similar to: Question about TE405P

hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI of our Ericsson BP250. the setup before: Arcor TelCo PRI(E1)

I am trying to install a TE405P which is to be connected to E1 trunks. I have set the jumpers to E1 (closed), but modprobe wct4xxp still fails with the message: ZT_CHANCONFIG failed for channel 97 - which obviously means the card is still in T1 mode. I have tried everything I can think of, any ideas? TIA

We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p EuroISDN. We have 2 ISDN lines, one we had for testing, and one for general (40+ users) use. During the testing phase, we had 10 phones linked to the second ISDN line, and there were no problems with echo at all. Lucky me. However, since we have started rolling out, we've had quite loud complaints that there is

Hi, I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02)) 4 ports I want to make a loop test between digium card E1 to test the configuration of dahdi What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall-->E1 Digium port 1--->(Loop)E1 port 2---->sip extension local. kindly can any can help me to

Hello! I am trying to install a TN405P on a P4-3GHz-HT machine running Debian Sarge with kernel 2.4.27. When I start Asterisk in -vvvvc mode it always shows == D-Channel on span 1 up == Restart on requested on entire span 1 == D-Channel on span 3 up == D-Channel on span 2 up == Restart on requested on entire span 3 == Restart on requested on entire span 2 == D-Channel on span 4 up == Restart

Dear, I always had one E1 card with one span, so I've never had any problem in set it up through /etc/dahdi/sustem.conf and /etc/asterisk/chan_dahdi.conf because I put span=1. But now I have a PBX with two E1 cards with 4 span (8 span in total). How do I have to define both card in system.conf and chan_dahdi.conf, and how do I have to refer each span to the corresponding card ??? Thanks a

Hi all, i am stuck with the configuration of asterisk - modules are loaded ( zaptel and wct4xxp ) - i have zaptel.conf configure, output of ztcfg -vv --- snip -- rapid:~# ztcfg -vv Zaptel Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet

We have a te410p, with the following connections: span 1 connected to a 32 Channel EuroISDN span 2 connected to a card in a legacy pbx (Meridian) span 3 connected to a 10 Channel EuroISDN span 4 connected to a card in a legacy pbx (Meridian) We have no need for the meridian now, and decided to turn it off. I did not change the zaptel.conf settings, nor the zapata.conf settings. When the

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

Hello, I have a problem with Asterisk, on 10% of incoming calls the IP Phone ring but I don't hear the caller and the caller doesn't hear me (all IP Phones have the same problem). This problem appear also if the call is directly send to the second E1 of the digium card who is connected to an IVR. It does not depand on the charge of the server (I have the problem with only one call).

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello List, since some days i run into the problem that one span on a TE407P is not comming up correctly. With intense debug on that span i get: < [ 02 01 7f ] < Unnumbered frame: < SAPI: 00 C/R: 1 EA: 0 < TEI: 000 EA: 1 < M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] < 0 bytes of data

Hello, There are 3 PRI's connected to the card each from different operators. Especially echo occured on span 3 is really annoying. Configuration files are as follows. Is there something wrong in conf ? Zapata.conf -------------------------- [channels] context=default switchtype=euroisdn usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes

hello list, i have installed 2 diguim cards in my server using asterisk 1.4 (i use the old version with zapata.conf and zaptel.conf) i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i want to active the round-robin for span 2 and 6) in order to activate the WIMAX and FH please see the configuration below and tell me if there is anything wrong question 2: what is

hi, I'm trying to configure my newly acquired TE410P card to work as 4 E1 spans. This is supposed to be a drop-in replacement to the earlier E100P card. However, on loading the zaptel module it gets configured as T1 spans basically doing a 'cat' on /proc/zaptel/1 thru 4, it shows 24 channels per span. After this ztcfg fails saying 'ZT_CHANCONFIG failed for channel 97'.

Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT <-> Swyx The above setup works fine... what i'm trying to achieve is BT & SIP Trunks <-> Asterisk <-> Swyx I have connected to our BT (2 x ISDN30 UK) with

I have a 'CONGESTION' Status with R2 protocol. While testing this scenario sip GW--?Asterisk ?Digium E1 R2 Protocol?Cisco E1 R2 protocol?sip Gw Find below my error and configuration ,where are the errors in my configuration ? ========================================================================= Connected to Asterisk SVN-branch-1.6.2-r235775 currently running on

Ok Guys, I've done a tonne of hunting around on this problem, but can't find much help. I'm running: asterisk 1.4.19.1 libpri 1.4.3 and zaptel 1.4.9.2 which I believe has been modified by RedFone to add the ztd-ethmf module. My interface is a RedFone foneBridge2 4 Span; and I'm connecting to a BT E1 PRI / ISDN30e with 15 lines on span 1, and a legacy Panasonic PBX on span 4. Upon

I have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S). Does anyone have a valid XMLDefault.cnf.xml they could share? I have tried the version at voip-info<info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP&view_comment_id=14768#Troubleshooting>for the 7941/7961 but unfortunately /var/log/messages shows in.tftp stops sending after

Hi all, I still haven't seen my first post (cry for help) so I'm not sure if these are going through or not ... I've played with the box all day today ... I got it to quit seg faulting finally by playing with the config files. Currently I have 4 spans configured for a Digium TE420B (With Octastic DSP) Here is my zaptel.conf: (I know the device connected to spans 1-3 are providing

Hello all, I really checked voip-info.org but it still seems to be not very easy and I just hope that there is anybody with a simular config. We have one PRI (euroisdn with 30 channels) coming from the DTAG. The second PRI should be connected to a Siemens Hicom 150E Pro Office PBX (was cheaper than a channel bank :-) Carrier ----S2M------ * -----S2M------Siemens | |