similar to: Asterisk Voicemail that reacts to my AIM status

I run an Apple OS X workstation and I've got a server on the same LAN that's both a webserver and an Asterisk PBX. I wanted to be able to originate calls in the OS X Address Book application, and have Asterisk dial them and connect them to the phone on my desk. I've assembled a system that uses AppleScript to connect, via XML-RPC, to a web application that, in turn, connects to

My setup is as follows: Handset -> Sipura SPA 2000 -> Asterisk -> VoicePulse and Handset -> Sipura SPA 2000 -> Asterisk -> Digium X100P -> POTS I notice when making VoicePulse calls (but *not* POTS calls through the X100P) that there is significant "popping" and "clicking" on the line. This isn't enough to interfere seriously with the call, and

Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the "Pickup" and "Hangup" functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of

Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe("SIP/-08118800", "1234") in new stack == Parsing

I've been monitoring this problem for almost a month now. I realized that it is more extensive than what I described previously. I can very easily replicate this problem on every Zap channel. Following is the senario: 1. Call Zap/5 via say SIP/15 -> Zap/5-1 created and starts to ring 2. Call Zap/5 via say SIP/21 -> Zap/5-2 created and starts to ring 3. Hangup SIP/15 ->

Can someone tell me what this setting is supposed to be? peer->nat = globalnat; It looks like it's inheriting a parameter, but I'm curious, is globalnat an option that we're supposed to set(or let default) in sip.conf? ----- Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close

Hi Guys does anyone have a ''working'' macosx vmware instance ? i.e. you have the networking working ok so you can use ichat. would love to use macosx and textmate for RoR development but I dont fancy parting with > $US1,500 for a new Mac that is half the speed, one quarter the RAM and a fraction of the HD of my current PC. regards damian

Hi, I'm now setting up a VoIP conference room using Asterisk. All the clients are SIP phone (to be exact, Xlite), number of clients that should be registered are around 50 and concurrent users are maybe 15 clients at most. So, basically I think I can handle the situation only with Asterisk. I'm wondering however, most of my clients are behind NAT of home router and using SER together

Dear R listers, I posted this problem several days ago but it seems nobody answered. I use nlm to optimize a given function ,but it always generates the following warnings " Error in nlm(foo, theta.start) : non-finite value supplied by 'nlm' " I don't know why ,can anybody give me some hints ?? thanks in advance. regards .

Hi. I'm trying to get 'mu' to show up as a Greek symbol but, despite trying every example I could find, can't get it to work. Any insights would be welcome. This is what I'm using that works, but displays mu with the letter u. plotTimeXMastPAR <- qplot(DT,MastPAR, data=A, xlab = "", ylab = quote(PAR (uE ~m^-2 ~s^-1)), geom="line") +

hello all, I would like to introduce a summary table into the pdf along with the plots (in order to archive my data into single files automatically). Similarly, It would be great to have the result of the statistical analysis (for instance anova) in the same file. Is there a way to do that? example: pdf("example.pdf") layout (matrix(1:2,1,2)) plot (groups, scores) Result <-

Hi all, Thanks to Steven Sokol great work, the IAX2 bug in DIAX is now solved. For the interested people, you can download the new DLL (just the IAX2 version) from the following location: http://www.laser.com/dante/diax/wiax2.zip Replace the wiax2.dll file in the app directory with the new one and this is all. Please test it and send me your feedback. I intend to release a new DIAX version this

Hi, There is any other user of DIAX with this problem? Thanks, Dan

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm having a problem with the following expression examples. exten => s,1,NoOp($[$[${value} >= 10] & $[${value} < 18]]) exten => s,1,GotoIf($[$[${value} >= 10] & $[${value} < 18]]?3) ${value} is 13 in both examples above. First extension evaluates to 1 while second evaluates to 0 even though it's the same

I would like to know if anyone has come up with a script for 911 dialing rules that put correct information on our locations. We have our office in 3 different building one being our production & shipping dock. It is almost 2 blocks away. We are connected with Ethernet Wireless between the buildings and have Sip phones setup in the other 2 locations. All the phones are working just fine.

Hi I just signed up with voicepulse's voice connect service. then emailed me over configs for my extentions and iax i enter in all the info and when i start up * and do show registry it seems to be rejecting my login. Has anyone seen this before.. Any further insite will be greatly appreciated. thanks frankie (aim)cronparser (irc)crontibs 17006240093 -------------- next part

Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/1d6c78cb/attachment.htm

Hi all, I'm new to Asterisk and I was wondering if the following setup can work. If it can how would I go about setting it up: Phone------PBX------Asterisk Server------Cisco Router | | WAN

Hi, Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL. Does this new version of * only work through ODBC ? Do I have connect to MySQL through ODBC now ? Regards, Dave

I'm getting some warnings: NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Asterisk Version: CVS-01/06/04-13:50:26 Cisco ATA 186 version: v3.0.0 atasip (Build 031210A) Is this something I should be concerned about? Anyone know how to "turn off" the RFC3389 support on the ata 186? Thanks!