similar to: usb-phones

Displaying 20 results from an estimated 10000 matches similar to: "usb-phones"

2010 Dec 12
1
Atcom IP-4B ISDN IP PBX?
Hello For customers who need a small IP PBX to handle up to four ISDN lines (in France, so I guess that means EuroISDN) instead of a PC + Asterisk and an ISDN gateway box, has someone already played with the Atcom IP-4B? www.atcom.cn/IP-BRIM.html Any feedback appreciated.
2010 Nov 24
1
action at registering or de-registering
Hi all, Perhaps someone has dealt with it before. I want to activate a bunch of my own scripts after someone has registred om my asterisk, or when his cient has de-registerded. have been skimming through AGI and AMI, and seen a lot of nice features, but not the (de-)registering events. Kind regards, Hans
2007 Jul 05
1
sounds
Just curious, I noticed that with SetLanguage() you can change it into a lot of other languages. Yes one can record them easily enough with "record", but don't like to re-invent the wheel.. Browsed through a lot of google-pages but failed to find any other languages (except for FR and ES on the digium site) Any pointers for german, dutch, greek, italian, .... prompts? Hans --
2007 Nov 06
1
dtmf / misdn
Hi all, Perhaps someone can give me a hint i the right direction... Sometimes dtmf is recognized, sometimes not. I'm using 1.2.19 asterisk with misdn for my hfc card. When i got in incoming sip-call, dtmf is recognized, When i phone my self (isdn-phone or gsm-phone) no problem with dtmf When SOME (not all) people phone me (isdn-incoming) DTMF is not recognized. How come? Either it works
2008 Oct 31
2
Friday Halloween Edition Oct 31 12 Noon EDT
Morning! This may be the "day of the dead" in some regions, but we expect the usual lively discussion today at 9AM PDT, 11 Central, 12 Noon EDT, 4PM UK and Portugal, 5PM Paris, $deity-forsaken hour down under. This Sunday, I believe the USA falls back to Standard time. Future VUC are still at 12 Noon EST. Info site: http://voipusersconference.org PSTN (724) 444-7444 enter 22622# 1#
2010 Apr 14
2
[Conference] Audio/Video
Hi guys, I'm planning of creating a speech/video conference application. This application will provide a system to see/listen to each personn present in the conference. So each ppl will have a audio and video stream. I'm wondering if you know a way to do this with asterisk or if it's supported ? If it is, i'm asking you about some documentation or related article (if you know
2014 Mar 29
2
[LLVMdev] Named Register Implementation
On Sat, Mar 29, 2014 at 12:36:45PM +0000, Renato Golin wrote: > On 29 March 2014 12:27, Joerg Sonnenberger <joerg at britannica.bec.de> wrote: > > declare void @llvm.write_register(i32 regno, i32 val) > > declare i32 @llvm.read_register(i32 regno) > > > > where regno is the DWARF name or a special reservation e.g. for IP or > > SP. > > Do front-ends
2018 Aug 27
2
feeling n00b again
Retrying, falling of the list some how :-( -------- Original Message -------- Subject: feeling n00b again Date: 2018-08-20 09:51 From: asterisk at a-domani.nl To: asterisk-users at lists.digium.com Hi all, Long time ago, I followed a Asterisk training, and both at work and at home, was able to deploy Asterisk, make all sorts of internal call (hard/soft voip-phones, incoming/outgoing,
2013 Mar 10
2
chan_mobile
Hi, I've been looking at the list at: http://www.voip-info.org/wiki/view/chan_mobile But when googling of any of the "known working" devices, there ain't any for sale anymore, probably replaced by more recent types. So, anyone around here who bought recently an BT-dongle that is working with asterisk? hw
2012 Sep 10
0
[LLVMdev] Unaligned vector memory access for ARM/NEON.
On 10 September 2012 06:44, Bob Wilson <bob.wilson at apple.com> wrote: > I don't know if anyone actually uses arm processors in big-endian mode, but it shouldn't be too hard to conditionalize it. If it does turn out to be difficult for some reason, we should at least have comments to indicate where the endian assumptions are being made. AFAICR, people used to use big-endian on
2007 Jun 12
4
GotoIf Dialplan inquiry
Hi all, I have the following in my extensions.conf: exten => s,4,GotoIf($["${CALLERID(number)}" = "8585979857" | "8585970327"]?15:5) The numbers listed above are known spammer numbers. However, when I call from any other CALLERID, it still directs me to s,15 which is the Hangup() application. Here are logs from the asterisk CLI: -- Executing
2011 Oct 18
2
[LLVMdev] Optimization for size
On 17 October 2011 15:58, James Molloy <james.molloy at arm.com> wrote: > -Os doesn’t actually exist for llc, and I can’t see an obvious place where > that condition would be set. Where do we specify if we’re optimizing for > codesize or performance? The pass manager builder has an option for Os (0, 1, 2). But all it does, AFAICR, is to disable one explosive optimization pass.
2007 May 30
3
Dial plan inquiry using GotoIf()
Hi all, I'm looking for some rudimentary insight on GotoIf() which seems to be failing on me in my dial plan. All I basically wish to do is block a particular caller. Sounds easy enough, but my ternary operator/plan currently is not properly being implemented. Can anyone spot where I'm being a momo? All extensions get forwarded to the following macro: [macro-forward] ; arg1 = phone
2015 Jun 12
2
RFC: Declaring "foo.bar" as nonS3method() ?!
>>>>> Duncan Murdoch writes: > On 12/06/2015 4:12 AM, Martin Maechler wrote: >> This is a topic ' "apparent S3 methods" note in R CMD check ' >> from R-package-devel >> https://stat.ethz.ch/pipermail/r-package-devel/2015q2/000126.html >> >> which is relevant to here because some of us have been thinking >> about extending
2008 Nov 20
2
A way to run extenrnotify when IMAP events take place...
I have IMAP voicemail working with Exchange 2003 using a single username and password for multiple mailboxes. Right now, I am setting up asterisk to use voicemail with my Cisco Call Manager (Which I detest BTW...) and I have everything working, EXCEPT: I cannot get my externnotify script to run when any changes have been made to the VoiceMail... Scenario: Bob gets a call -> Bob
2012 Sep 11
1
multiple users for jabber.conf
Hi all, Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and 11 version of asterisk. In each example i got the impression that the asterisk server is registering on a XMPP server as a single user with the credentials as specified in jabber.conf. Instead of a single xmpp-user, could that also be multiple users? For instance, for each sip-user an xmpp-user? When i skim
2010 Jul 29
1
Kerberos: Principal may not act as server ERROR
Our environment: samba4 (alpha12) running on centos 5.4. We are experimenting with Hyper-V 2008 R2 Failover Clustering, which requires Active Directory. We are trying to see if samba-4 will work as the AD server. We are trying to create 2 node failover cluster. Both nodes have joined the domain successfully (with samba-4 as the DC). But subsequent steps of creating the "Failover
2006 Apr 22
4
Pinouts for T1/E1 crossover cable WAS "RE: what cable to connect a legacy PBX to a TE410P ?"
Can't anyone stop self-promotion and tell the poor guy what he needs. A T1/E1 X-over cable using an RJ-45 (8-cond.) is pinned out as follows: 1 - 4 2 - 5 3 - NU 4 - 1 5 - 2 6 - NU 7 - NU 8 - NU NU = Not Used I have not in my experience seen any problems with using a Good Quality Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you should be fine. As far as the shielding
2006 Mar 24
2
Return all rows, split then show uniques
I''m returning some rows from a tags database that look like this: ID WORDS 1. apple banana pear 2. banana melon 3. apple peach lime What I want to do with that data is use the .split method to divide them into separate values in an array, then use the .uniq method to return a unique list of the words like so: apple, banana, pear, melon, peach, lime So in my controller I
2011 Nov 01
10
State of Asterisk+Virtualization+Timing
Greetings- I'm about to dive into the process of virtualizing some of my Asterisk (primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, the primary issue preventing me from moving was the lack of proper timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either OpenVZ or KVM, but each seem to have independent "issues" that