similar to: Dialing time-out

Does anyone know if FreePBX can be installed on a Linux box that was built using Asterisk@Home. I would prefer to manage Asterisk with FreePBX over the AAH build. I have just not had good luck building an Asterisk system from scratch and the Centos based Amp ISO and prebuilt config files are a wonderful place to start. Nothing against Asterisk or Linux. My build from scratch issues are only

Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_ W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewIte m Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid =27 But looking for real world feedback. Cheers,

Hi all, Muppet from the UK asking for help We are just about to have a T1 line installed in our office in Dallas and "Advantex" the supplier has sent a questionnaire asking a number of questions. I have put the question area at the bottom of the email, we will be using Digium's hardware. could anybody help :-) In the UK when I asked for a E1, number of trunks required and the

Calling all Definity admins, Got a few questions about Definity -> Asterisk interoperability. 1) What are the options for integration? Can I hand off extensions from the Definity and vice versa? 2) Anybody have any working configs they would like to post? I've found and read the legacy integration on the wiki about the two systems. I've also googled and found a few threads that were

I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? bye Ronald Wiplinger

Greetings List! I'm currently rolling out a new deployment of Asterisk 1.8 to replace existing 1.2 servers...and have run into an issue which could use your assistance! For testing I have trunked (iax2) two of the servers - one running 1.8 and the other at 1.2. Calls placed from SIP --> SIP sound fantastic and crystal clear. However, when I place a echo test call (*43) from 1.8 to 1.2

Hello: We noticed that opus reconstructed noise is pulsing with a 400ms pattern when dtx is enabled in silk mode. This is independent of the background noise level and is found with speech + non-speech period test files as well as variable level noise-only test files. This issue can be reproduced with opus v1.1 using this command: ./opus_demo voip 16000 1 25000 -dtx input.bin

On Tue, 2005-01-18 at 09:49 -0600, asterisk-dev-request@lists.digium.com wrote: > > Hi, > > I am Arnaud F?vrier, I teach in a technical university in Marseille. > > I'd like to know if is is possible to connect a very old phone to > asterisk and dial pulses with it? > > Are digium cards pulse dial compatible? > > Is there any specific configuration

This may not be the right place to ask, but I have an annoying issue with a Sipura/SPA1000-2.0.10(e) ATA device connected to an Asterisk box. (The system is remote to me, so I've only been able to observe this by dialling into a VoIP phone on-site, then run commands on the box remotely!) First of all it's all working fine connected to an Asterisk box and the user can make/take calls

Hi. Does anyone know about a simple one-fxs ATA with pulse dialing support that can work with Asterisk? A SIP one would be ok. I've been told that the Digium S101i IAXy does support pulse dialing; although it's a iax2-only ata it could be enough. I need a bunch of them to convert some old fashioned rotary phones into VoIP ones (I'd like to disassemble the ATAs to remove the boards

Back to basics here. I want to match on one OR two digits. The following two both work, but ONLY for more than one digit, which is not as expected from the docs (see below). exten => _X.,1,SayNumber(${EXTEN}) exten => _[0-9].,1,SayNumber(${EXTEN}) This next one will ONLY match 2 digits, as expected, but the first two SHOULD match one or more, right? exten => _XX,1,SayNumber(${EXTEN})

Hello. I have not seen that asterisk software have a possibility to dial pulse on outgoing calls. Don't you know is there any plan to do it? Thanks. Good luck. Lev.

Hello, I need to write a script which prompts the callee to type a number, and then read it back to them as confirmation: ======= extensions.conf [robocall] ;Expect 10-digit number excluding final #, 2 tries, 20s time-out exten => s,n(nbr2call),Read(NBR2CALL,please-type-number,10,,2,20) exten => s,n,GotoIf($[${LEN(${NBR2CALL})} != 10]?end) ;exten => s,n,SayDigits(${NBR2CALL}) exten

I had normal version of wine for fallout2 and war3, but I've deleted it and I can't reinstall it now. I have Debian testing and newest wine 1.3 from ubuntu ppas. Nvidia video When I start fallout 2 I see start picture and game hangs. Same problems with warcraft3. there are some strange sound artifacts (like pulsing) in both games. The problem exists both in windowed and fullscreen mode

hi all, i am trying to get my TE410P (see previous posts) working in Austria (telekom Austria - i am still waiting for an answer for my questions). my /etc/zaptel.conf looks like -------------------------------- span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 loadzone=at defaultzone=at -------------------------------- after

http://bugs.digium.com/bug_view_page.php?bug_id=0001429 * Support for other language syntaxes in saynumber Accidentally I opened this can of worms to see if we can add support for other language syntaxes for saying numbers. Seems like Swedish, english and norwegian follow the same syntax. I've integrated existing patches for french, danish and soon portuguese syntax. The steps we're

; *99: ; 99 bottles of beer on the wall. exten => *99,1,Noop(99 Bottles of beer on the wall) exten => *99,n,Answer() exten => *99,n,Set(bottles=99) exten => *99,n(loop),Noop(There are ${bottles} bottles of beer on the wall) exten => *99,n,SayNumber(${bottles}) exten => *99,n,Noop(Take one done and pass it round and there's) exten =>

Hi, Currently, it seems impossible to use the output of SayNumber application as an input to Read application. So, if you want to develop an IVR with something like "You've got 23 messages. Type 1 to listen to the first one. Type 2 to leave", you must parse this message into 3 pieces and want for the last one play to start listening of user input : Playback ( "You've

Hello, I use the SayNumber() with variable. for example the number 1234 - asterisk play the number without and. -- Executing [888 at from-internal:1] Set("SIP/103-0000035d", "LANGUAGE=en") in new stack -- Executing [888 at from-internal:2] SayNumber("SIP/103-0000035d", "1234") in new stack -- <SIP/103-0000035d> Playing

Hi all. I have this configuration: Telco <-----(E1)----->TE410P//Dual Xeon Server 2.4Ghz<-----(Ethernet)----->Switch<----->GS//BT The Server is running RedHat Linux 8.0 with kernel 2.4.18-14-smp and we are having the following 2 issues: 1.- When making calls from the GrandStream to the PSTN the audio is choopy, plus theres is a pulsing sound, but when the GS