similar to: [zapata.conf] What is "wink"?

Hi, Did anyone manage to connect either Digium or Sangoma T1 card to any other PBX/gateway using T1 E&M Wink/Wink signaling? I'm trying to connect Avaya Definity to an Asterisk box with T100P and so far no luck. (I know I can do so with ISDN PRI, but need an additional ISDN processor card for Definity.) I tried to connect Definity to Cisco 3640 CCME (call manager express) to test the link

We're having some difficulty connecting a T1 fax board card to asterisk, with E&M Wink. We've successfully connected it to other E&M Wink T1s, but have been unable to get asterisk to hear it's dialing. Asterisk can dial down to it with no problem, and it hears all the digits, but whenever it sends digits down to asterisk, * always only "gets" the first one. This is

Does anyone know the default E&M Wink timings for Nortel DID ports? The default settings on Asterisk are: ; prewink: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ;

Attempting to talk to an Eagle Telephonics switch at a disaster exercise. Didn't think a plain old E&M wink start T1 would be this much of an issue. We finally got the Eagle to accept a call from *, but whilst I can hear the person on the Eagle, they can't hear me. When they initiate a dial out I only get the first 2 digits from their switch... Does anyone have decent

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

Hey folks, I'm pulling my hair out on this situation and would welcome some advice: I'm using the AMI Manager to Originate calls onto 2 E&M wink T1 circuits via a T4XXP card from Digium. Everything works fine for about 600(+/- 50) calls then the Manager is suddenly unable to launch calls. Using ZapBarge to listen to the channels themselves you can hear the dtmf digits being dialed,

I am setting up a box with a T100P. Everything is going well. The company I am working with has their one phone switch gear. They provisioned me a E&M Wink T1. Cannot do PRI unfortunately. We chose E&M so we could pass an unlimited number of DIDs to the trunk as apposed to FXS loopstart signaling. I can make outbound calls no problem, but I am having problems with the dial plan for inbound

I am an Asterisk newby, and I cannot seem to get Caller ID information from our T1 line. When calls appear at the phones, they say the call came from "asterisk" and unknown number. I know how Caller ID information is passed on an analog phone line (between the rings) but with a T1 line, I don't know technically how it is done. I don't see the caller's number in the

I could only get *ANI*DNIS* working one way and that was setting my signalling type on the Asterisk side to 'featd' The Definity won't send *ANI*DNIS* information back to the asterisk as far as I can tell. Other than that, I've been running it with wink/wink E&M for a while now. TN464 circuit pack on the definity side, te110p on the asterisk side. Just a crossover cable

I have about a dozen SIP phones up and working, now I want to connect the asterisk box to our Fujitsu 9600 PBX. I currently have two dial-up servers conencted to the Fujitsu PBX that I built with mgetty/pppd and have the lines provisioned the same way as those dial-up server, ESF, B8ZS, and E&M wink start, so I have confidence in the guys who set up the PBX. I've built a loop back plug

I am preparing to connect an asterisk box with a redfone fonebridge to a T1 service provider. I am doing this by testing first with another asterisk and a Sangoma card playing the role of telco. I formerly had this test configuration operating flawlessly as a PRI connection. But I discovered that I will need to use E&M, thus I've chosen the parameters as described in the subject

Howdy, New installation, trying to connect an RBS T1 with AMI/D4 framing and E&M Wink. Using a Sangoma A102d and asterisk 1.4.22-2 on Centos5 (Trixbox 2.6.2.1). Outbound calls work fine, but inbound calls fail to read the DID information, and with debug set to 10 I get the following: [Feb 2 19:40:23] DEBUG[25184] chan_zap.c: Monitor doohicky got event Wink/Flash on channel 3 [Feb 2

Hi List, I have some T1 750s; hookup via a T1 to a TE420P card. We have them all set as E&M wink trunks. We get a dial tone and can call numbers, but when i call another number the ringing I hear on my phone is very weird, its not the standard ring tone. What will cause this? Is this a * setting or something in the Adtran I am fixing, I just want good ole ringing to be heard! Any help would

I've noticed a problem with calls to Hangup when talking to my Norstars over channelised T-1 E&M trunk lines - it's been present since I started to fiddle with Asterisk last December and it's still present in 'Asterisk CVS-HEAD-08/13/04-10:37:13'. Specifically, when a call is connected to Asterisk from the Norstar DTI card to my T100p I get the following conditions

List: I already have asterisks up and running on a PRI, but where we are moving we cannot get a PRI so we are going to get T1. My question is: We are going to us E&M Wink for signaling with DTMF and caller id. The channels are going to be setup like this, 12 channels for 2-way and 12 channels for incoming only with DIDs. How would I configure the zaptel.conf? I realize that I will have two

With the new signature file I was able to verify, but there was still a bad signature: "gpg: key 0x0FC3042E345AD05D: 1 bad signature" which I highlighted below. Didn't seem to be a problem, but thought I'd point it out. I'd be glad to do additional tests if you'd like. $ gpg --list-keys /home/wink/.gnupg/pubring.kbx ----------------------------- pub

Reid Spencer wrote: > On Mon, 2006-02-27 at 20:43 -0800, Wink Saville wrote: > >>> Try passing "-L/opt/llvm-1.6/llvm-gcc/lib/ -lcrtend" to llvm-ld. >>> >> This didn't work for me: >> llvm-ld -o t1.app t1.bc t1.a -L/opt/llvm-1.6/llvm-gcc/lib/ -lcrtend.a >> >> I get two warning's about library 'crtend' missing

I ran "ninja check-asan" and no errors. But "ninja check-msan" had 117 errors. I took the first FAILED test, which was for eventfd.cc, and executed the command line creating an eventfd executable in a temporary directory and then executed that file using gdb. Finally, used bt to dump the stack. I've emailed llvm-admin at lists.llvm.org to setup an account since

And again... LLVM_BUILD_LLVM_DYLIB:BOOL=ON LLVM_LINK_LLVM_DYLIB:BOOL=ON This one is the good one... maybe. On Tue, Sep 6, 2016 at 11:35 PM, Alexandre Isoard < alexandre.isoard at gmail.com> wrote: > That is because I mistyped it: > LLVM_ENABLE_LLVM_DYLIB:BOOL=ON > LLVM_LINK_LLVM_DYLIB:BOOL=ON > > On Tue, Sep 6, 2016 at 11:31 PM, Wink Saville <wink at saville.com>

Hi All, I have a rather peculiar problem. Whenever I dial out over ZAP/g0 the phone will just ring and ring, even if I answer the phone on the other end. Whats strange is that the * phone will continue to ring even after I've answered and (sometimes) hung up the dialed phone. If I make an extension to just directly dial out on ZAP/1, its almost the same behavior, it will continue to