similar to: E&M and Dial tone

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system correctly

Maybe one of you can help me with this: We have T1's that come from both MCI and Global Crossing as uses channelized (24 Ports per T) with inband (DTMF) ANI and DNIS delivery (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk Server. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure

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

I am trying to make an in/out trunk group comprised of 4 DS0's using E&M Wink signalling. The first four channels of a DS1 on a T100P are being used for the group. Outbound calls work fine, but inbound calls fail. The other 20 DS0 channels are used for a PRI. Does the configuration shown below look okay? I've tried setting 'immediate => yes' without success, but it

When I send a call from my TE410P using E&M, the legacy PBX answers the call but doesn't route it. Any idea what this could be? I assume the digits aren't being delivered properly to the legacy pbx. Any suggestions on what config settings to muck with? Asterisk SVN-branch-1.2-r40901 built by root @ pbx04 on a i686 running Linux on 2007-01-14 14:05:02 UTC zaptel.conf

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

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

I have a couple of live T1s sitting around and they are not ISDN(like most of the people that are using Asterisk seem to be using), they are regular old 24 channel, robbed-bit, E&M wink start, D4AMI T1 circuits. Can I get these T1s to work with a T100P Digium card and asterisk? Searching through the lists and the documentation I haven't seen any examples of how to configure this kind

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

Hi, Can I make e&m wink start lines just wait for digits - instead of going to default? Someone else cleared a similar problem (as described below) on an fxo port with "usecallerid => no" but it is not doing the trick for me. In this case the line when straight to default which would be ok also. John I posted the stuff below about a week ago... I set up a t1 from my sys75 to

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

Howdy, Just installed a new switch in a new location (Ubuntu, 2.6.24-24 kernel, zaptel 1.4.12.1 built from source, libpri-1.4.10.1 built from source, asterisk 1.4.26 built from source, wanpipe 3.5.4 built from source, Sangoma A104d with firmware that is probably a year old). I plugged in an RBS T1, ESF, B8ZS, wink start, and MF signalling. I stuck with the defaults that the wanpipe build

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

Hello, I have a T1 (not PRI) plugged into my Asterisk server. Everything is working well, except I only get the first digit of the 4 digit DID in Asterisk. The T1 provider (Eschelon) tried switching to 7 digits, and I only got the first digit of the 7. Can anybody help? We're adding another DID and I need to trap it correctly. Thanks. Here are snippets from the relevant files: --

I have a TE410P with two real Telco T1's and the other 2 ports terminate into an in-house voice mail/IVR system. Calls arrive from Telco are routed to the appropriate in-house system based on the DID Digits. This part works perfectly. Now I what to allow the in-house VMS to dial though asterisk to the Telco T1's. The VMS complains there is no dial tone on its T1 and drops the attempt.

Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system to work with *? I'm not getting inbound or outbound calls to work. (Inbound caller gets line busy tone.) SETTINGS FROM MITEL: I built a Crossover cable and connected it like this: PSTN--T1--ASTRISK--T1--OLD MITEL -Crossover Cable Pin-out: 1 - 4 2 - 5

All, One of our customers is using a Telrad PBX, we are providing phone server through asterisk via a T1 using e&m directly connected to the Telrad system. We're using a T1 cross cable as normal, the T1 part works great. No alarms. When we try and dial out the Telrad using a direct trunk group, the call fails. When looking in the asterisk console I noticed only 1 or 2 digits are seen,

Hi All, I'm trying to configure a Digium T100P to talk to a legacy voicemail system. I have the signaling specs verbatim from the original manufacturer documentation as follows: [T1 Signaling] Service Type: T1,D4 format, AMI(Super Fram) Signaling: Four wire, terminated, E&M (Robbed bit) Start Protocol: Wink start; 250msec duration Dial Tone: Enabled Digits: DTMF, 4-digits DTMF: 50msec