similar to: zaptel DTMF delay

Are there any debugging tools for the digium zaptel cards that would report the activity on the line, such as DTMF and/or connection protocol? I'm looking to debug the connection with a T100P, & I don't have $2000 for a T1 test set. Thanks, Glen

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 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 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 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

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

I have a problem which I need to resolve. We are trying to put an asterisk between a Mitel PBX and the world. We are adding Voip service via Asterisk. Here is are config files for the settings but our problem is the following. We are able to send calls to the Mitel pbx and it's the T1 connections is green saying it's ok. The support department from Mitel said that they use e&M and

I have an nec electra elite 192 with a t1 card; and am looking for suggestions as to integrating them (can't throw out the system yet!). I have a fully working asterisk server -CVS-04/27/04-19:01:05- (found a hp d220 for $350.00!), 2 digium t100p cards, a plain t1 with loopstart signaling, and 2 working bt102 grandstream ip phones (thanks again Matt for your "start from scratch"

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 need to know how to setup the data side of the T1 on my Linux Box. I have found information about configuring a PRI and HDLC but nothing about the Frame-Relay type setup for data. The following is information from our T1 provider. Network T1: Framing = ESF Line code = B8ZS Build out = 0-133ft(DSX)/0dB(CSU) Clock = network Pulse-density-enforce = off alarm-option = on alarm-delay = 15

hi, i have some newbie questions about channel banks. i have an adtran act-1241 sitting around. it accepts D4 modules, and it contains a number of e&m cards. first of all, how does this thing work? a t1 contains 24 channels, and i noticed that the channel bank has space for 24 cards. what do these cards do? what are their outputs? the ones that are in there have some outputs on the front

When people call into my * box over the T1 interface, they get no ring tone. It rings the SIP phone and when the SIP user picks up, both parties can hear each other ok, its just the PSTN user calling in hears no ring. What could be causing this? I tried setting immediate to yes in zapata.conf, but that causes my DNIS and CallerID to stop being available. T100P with E & M Wink start

I'm getting red alarms with my T100P card that last from 4-15 seconds. They seem to happen randomly, every couple of days. Has anyone seen this behavior, or does anyone have any ideas regarding what would be causing it? Thanks, Glen

I have been experiencing a rather odd issue with my (newly connected) T100P card. It is installed in my Asterisk server, which is working great with SIP clients. The card is connected to a channelized T-1 line (24 channels). The odd bit is that outgoing calls (Asterisk --> T-1 --> PSTN) works just fine. However, incoming calls (PSTN --> T-1 --> Asterisk) are dropped after about 20

I need to create a couple of tie lines between a legacy system and an Asterisk system. I was told that the tie lines are E4 Superframe E&M. I have done E&M wink but have no idea about E4 Superframe E&M and Google is not helping me here. Does anyone know about this type of signaling and if Asterisk can handle it? Thanks, Steve

Dear all, I am a newcomer to R programming, I met the problem: I have a lot of .txt files in my directory. Firstly, I check whether the file satisfies the conditions: 1.empty 2.the "Rep" column of the file has no "useractivity_idle" or "useractivity_act" 3.even The "rep" has both of them, numbers of "useractivity_idle"==numbers of

I've posted this problem a couple of times before with little or no response. Basically I have a T100P in my * box. Incoming calls are working great. However outgoing calls are not working at all. I've copied a previous post into this message which should have all the necessary info. Any ideas or suggestions would be greatly appreciated. Thanks. Mark

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) ;

I have a Toshiba DK424 connected via T1 E&M to a TE110P card. Intermittently when a user dials a number I am getting 'getdtmf' errors on the Ast server and the calls do not go through. If they dial the number once or twice more, it works fine and I receive no DTMF problems. On another note, end users are complaining about intermittent disconnects. T1 is ESF/B8ZS - 24 chan. Other

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