similar to: Got *80 working ... now some Blacklist questions

Displaying 20 results from an estimated 1000 matches similar to: "Got *80 working ... now some Blacklist questions"

2004 Sep 18
2
IP Intercom's
Im looking for an Intercom solution thats interoperable wit Asterisk. Ive read several posts about people using the 2nd lines on some SIP phones w/speaker phone. Unfortunatley I dont that is going to cut it in a large warehouse enviroment. Does anyone have a solution that uses a "loudspeaker" ? Thank you, Steve Maroney
2005 Jul 20
4
OT: Hottie ?!?
Anyone know who that good looking female is thats on the Digium.com website ? Ok, my Real question is I noticed that Digium has relesed a new T1 card with an echo canceller. I also noticed that its supports E&M Circuits. Im I have very little knowledge on T1 circuits and traditional PBX's so what Im asking is can I use Digiums T1 card to connect to another PBX via a tie line ? Or does
2004 Aug 29
2
Servers
Hey guys, Im interested in hearing about servers (and thier hardware specs) that successfully run both asterisk and samba for an office of maybe about 12 extensions (SIP) and about 12 workstations. Im hopeing to not only replace a traditional PBXs with Asterisk/Linux but to provide a solution to needs such as a file serving, email serving, etc. Ive read the Success stories form voip-info.org but
2004 Sep 12
2
(no subject)
Hey guys, Im about to sign up for VoicePulse Connect. Of course, I plan on using my asterisk server to "register =>" with the service. I would rather sign up with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is behind another Linux box serving as my nat/firewall. Does anybody think I will have a problem ? Should I stick to IAX and VoicePulse Connect or can I use
2004 Sep 11
3
FWD
Im trying to get IAX to work between my * and FWD. I activated my iax2 account on iax.fwdnet.net and I get the output: "Registered to '65.39.205.121', who sees us as 68.14.203.254:4569" when I start asterisk. I tried used the Call Me tool on fwdnet.net but I dont get any calls even though the Call Me tool says everything looks ok. Can someone call my FWD number and just leave
2013 Nov 13
3
[Patch] credit: Update other parameters when setting tslice_ms
From: Nate Studer <nate.studer@dornerworks.com> Add a utility function to update the rest of the timeslice accounting fields when updating the timeslice of the credit scheduler, so that capped CPUs behave correctly. Before this patch changing the timeslice to a value higher than the default would result in a domain not utilizing its full capacity and changing the timeslice to a value lower
2004 Sep 03
1
Voicemail Size on Disk
Hey guys, How much disk space is used by Asterisk to store voice mail for about 10 - 20 users/mailboxes Thank you, Steve Maroney
2018 Sep 19
2
Auth process sometimes stop responding after upgrade
In data mercoled? 19 settembre 2018 09:30:47 CEST, Timo Sirainen ha scritto: > On 18 Sep 2018, at 15.15, Simone Lazzaris <s.lazzaris at interactive.eu> wrote: > > I've got a core dump, and here is the backtrace. Let me know if you want > > the core file. > It would be useful if we're able to use it. Could you use > https://dovecot.org/tools/core-tar.sh >
2004 Oct 01
2
HT 486
Does anyone know if the HandyTone 486 has the option to turn the two ethernet ports into either a switch/hub, or does it have to do NAT ? Thank you, Steve Maroney
2004 Aug 28
3
POE
Hey guys, I was wondering what POE solutions are being used ? Ive done some searching on google and found that PowerDsine seems to be good brand. Any comments,suggestions, and experiences on POE hubs other POE products would be greatly appreciated. Thank you, Steve Maroney
2018 Sep 19
2
Auth process sometimes stop responding after upgrade
On 19 Sep 2018, at 11.30, Timo Sirainen <tss at iki.fi> wrote: > >> >> On 19 Sep 2018, at 11.11, Simone Lazzaris <s.lazzaris at interactive.eu <mailto:s.lazzaris at interactive.eu>> wrote: >> >> In data mercoled? 19 settembre 2018 09:30:47 CEST, Timo Sirainen ha scritto: >> > On 18 Sep 2018, at 15.15, Simone Lazzaris <s.lazzaris at
2005 Jul 21
1
IAXY & Voicemailmain problem
I have the original version of the IAXY. I had it laying around collecting dust, now Im actually putting it to use. When I call my voicemail extension (8500), Before I get the voice prompts from the voicemail app, I hear tones that sound like the caller id tones that are heard when montoring a phone call. While watching my Asterisk CLI, I see this error at the sound of each tone: Jul 21 23:06:03
2013 Nov 11
2
problem using rJava with parallel::mclapply
Dear all, I got an issue trying to parse excel files in parallel using XLConnect, the process hangs forever. Martin Studer, the maintainer of XLConnect kindly investigated the issue, identified rJava as a possible cause of the problem: This does not work (hangs): library(parallel) require(rJava) .jinit() res <- mclapply(1:2, function(i) {
2018 Sep 18
4
Auth process sometimes stop responding after upgrade
In data marted? 18 settembre 2018 14:07:26 CEST, Aki Tuomi ha scritto: > If you are using systemd, create > /etc/systemd/system/dovecot.service.d/limits.conf and put > [Service]LimitCORE=infinity > > and run > systemctl daemon-reloadsystemctl restart dovecot Nope, I'm on a debian 7, without systemd. Anyway, I've resolved the issue: I had to set fs.suid_dumpable BEFORE
2004 Sep 24
5
Local Outbound Calls on PRI
I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had
2004 Sep 01
1
Really Wierd softphone problem ... must read
Hey guys, I have just developed this problem with my Windows XP box. I think it started since I installed XP SP2. Both SJPhone and Xlite does some kind of bridging with the speaker out port. When ever I make a sip call to where ever, the other party hears a lot of echoing. Well I noticed just now when I was playing mp3's via Winamp, the music was being played through my sip calls that I
2005 Aug 16
2
All Page ??
Does anyone know of any plans to add an intercom/all-page feature in *? The few SIP phones that have auto-answer capability would be better if Asterisk could broadcast one leg of a channel to many legs at one time. Thank you, Steve Maroney
2005 May 15
14
POE hub
I need to connect up to sixteen phones per building, I can use a cheap hub, but POE would be useful. Is there a cheap POE hub available? Everything I have seen has been expensive. Chris Mason
2005 Aug 11
0
Re: 24. Privacy Manager (Andi Strain)
Andi - I have experienced the same issue you mention and gotten no reply as to a way to fix it. I finally implemented "blacklist" into my Asterisk and added "Anonymous", "anonymous", "unknown", "Unknown", etc., into my blacklist file. When those come in with an IP address instead of a phone number but have no real name, they get the
2010 Dec 17
3
Alternative to extended recode sintax?
Dear R-users, I have a factor variable within my data frame which I derive week after week from a POSIXct variable using the cut(var,"weeks") command I have found in the chron package. The levels() command gives me: [1] "2009-03-30 00:00:00" "2009-04-06 00:00:00" "2009-04-13 00:00:00" "2009-04-20 00:00:00" "2009-04-27 00:00:00"