similar to: No sound on 10% of incoming calls

Hi all, Does somebody know how I have to setup my TE405P ? Is it correct my configuration below ? Otherwise, can somebody help me ? Thanks, Angel. zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,1,ccs,hdb3 span=3,0,1,ccs,hdb3 span=4,0,1,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31

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)

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

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

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

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

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 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 pretty large setup on one of my customers. Digium TE420B (with echo cancelling module), 3 Xorcom Astribanks with 32 FXS each and 1 Xorcom Astribank with 16 FXO. These FXO ports are NOT used for fax/data transmission, as they are connected to cell phones. Not really related to the issue, but there are also 250 SIP phones. The problem is that fax and dial-up connections are really

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

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

-----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! 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

Can't anyone stop self-promotion and tell the poor guy what he needs. A T1/E1 X-over cable using an RJ-45 (8-cond.) is pinned out as follows: 1 - 4 2 - 5 3 - NU 4 - 1 5 - 2 6 - NU 7 - NU 8 - NU NU = Not Used I have not in my experience seen any problems with using a Good Quality Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you should be fine. As far as the shielding

Hi All, I am facing problem with E1 line. I have installed Asterisk (1.4.20.1) on a system with Digium TE420 card (Zaptel- 1.4.10) But every now and then I face problem of down E1's. The log show lot of entries like "pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2" This happens on a regular basis and the E1 becomes up after some time. My

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

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'.

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

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