similar to: Polycom config and DTMF problems

I'm trying to enable writeback mode for the root partition. However, adding the normal option "rootflags=data=writeback" to my grub.conf kernel line doesn't result in writeback mode (according to `dmesg`). Does anyone know of a way to enable this? -Doug -- Douglas E. Warner <dwarner at ctinetworks.com> Network Engineer CTI Networks, Inc.

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On 12/07/2018 03:36 PM, Administrator TOOTAI wrote: > Le 07/12/2018 à 14:32, hw a écrit : > > [...] >> >> Queues seem to be the only way to have several phones ring at once, or >> are there other ways? > > Dial(SIP/Phone1&SIP/Phone2&...&SIP/Phonex,,) > Good to know, thanks! What are the entries needed in the queue_members table when using

Full_Name: Dennis L. Warner Version: 1.07 OS: Windows2000 Submission from: (NULL) (65.122.61.34) lapack.dll, although installed in the modules subdirectory, is not found when running the dev sample.

I used this site (and perhaps a couple other Google returned) as well as the Polycom Admin guide as reference. http://www.voip-info.org/wiki/view/Getting+MWI+on+Polycom+Phones+to+work+wit h+Asterisk 299 is the extension one dials to access their voicemail (with caller ID sending to the correct voicemail). I see the bold on the wiki page telling me to put in my voicemail context, but I'm

Hello there I have been struggling with queues, because i think this is the right module for our business. My main goal, is when we receive external calls, the receptionist should be able to transfer the call to us Technicians, and I am trying to add 2 extensions to a queue name [teknisk] Extension 301 and 302. I have a test setup now which I thought should look like this: When a external call

Hello, We are planning to replace our current PBX with an Asterisk / SIP solution, and are now trying to decide which phones to get. My first thought was the Cisco 7960G, but the Cisco licensing scheme irritates me enough that I'll probably end up going with either the Polycom IP-600 or Snom 360. If anyone has any opinions of these phones, especially in comparison to each other, I would

I am getting read to roll out close to 100 polycom phones and wondered if any one knows of a program to take a list of MAC addresses, extensions, and names and generate the configuration files? -- Bruce Nortex Networks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060504/d3bb612a/attachment.htm

Do the Polycom IP phones have some programmability so you can do some programmable phone buttons like you can on the Cisco phones? If there is programmability, such as for soft-keys and the like, how would you rate Polycom's vs Cisco's capabilities? And where can one find the programming documentation? Thanx. Ray. -- Scanned for viruses and dangerous content at

Hi, I'm used to the Polycom 501, with a big message light indicator that flashes when I have a message waiting. As far as I can see when looking at pictures, the 601 (and 650) do not have this indicator light (nor does the 550 for that matter). How does it show the user that it has a message waiting? Regards, Mike -------------- next part -------------- An HTML attachment was

Hi R-helpers, I would like to subset my dataframe, keeping only those rows which satisfy the following conditions: 1) the string "dnv" is found in at least one column; 2) the value in the column previous to the one "dnv" is found in is not "0" Here's what my data look like: ??? POND_ID 2009-05-07 2009-05-15 2009-05-21 2009-05-28 2009-06-04 4 ? ? ? 101 ? ? ?

Hi, I successfully connected 2 servers via IAX but I'm pulling my hair to connect 2 extra servers , Anyone connected 3 or 4 servers together ? is it possible ? I d like to share the dialplan so _2XXXX goes to server A _3xxxx goes to serverB _4xxxxx goes to server C etc from the 4 servers any example of which one is peer, which one is user or friend would help me :-) thanks jl

Hi R-helpers, I'm trying to use this code > pvh_dnv<-pvh[sapply(pvh=="dnv"),] to make a new dataframe containing the rows from pvh that contain the value of "dnv" in ANY column. But, it's not working. I get this error Error in match.fun(FUN) : element 1 is empty; the part of the args list of 'is.function' being evaluated was: (FUN) which, to

Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don't really know where to start on measuring jitter

On 12/06/2018 08:43 PM, Antony Stone wrote: > On Thursday 06 December 2018 at 17:49:25, hw wrote: >>>> How dynamic are changes made in the database? >>> >>> If by "dynamic" you mean "quickly used" then the answer is "immediately". >> >> There's a note in some configuration file saying that dynamic extensions >>

I remember seeing something like this on the list a while ago, but I'm darned if I can find it. We have a number of Polycom IP501 phones, some of which have more than one registration on them. When a voicemail is left for a phone with only one registration, the MWI lights up and stays lit until the voicemail is listened to. However, on our phones with more than one registration, the MWI

Hi there, I'm getting a bunch of these errors from Polycom phones in 1.2.1: ERROR[24301]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 4003 in context internal I've searched the Wiki and archives to no avail - what do these errors mean? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:

Trying to figure out if a bug report should be submitted. Can anyone on 1.2.x verify of this has been corrected? I am on CVS 8/2005 If a call comes in to an extension that dials more than one channel (rings at more than one phone) both calls in the CDR show a status of answered when only one is answered, the source channel is bridged to only one of the two destination channels, but both CDRs

Hello all, I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option 81C system. The PRI line is currently setup as DMS100. Here are the relevant lines from zaptel.conf and zapata.conf: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone = us zapata.conf: [channels] language=en context=from-internal musiconhold=default switchtype=dms100

I was wondering what peoples thoughts are about this. It seem that * works just as well on Linux 2.6 as 2.4. Maybe a few small issues here and there but generally it seems to me that * is just as stable on either platform. 2.4 is the obvious choice for the highest possiblility of a stable well tested environment but 2.6 seems to have some enticing benefits. Can Linux 2.6 be considered a