similar to: E1 Card recommendation

Peter - 40 phones and only 3 PSTN trunks?. I would recommend at least 2 BRIs for this. If you have ISDN you can also get DDI to the extensions.I would strongly recommend abandoning the analogue PSTN lines and using ISDN. The setup pain you will go through will be significantly less, combined with better audio, more features (like DDI numbers!) and much more robust connections. You should look

I have gone through some archive about Nortel MICS (Meridian ?)+ Asterisk Integration but I'm not sure whether same as my case . 70 telephone sets | | Nortel MICS 4.1 --------- Asterisk | PSTN I have read the David Gomillion's Guide and got the idea . However, my plan is slightly different from what he did , I need to use Nortel MICS to connect to PSTN (I

I'd be grateful if anyone here could comment knowledgeably on an idea that I have had, as to whether it should be possible or not. Consider two Asterisk boxes, each with one or more E1s on EuroISDN. Each box has a different telephone number that hunts across all its E1 channels. In addition there is another number that hunts across all the channels on all the boxes. A call comes in to one of

Hi all I have a server that has two TE412P (T1/E1+DSP) cards installed. One of them configured as an E1 PRI connected to PSTN and another one configured as a T1 E&M connected to Avaya PBX. Each card only uses two ports, so there are 2 E1 lines and 2 T1 lines connecting to this server. The purpose of this server is as a TDM trunk gateway that gets call from E1/T1 and then forward to an IP-PBX

Hi everybody. I'm about to ask a newbie question, be warned! I have a NEC 2000 IPS PBX connected to a E1. Now I want to set up an asterisk, with some digium card connected to that E1. (suggestions about the card? I'll have maybe another E1 more). The "newbie question" is.. How can I connect the asterisk PC with the PBX? I don't know so much about E1, so I don't know

Hi, I bought a Digium Quad E1 card model TE406P. Till now, I can't make it work... I mean, I have red alarm when I configure one E1. The provider is in France (France T?l?com) and I use the following zaptel config : span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 I'm using Linux 2.6.15 and when I run ztcfg -vvvv, it seems that all channels are configured... So can someone

hi list, how can i organize several pcs installed with asterisk and e1 cards to be seen from an asterisk server as one? so if there is a voip call that needs to be forwarded towards the pstn the asterisk server should find a pc that has free channel on it's e1 cards that is connected to the pstn side. /-- Pc1 - E1-\ VOIP -> Asterisk--- Pc2 - E1--PBX->PSTN

Hi List, I am from Peru, I have installed an asterisk server in my company with digium card E1 TE120P, I am having issues when i make calls, here the error from my server [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No channel type registered for 'Zap' [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause

Hola! I've a Digium E1 card, zaptel recognizes it as (loading the module with debug=1) Setting up DMA (write/read = 321b2000/321b2200) Controller version: 43 Framer: DS21554, Revision: 5 (E1) Found a Wildcard: Digium Wildcard E100P E1/PRA Using CCS/HDB3 coding/framing with CRC4 120 Ohms Calling startup (flags is 4099) Started DMA Got interrupt: 0x0005 /etc/zaptel.conf:

People, I have an Asterisk SIP server and I have to connect my voip users to the PSTN via a E1 trunk. I have two ways to do this: 1) Connect a E1 interface to my Asterisk (is it possible ???) and then connect my Asterisk to the PSTN 2) Buy a commercial IP PBX with a E1 interface, generating the SIP users again, and connect the IP PBX to the PSTN What do you recommend to me based on your

Hello, I have something like this: SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN After calling from SIP to PSTN (and from PSTN to SIP too) I can't hear anything only in my SIPUSER. At the PSTN side everything is OK. I have another network with another h323/sip (in the place of asterisk) and there everything is OK. In AUDIOCODES logs I see that everything goes

Hi, I've been debuging the call disconnection problem in our architecture: PSTN---E1---OldPBX---E1---Asterisk This is our problem: -SIP user agent "A" calls a pstn phone "B". -"B" hangs up the call. -SIP user agent "A" starts listenning busytones... But the call still on. (and being payed). - Call only ends when it is correctly hanged up in the

hi, after several problems getting the right callerid on a E1-PRI there is (so far) only one problem left: when receiving calls over the telephone network from another E1-PRI that has a "Caller ID no screen" capability (e.g. a bank and a customer of us), asterisk does not get the callerid that is set up by the calling PBX, but the callerid of the trunk of the calling PRI. no matter

Hi all I am beginning in asterisk and am making tests with an ata-186. For the time being the tests are going well, however have a doubt. I am thinking about using a canal e1 with plate digium. Assuming that the company of telecommunications supplies e1 with 30 canals and numeration to me 4000-0001 4000-0029. she is possible to configure asterisk in way that somebody of is dials 4000-0025, to

Hi, A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN trought another E1. When the legacy user dial to the PSTN the call pass trought Asterisk. All works OK, the only problem is the delay on the Asterisk server when it receives the digits from the 1st E1 link. It will only make the call when the digit timeout expires. Is there a way to make something like

At Sangoma we do quite a lot of back-to back T1 and E1 connections. T1 is not a very fussy connection, as the baud rate is only about 750 kbps. In our experience, for error free communications you can use the following rules of thumb: Up to 50 ft: Flat patch cable Up to 500 ft: Ordinary twisted telephone cable Cat 5 may be overkill unless you are going hundreds of feet. David Yat Sin

Hello List, I just have a query.... is it possible to have 2 or more telephone number mapped to the same E1 line and if so will the TE120P card pick up the last 4 digits of each number - as it is currently doing for the one? -- Kind Regards Etienne Pretorius -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, (Resending this email due to a typo in previous copy) I need to do E1 to T1 conversion for a project, and was wondering if there exists a card with both E1 and T1 on it. Or is it possible to use two separate cards in an asterisk box, one for E1 and one for T1? (Please don't mention aculab or adtran, dealt with them in the past, won't deal again.) I talked to Digium and the

Dear all I have setup with mediant 2000 with avaya now i want to install E1/PRI card with asterisk and trunk with E1 with Avaya E1 port so i want to buy E1 card for asterisk so which card is best and cast effective for my setup i want 1 port E1 card so can you suggest me which card is best for my setup and i want QSIG singaling with avaya Regards satish patel

Hi, I've looked around the archives and have spent a while on voip-info.org but not found an answer so forgive me if this is in a FAQ somewhere. We've got several Asterisk servers with E1 cards (some Digium, some Sangoma). We provide non geographic numbers for customers and route calls to their existing phone numbers. Calls come in over the PSTN and into Asterisk. This works perfectly