similar to: Definity <-> Asterisk w/callerid

I'm not a big fan of Line Side for Asterisk applications.... the individual channels have to be reserved (reminds me of the "old" days in telephony) and I don't like that... rather make it a tie trunk and use ARS/AAR routing in the Definity... much more general application. This also makes routing on the Asterisk a lot simpler. Easy to make the changes.... will advise if

Hi, I have two t100p cards installed. One card connected to the PSTN, the other card is hooked to the Definity PBX with a T1 cross over cable. The card connected to the PSTN works fine. The problem is with the card hooked up to the PBX. The light behind the card is green, but when I dial from this card I get an error. Its says it could start the D-Channel. Could someone help me out with this

Hello Group I have a Asterisk server running with 2 Digium T1 cards installed. 1 card connects to Telco via a PRI. The 2nd card is connected to a fax server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to have Asterisk route the calls based on DID or FAX tones. Everything is working great so far. The only problem is the Fax server does not see the DID. How can I tell if Asterisk

I don't see that anywhere. Here is my zapata.conf This is only happing on my 7970 all other phone are working without trouble. [channels] context=pri signalling=pri_cpe switchtype=dms100 group=1 usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=no useincomingcalleridonzaptransfer=yes callerid=asreceived faxdetect=incoming musiconhold=default echocancel=yes

Hi Srs. we have to integrate a Nortel MATRA M6501-L with Asterisk with a TE410P. All call from outside get into asterisk and asterisk send to Nortel in a correct way. My problem is when a call is made from Nortel to Asterisk. If we digit a national Number in Spain([98]ZXXXXXXX or 6XXXXXXXX) all work find. But if we digit an international number call doesn't progress. I Have seen in

Hey everybody, I'm looking for some information that I'm not finding by searching the 2004 Asterisk archives. I'm currently playing with a Digium T100P card and 2 Grandstream phones, things are working well. I wanted to move on to linking our Definity G3R Rev 8.2 to the T100P. Everything that I've read so far shows that you need a TN464 to accomplish this. We have a TN767E

Deat all, I am in middle of integrate Asterisk with Toshiba astrata legacy pbx. Following is my setup *Asterisk <-> Digium TE110P <-> E1 card in toshiba pbx <-> Toshiba PBX* A =============================================> B C <============================================ D Asterisk PBX and strata PBX connected using back to back E1 cross cable. Physicall connectivity

I just managed to put in a TE410 card in an Asterisk box to work with OnRamp 20(E1 downunder). I am able to dial in but was not able to dial out. Can anyone offer me some advice please? In my extensions.conf, I just put in: [default] ... exten => 0,1,Dial(Zap/g1) and I get this on the console when I dialled 0. -- Executing [0 at default:1] Dial("SIP/5166-b76004f8",

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

Hi *, now for a long time i am trying to set the outgoing callerid, without luck. I am here in Germany, my asterisk has a pri interface connected to a PMX installed by Telekom. All telephone calls are preselected to EcoVoice. I am using asterisk 1.2.7.1, zaptel 1.2.5 and libpri 1.2.2. A week ago we tried with a device able to simulate a telephone system so send out a callerid, and that

Is it possible to send CLID NAME on a PRI? The numbers we send out are being received by telco and propagated, but the names we send out are not showing up. Is this a feature in PRI? Do we need to set PRI_NET instead of PRI_CPE? Is this just not possible? Is this a telco config issue? Thanks for your help... I've read voip-info, and various other sources, and search engines, and google...

Calling all Definity admins, Got a few questions about Definity -> Asterisk interoperability. 1) What are the options for integration? Can I hand off extensions from the Definity and vice versa? 2) Anybody have any working configs they would like to post? I've found and read the legacy integration on the wiki about the two systems. I've also googled and found a few threads that were

Asterisk Version: 1.2.9.1 Zaptel Version: 1.2.6 LibPri Version: 1.2.3 Hey List, we are running an asterisk server in connection with an octopus telephone system. I have expired some random drops of zap channels bridged to SIP Telefones ( snom 190 ). Asterisk Messages shows something like that: Jun 9 09:32:33 WARNING[3207] res_features.c: Bridge failed on channels SIP/fon01-6945 and Zap/32-1

Hello! I use asterisk with TE420 as PRI switch for two channels : ;panasonic uplink group=3 context=panasuplink ; relaxdtmf=yes ; immediate=yes rxgain=0.0 txgain=0.0 mohsuggest=default jbenable = no ; jbenable = yes ; jbmaxsize = 200 ; display_send=name_initial display_send=name

Hello All I'm using asterisk 1.1.X and MFCR2 lib version 0.03pre2. when i call to E1 (connected with asterisk), chan_unicall don't detected event incoming call and show error. error messages: *CLI> Warning, flexibel rate not heavily tested! Rx CAS bits 0x9 [ 10000/ 0/ 0] Line unblocked -- R2 Channel 4 unblocked Rx CAS bits 0x9 [ 10000/ 0/ 0] Line unblocked -- R2

> > > Hi, > > > I checked by changing to from-zaptel, but no luck yet. Pls guide me on > this. > > Regards, > vudura senadeera > > > ------------------------------ > > > > Message: 9 > > Date: Fri, 19 Jan 2007 16:47:18 -0000 > > From: "Robert Jenkins" < raj@jrw.co.uk> > > Subject: RE: [asterisk-users]

Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2 I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf. The detection is not working with call file, manager originate and not with the dial command to the mobile. I have no ideas left. I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...) But with the same vaules on a second call there

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

Hi everyone, We have been working with Asterisk for a while now and would really like to expand its capabilities by fully integrating it with our Definity g3si and are wondering about other peoples experiences with similar setups. Thus far we have only been able to achieve a partial 1 way integration, but ultimately would love to route inbound voip calls to the asterisk out of the lucent.

Is anybody currently running a Definity G3 connected to Asterisk via PRI and a TN464? I've upgraded today and it would appear that caller id info coming across the PRI from the Definity is no longer showing up on the SIP phones. Caller id info from the SIP phones to the Definity is working fine. I'm going to be testing previous versions of Zaptel and Libpri to find the culprit and