similar to: Setting up Polycom 501 with 2 Different Extensions

Hello all, I have some polycom 430's which I'm trying to get to work with asterisk. I have them working for the most port other than one little issue. The 430's have two line appearances. I'm trying to get the second line registered to a different extension but for some reason it's not allowing me to do this. The first line will register fine but the second line never

I am having an issue using a Polycom 501 and VoIP for outgoing calls where if I call say my credit card company and try to follow their PBX menu, the key presses never register with their PBX. It's as if every key press I make absolutely nothing is being sent to them. Is there some setting in the phone or Asterisk that I need to change to fix this issue? Thanks for any help, Andrew

Hi: I use shared internet bandwidth and the calls are very clear from around midnight till about 4 pm when it goes bad after that. Is there a way to boost the internet bandwidth for Asterisk at the peak time? Thanks Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Is the channel physically being hung up before the * tone is heard? Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't support Kewlstart-style disconnect notification. The sequence I hear on the extension, when I plug in an analog phone, is the click of the

Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us

Hi Andrew - > I have a need to be able to identify incoming calls based on some factor > (could be time of day, caller ID, dialed number, it doesn't matter.) -- > Assuming Asterisk can differentiate between the calls I want, how do I inform > the IP501? There are "only" three line appearances -- I can't simply just > ring a different appearance since there

Hello, I am just asking this because I am note sure if the problem is on my side or not, I saw some comments on SIP realtime today so I was wondering, has anybody has SIP realtime working with a softfone ? If yes, please confirm, that would give me a light. My previous message to the list is below. Thanks. Frederic ----- Original Message ----- From: Frederic Jean To:

In R 3.5.0 using the `encoding' argument of source() prevents loading files from the internet; without the `encoding' argument files can be loaded from the internet, but if they contain non-ascii characters, these are not correctly displayed under MS-Windows (but they are correctly displayed under GNU/Linux). With R 3.4.{2,3,4} there is no such problem: using `encoding' the files are

>>>>> peter dalgaard >>>>> on Sun, 3 Jun 2018 23:51:24 +0200 writes: > Looks like this actually comes from readLines(), nothing > to do with source() as such: In current R-devel (still): >> f <- file("http://home.versanet.de/~s-berman/source2.R", encoding="UTF-8") >> readLines(f) > character(0)

Hi all I been looking into the whole code strugture of chan_iax and i see there is a option to specify the refresh rate of registrations: But there is no code to actually load this from the config file thus i changed the setting in chan_so.h, and recompiled. But still my refresh rate is 60 sec. I need to get this down to 15 sec (nat /pat firewall issue) any ideas? thanks Liaan

I'm trying to use Jeremy's suggestion of dialling using just the peer name instead of user:pass@peer but I'm running into some really funky issues. It does the same thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows

At 1/16/04 7:25 AM, Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> wrote: >That's pure bullshit -- I use software RAID *specifically* because I value >my data. I don't want to buy two hardaware RAID controllers to have one >sit on the shelf just in case the first dies... and if the second dies >you're SOL because they've lasted long enough that

On R 3.5.0 (Mac) The issue appears when using the default (libcurl) method and specifying the encoding Note that using method='internal' causes a segfault if used in conjunction with encoding. (and works when encoding is not set) urlR <- "http://home.versanet.de/~s-berman/source2.R" # works url_default <- url(urlR) scan(url_default, "") # Read 7 items # [1]

Sorry to have had to post this, But I need urgent help with configuring one adit 600 I picked up from e-bay. Issues. I cannot access the console port, I am using HyperTerminal with settings VT100, 9600, 8-N-1 I also do not have any user-manual so I am kind of stuck. Any help in getting me started would be really appreciated. Any default settings like Ethernet port address, that can help me

I just built the latest R-patched from source (SVN-Revision: 73548, Last Changed Date: 2017-10-12) and the build completed without a problem but `make check' errored out: running code in 'reg-tests-1d.R' ...make[3]: *** [Makefile.common:100: reg-tests-1d.Rout] Error 1 make[3]: Leaving directory '/sources/R-patched/tests' make[2]: *** [Makefile.common:275: test-Reg] Error

*spews coffee over keyboard* - FUNNIEST - THREAD - EVER - Also one of the most insightful. Teddy, your gmail invite is on the way.

I have an issue with call parking and hope there is some undocumented feature for this. ;-) We are replacing our legacy PBX with asterisk, but to save money over time (handsets and network), I am trying to maintain the use of our legacy PBX. Asterisk extensions can not use the call parking features (not usable over trunk cards) of the old PBX, so I have to get the old PBX to use asterisk's.

Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as the 102D. And to make matters worse he starts the bid at $90.00 Beware. http://search.ebay.com/search/search.dll?query=sip+phone&ht=1&sosortproperty=1&from=R10&BasicSearch= -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

Hello, I resolved my echo issue using grandstream/estara etc etc sip phones and wcfxo interfaces from digium. I swapped out my via kt400 based msi kt4vl motherboard for an asus p4pe? i845? based motherboard and the echo has completly gone away along with aggressive suppressor option in the makefile. I hope this helps others. Brian J. Schrock Anistone Technologies, LLC 6926 Avery Rd. Dublin, OH

On Fri, 13 Oct 2017 22:36:48 +0200 Stephen Berman <stephen.berman at gmx.net> wrote: > I just built the latest R-patched from source (SVN-Revision: 73548, Last > Changed Date: 2017-10-12) and the build completed without a problem but > `make check' errored out: > > running code in 'reg-tests-1d.R' ...make[3]: *** [Makefile.common:100: > reg-tests-1d.Rout]