similar to: Dialing long-distance locally

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

A thought occurred to me to on how to further quantify the impact of "glare" on a properly dimensioned trunk group and debunk the ground start glare concern. A cursory traffic analysis clarifies: 1. Assume you have a two-way trunk group, dimensioned for average busy hour, average busy season for P01 grade of service, Erlang B or Poison distribution. 2. Further assume that the

I'm a bit confused about how to handle this. I have Asterisk sitting in the middle between a Qwest Long Distance T1 (Voice T1, D4, SF, AMI) and an external voice mail PC using a Dialogic D/240SC-T1 card. The Qwest T1 originally was connected to the Dialogic card directly. The signaling was set to E&M Wink Start because Dialogic used this as its default settings, so it just worked

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both 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

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

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.

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

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

Ok, I'm about to take the plunge, and am trying to decide between Channelized T1 E&M and PRI. I'm getting an "Integrated T1" which will have data and voice capability, all plugged directly into my digium single T1 card. In either case the data piece looks pretty straighforward, just setup the channel properly, hand it off to the linux hdlc layer, and route away.... the

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

Hello, is there a full guide to what kewlstart is supposed to do with FXO or FXS lines ? is it only applicable to one of the interfaces FXO -or- FXS but not both ? I asked earlier if FXS lines can be made to reverse polarity, and someone else pointed out that the chipset on the FXS ports seems to support it, perhaps the driver in the asterisk zaptel interface module needs to be modified to support

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

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

Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup E&M in zaptel.conf and EM_W in zapata.conf. They

I'm trying to get this zap dial to work. I want to send DNIS and ANI to other system (ZAP/g2) at answer, while the caller hears ring (RBT). I hear the RBT, but callerid and exten is not sent to other T1 - The ZAP/g2 T1 is standard D4, SF, E&M Wink Start. - At ZAP/g2 wink, asterisk should send DTMF "*ANI*DNIS*" exten => _XXXX,1,NoOp,${CALLERID} exten =>

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

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) ;

We have 2 T1's coming from our phone switch to a digium TE220B. We have managed to get CPN and the extension outpulsed from the switch, but call setups are really slow. Our T1's are set up as E&M Wink, and they send us the last 5 digits dialed followed by the 10 digit calling party number (we couldn't get the switch to be happy with *CPN*+5* to use featd). We are using asterisk