similar to: Details on TE410P Digium cards

Hi all, I replaced a TE410P Rev C 1st Generation Firmware with a TE411P without any cfg changes (zaptel/zapata). As a result Asterisk crashes on outbound from PRI4 going to PRI1 calls: Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1

I am the proud new owner of a TE410P, and installed it on a RedHat 9 box. After compiling just fine, running like a champ in tests, and having my extensions.conf configured to taste, I went ahead and did a live beta test this past weekend. The phone system stopped responding 3 times. The first time, I got into the asterisk console, vi -rvvvvv. I didn't see anything too out of the ordinary,

OK. Now I show my ignorance. What's the difference between a T-1 and a channelised T-1? I see that Covad's voip service (formerly GoBeam) requires a channelised T-1. Then I read recently on the list that many T-1s being installed are actually HDSL....which would be not a T-1 at all...right? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

According to digium's site, "Note: Please do not attempt to use the G.729 code in a SCSI-only system. We are currently working with VoiceAge to correct this issue." (found at http://www.digium.com/index.php?menu=asterisk_g729). Does anyone know what these issues are? Can anyone define SCSI-only system? I know this sounds kinda dumb, but I have a server with SCSI and IDE interfaces,

Has anyone else experienced a problem with the LED for span 1 on a TE410P or TE405P? I had a TE410P on which the span 1 LED would not light red, but once the span was connected, it did correctly light green. I RMAed the board to our UK distrbutor and received a replacement. However, the replacement board displayed the same problem! Wondering if it was related to the computer I was putting it

I'm experimenting (using for the first time) with using a cisco3620 to connect to the PSTN via a channelised E1 interface, with * handling all of the SIP calls. If anyone has any installation tips / help / documentation I would be most appreciative :) However, my first question is this: when I am in the setup, I see the following: Current interface summary Controller Timeslots

Hi, one short question: Is it possible for the zaptel driver to deal with multiple phone numbers on one single E1 PRI line? I could make my carrier route +49 xxx aaaaa-zzz and +49 xxx bbbbb-zzz and others down one single PRI trunk to our asterisk box terminating in a Digium TE410P. Does the driver handle this and can I put calls coming in all on the same physical interface put into

Hi, I'm in the need to use a line extender. I found a device, which should best be placed in the middle of the line, which would make power supplying difficult, unless I'd use its "telco power supply" features: The device is capable of taking current from one signal wire pair or voltage from the potential difference between both pairs. Both methods seems to be a sort of

Hi all, I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns QuadBRI running in NT-mode. Connected to the BRI-Ports are 12 Fax-Modems (Elsa MicroLink ISDN/TL V.34) which are only operating in dial out analog mode to deliver fax messages. After a while of running fine (50-200 dial out connections) on some S0 spans the following message occurs

Hi all, I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P. Box A is connected with pri1 to the PSTN. Box B is connected with pri1 (cpe) to the Box A at pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only

Hi We interfaced our ASCOTEL PBX to Asterisk. by EuroISDN PRI , DSS1 It works fine on channels 1- 15, but on 17-31 the miststood each other. Asterisk speaks in Timeslots, the PBX in B-channels The signalling is ok, but the bridging is shifted. The first incoming connection is bridged to "nirwana" also no indication is hearable, calling a second internal subcribes bridges them to the

Hello all, I have some questions about the libpri and zap lib. I've sent mail to Mark for help. Also I wonder whether anyone else would be interested and helpful in them. I'm going to migrate a program to Wildcard E400P card. Our original program is based on C API similar to the functions provided in libpri.h and zap.h, so I want to use the E400P card in the following way: 1. install and

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

Hello ! I 've some questions how bridging of ISDN calls is done. Assume an asterisk system with a TE405 card equipped. (PRI1 - PRI4) An incoming ISDN call on PRI1 is transfered back to PRI3. Unless there is DTMF detection or other things involved, the bridging is done without Asterisk. Does this card have a some sort of cross connection ? Does the PCM leave the card ? Or is there some DMA

I am looking for a large number (probably about 100 or so) low-cost phones that I can hang on the wall. I need the phones to use PoE. Do the Uniden phones support wall-mounting? These phones are not going to be high-usage; they simply need to be there in case of an emergency. Another question, along the same kind of lines, has anyone figured out how to keep the SoundPoint IP 600 receiver

Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from

I am currently working on an Asterisk test system, and will be presenting a demo to the Board of Directors tomorrow night. I want to make sure I have all of my ducks in a row. The Asterisk system will be used to replace a Norstar MICS. The location has two PRI's coming in, with a few hundred DIDs. I know how to make * use the DIDs incoming, and I know how Nortel uses the DIDs. Now for the

Hi, I have an asterisk installation with 2 E1 cards Software version is Asterisk 1.2.6 Libpri 1.2.2 Zaptel 1.2.5 I'm having problem with fax transmission, let me explain better my setup: My fist TE110P E1 card is connected to the telco line the second TE110P E1 one to an Nexspan PBX so the server is basically sitting between the line, and the pbx. every call coming from the line is

Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1 # Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kan?le sle=$4 # Timeout bis zum n?chsten Versuch if [ -z $4 ]; then sle=0 fi s=1

Thinking outside the box, four possible solutions come to mind: 1) use ssh (preferred for security), vnc, or xwindows to remotely access the machine (I've heard it is not good to use vnc and xwindows as the GUI tends to interfere with Asterisk timing) 2) in our server room, we have a keyboard and monitor on wheels and connect up to any server when it requires local servicing (or use a KVM to