similar to: Yellow Alarm issues with second TE410P installed

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.

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 encountered this strange problem which is I can call into Asterisk box but I cannot call out. I was successful before using exactly the same euroISDN line but with TE110 and different versions of Asterisk. This time, I am using: . TE410 . Asterisk 1.4.13 . Zaptel 1.4.6 . Libpri 1.4.2 1) I put the following into extensions.conf to get to the outside line exten => 0,1,Dial(Zap/1) 2)

I am trying to test the T1 card in our legacy PBX but the connector to the card is a 15 pin serial cable. I would like to make it myself so I can try this test today. Does anyone have a pinout for it? I just made a T1 RJ-45 loop back to test my TE110P and it tested out fine. I'm trying to resolve some timing issues and a tech from Digium is going to ssh in and walk around my asterisk server

We have recently started routing about 3 PRI's worth of traffic thru our asterisk box. The text on the console now flys by so damn fast, I can't really see what the heck is going on. Even with verbosity 0 and debug 0 it is still so fast. Is there some way I can attach to the console in a way that will allow me to grep or otherwise filter the text so I can focus on something in

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

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

Hi, Is there anybody in this list who had experience with two TE410 cards on a server ? I know that the cards can?t share IRQs and I?m seeing to have two cards on a x335 IBM Xeon server. TIA -- Juanjo sin .sig

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

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040810/16371a21/attachment.htm -------------- next part -------------- ? Hi, I'm using TE410P card with four T1 lines. I've configured all the channels in my /etc/zaptel.conf file. In zttool i'm getting "OK" for the "Span-1" but the other three spans giving

Greetings... I have a TE410P with four T1's going into it. Things look roughly like this: #1 Goes to PBX -- we're responsible for timing #2 E&M span to telco 1 #3 PRI span to telco 1 #4 PRI span to telco 2 If I set primary sync source to span 2, users report strange echo, distortion, and crosstalk problems, which sound remarkably like frame slippage on spans 3 and 4. If I set

Basically, have an old IVR application running under Apex's Omnivox software on a box with 4 old intel dialogic D240SCs, and would like to allow remote clients to gain access to aforementioned IVR application via softphone, 7960, ata, etc via asterisk with a TE410P. Unfortunately, all I know about the dialogics is that they're configured for D4/AMI, which I believe I have configured right

I am having a problem with asterisk not being stable enough for production use. I have two cards, the digium TE406P, and the TE410P. The TE410P is the primary card that I am using but I would like to move to the TE406P for the echo cancellation and more flexibility of PCI slots available. General config info: 3 PRIs, configured as such: span=1,2,0,esf,b8zs span=2,0,0,esf,b8zs

Hi, I've been asked to find a server that is capable of running the Digium TE410P card. We usually get Dell PE servers and after a quick look I think the PE 1800 has the required slot: Six Total: 2 PCI Express (x8 lane & x4 lane); 2 x 64-bit/100MHz PCI-X; 1 x 32-bit/33MHz PCI (5v) and 1 x 64-bit/66MHz PCI Has anyone got this hardware, does it work with the card and asterisk? Thanks,

We have a Te410P with 3 Telco T1's (D4 SF ) with DID's (non-PRI). ANI & DNIS is received in-band DTMF in a format such as *7145551212*8002* What happens when there are 30 or more calls, asterisk might see is DNIS = 802 or ANI = 4551212 for examples, where parts of the numbers are dropped. All the traffic arrives into a simple IVR script where a message is played. We are

i get this message with a red signal on TE110P card: ***************** TE110P: span configured for... Calling startuo (flug is 4099) wcte1xxp: Setting yellow alarm ***************** what does mean ? thank you :)

Hi, I have this strange problem I need some help with.. It appears that I have harddisk noise captured by a Digium TE410P card (Same problem on 2 identical machines..) The machines are two Compaq Proliant DL320 G3's... Does anyone else have this problem.. Kind Regards Claus Futtrup --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com).

How would you categorize a Yellow Alarm sensed by the Asterisk side in a Legacy PBX integration? We have a Mitel SX200 connected to an Asterisk(1.2.4) with a TE110P. Twice today (first time in over a month) we received a Yellow Alarm on the TE110P. I have been able to clear it easily by restarting zaptel. Thanks in advance! -------------- next part -------------- An HTML attachment was

Greetings, According to the FAQ "When you load the module and have no circuit/channel bank the LED's should flash red" I get the "knight rider" lights before the module loads, but after the modules are loaded I don't get any lights, other equipment plugged in or not. The card has been on my shelf a while so I wonder if it needs a firmware upgrade ? Can the cards

Folks, I really hope you can help me here - I'm beginning to tear my hair out! About 10 days ago my company moved to a new office. As a result of this, we've plugged our PBX box, which has happily been running for the last three years, into our new E1 line. Since then, I've been seeing intermittent yellow alarms. Obviously, since this was working fine in the old office, the thing to