similar to: PRI Card

Hi I am not sure why this is happening or whether it has anything to do with my iaxmodem setup. When receiving a fax via iaxmodem, I got an error message saying *chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy*

Excuse me if this is in the archives or on the net somewhere - I really did search. ;-) We are wondering if there is any way to shut down a single PRI without having to down Asterisk and/or interrupt other running PRI circuits. We use Asterisk servers with 4 port Digium PRI cards. In the last few days we ran into a situation where the 3 of the 4 PRIs were operating fine and had live

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

Dear List: I have a data manipulation problem that I was unable to solve in R. I did it in SQL, and it may be that the solution in R is to do it in SQL, but I wondered if people could imagine a vector-based solution. Imagine a list A[i] of observers who observe some set of events B[j]. Each observer i may observe one or more events, and each event j may have been observed by one or more

Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P (same problem with various previous versions; same problem with different TE120P cards). The customer has a partial (10 B-Channel) PRI that when it is busy (eight or more B channels in use), tends to fail as shown below... [Jan 26 23:00:31] ERROR[31893] chan_zap.c: Write to 28 failed: Unknown error 500 [Jan 26 23:00:31]

Greetings- I'm wondering if the Digium PRI cards can be used for data (Cisco HDLC, PPP, etc) or if they're for voice circuits only. I haven't been able to find any information on this. All documentation direct from Digium seems to indicate their hardware is for voice applications only. Sangoma's cards work in either voice or data mode but of course this is configured in their

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

Pardon my ignorance, but I was hoping someone could clear up something for me. - For a few POTS lines, digium has a single port card for that, or a T1 card to a channel bank. - For 10 or more lines, digium has a T1 or E1 card for that too based on PRI channels - For 100's to 1000's of lines, I suspect a soft-switch is in order??? A traditional phone company will sell: - POTS lines for

Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance

Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with

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

Heh guys, I just received a T400 card, I've been using a T100 for a little while, and it works fine when using a raw channelized T1. I'm relocating my asterisk machine, and PRI's will only be available, haven't found any good config info for PRI's, can someone point me to PRI config info, or let me know what changes I need to make in order to bring them up, I imagine,

I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP,

We are in the process of ordering a Voice PRI to plug into Asterisk. Of course we will be buying a card from Digium for this. Question is this, there seem to be MANY options technically when ordering this PRI (in the US) but since this is the first time ordering a voice circuit I am clueless as to what options we need. Any clues would be helpful or maybe something has already been written

Hi, I have successfully configured DIGIUM card and successfully communicated through it to the another E1 card running application. Can anybody tell me does TE120P support MFC/R2 protocol. Thanks and Regards sanchal singh

I'm trying to get a Qwest PRI configured and working with my lab Asterisk server. They said that the switchtype is 5ess and the signaling is pri_cpe. My entries into zaptel.conf are: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us channels=1-23 And my entries in zapata.conf are: language=en context=telco-incoming switchtype=5ess signalling=pri_cpe

> The documentation mentions that the Digium channels can be split into some > voice channels and the remainder of the channels used for routing IP > traffic. > > Does any one have this in use in conjunction with Asterisk? Does it work > well? Would you recommend it for a production server? > > Obviously, if this works, this makes for a cost effective platform where

Hi All, I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its connected directly to the PSTN. But I am unable to make outbound calls on the zap channels. The light on the card is green. Asterisk CLI shows all 24 channels when I give the command 'zap show channels'. I also noticed that Asterisk CLI shows an incoming call every few seconds on the 24th channel. This must be

Would anyone care to recommend a T1 interface method for Asterisk that would function as an (external) alternative to a PCI card like the Digium TE120P? Like some sort of T1-SIP gateway? Also, would anyone with experience using these products care to comment on the practical value of the TE207P vs. the TE205P? -------------- next part -------------- A non-text attachment was scrubbed... Name:

Hello everyone, I'm hoping someone can help me with this. I have a business customer in the U.S. (Michigan, AT&T Territory). I need to get 4 trunks into an asterisk Box. My intention is to use an Eicon Diva Server card with 2 BRI Circuits. The reason for this is that the business needs DID's on the trunks (20 of them). A full or fractional PRI is overboard for them, as they will