bad call: --- (17 headers 14 lines) --- Sending to carrierIP:5060 (no NAT)

Hi. Does anyone know whether CA(Cellular Automata), GA(Genetic Algorithm) or Multi-Agent System is available for R for Win? I've already knowna software for these areas, SWARM or startlogo. But I'd like to link R's function with these methods. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, guys! How's it going? I would like to offer you myself again to implement one of the ideas during the GSoC 2012. I want to take care of the "Learning To Rank" project, but I would also be glad to pick the "QueryParser reimplementation", "Dynamic Snippets", "Gmane Search improvements" or even "Replace socket code with ZeroMQ" project (I

From: Nikolay Ivanets <stenavin@gmail.com> This patch adds '--blocksize' command line option for guestfish and other C-based tools. This option allows specifying disk sector size. --- align/Makefile.am | 1 + align/scan.c | 8 ++++++++ align/virt-alignment-scan.pod | 2 ++ cat/Makefile.am | 1 + cat/cat.c | 8 ++++++++

Implement the -m/--mount as available in guestfish to override the automatic introspection and specify which partitions to mount instead. --- edit/edit.c | 38 +++++++++++++++++++++++++++++++++----- edit/virt-edit.pod | 36 ++++++++++++++++++++++++++++++++++++ 2 files changed, 69 insertions(+), 5 deletions(-) diff --git a/edit/edit.c b/edit/edit.c index 07790be..7a2603c 100644 ---

Valery Khamenya wrote: > are there any attempts to use LLVM in graph-rewriting (term-rewriting) > language implementations? Pure is a dynamically typed FPL based on term rewriting: http://pure-lang.googlecode.com/ It uses the LLVM JIT to compile and execute code on the fly in an interpreter-like environment, and can also batch-compile code to native executables. This includes the