similar to: Noise while passing channel using tde205p card

Hi, I am having problems with an Asterisk with a Digium TE205P card. The issue is that the Ring Back Tone is noisy. I am making modem's calls and this noise influences on the initial negotiation protocol, so modems have to recall. My configuration is: Asterisk version: Asterisk 1.4.21.2 Linux version: CentOS release 5.2 (Final) Card: Digium TE205P

I am having problem with T1 configuration. Following r the config.. Zaptel.conf # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order span = 1,1,0,esf,b8zs # fxsks=1-24 bchan = 1-23 dchan = 24 # Span 1:

Hi I am trying to connect asterisk with a Cisco GW 2600 with E1 pri using a Digium, Inc. Wildcard TE210P dual-span T1/E1/J1 card 3.3V (rev 02), Errors: [Jan 28 17:32:33] VERBOSE[6182] logger.c: == Primary D-Channel on span 1 up [Jan 28 17:32:33] WARNING[6182] chan_dahdi.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jan 28 17:32:34] NOTICE[6182]

Hi I have this escenario: |SIP or H323 phone|---->|Cisco2600|----E1-pri---->|Asterisk|------>IVR, A2Billing, etc... The problem is that I can not hear any audio when call from 'sip or H323 phone' and configure something like: exten => _01XXXXXXX,1,Playback(thank-you-for-calling|noanswer) ... It works if I remove the 'noanswer' parameter but in this case it connects

hi after some testing with A@H, i've decided to install my asterisk server on a slackware (because it's my favourite distro and it is still suggested here http://www.voip-info.org/wiki-Asterisk+Linux+Slackware) , so, i've installed the last 10.2 release, and i've recompiled the 2.6.15.4 kernel. then i've downloaded asterisk1.2.4 and zaptel1.2.4 i load module zaptel and

I've got an asterisk 1.4.24 box with dahdi complete 2.1.0.3+2.1.0.2 I've got a 2 port T2XXP card attached with on T1 currently plugged in. Inbound calls work fine, but outbound fail with the rather cryptic: [May 19 17:28:07] WARNING[11360]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) There are similar threads I've found on the

Hii, I've a Digium TE205P card and I'm running asterisk@home in a box.I want to configure this card in that box.But as it is a dual port card I'm not sure how to configure it in my box.By searching diffrent site I could able to make a sample configuration bellow I'm giving.Basically I want to connect one T1 connection in one port and in anather port of the card I'll

hi you, I'm having problem with voice quality on my trixbox using TDM2400B.The trixbox is connected via 20 FXO ports on a TDM2400 with the hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'background crackle/buzz' coming back when they talk. anyone have the same problem? pls help me. thanks a lot. my trixbox and config

Hello. I have a TE212 configured in E1 mode. This is shown in a cat /proc/zaptel/2 and 3 (where the card is configured): cat /proc/zaptel/2 Span 2: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" HDB3/CCS/CRC4 RED NOTOPEN 25 TE2/0/1/1 Clear 26 TE2/0/1/2 Clear 27 TE2/0/1/3 Clear 28 TE2/0/1/4 Clear 29 TE2/0/1/5 Clear 30 TE2/0/1/6 Clear

I'm using two Rhino channel banks (first 12FXO/12FXS, second 24FXS). These connect to a Digium TE210P card. I'm running kernel 2.6.10 and I've tried Asterisk (w/zaptel) 1.0.9, 1.2 beta, and CVS from today. The results are the same for all versions: Right after I reboot, and modprobe wct4xxp, my analog phone connected to port 13 of the first channel bank (first FXS port) gets a dial

the Telco, I can make calls in. What I am trying to get though is how to pass through the DID range. The E1 that I am connecting to the Telco with, used to connect direct to the NEC system and already has hunt group calling enabled for all 30 channels. Also, I was given a 100 number indial range (from 00 -> 99). If the E1 is connected to the NEC directly, I can call 5555 7320 and the NEC

Hello all. I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Ie, pick up NEC handset, dial

Hi. My configuration is Elastix 1.5.2-2 (asterisk 1.4.24, libpri-1.4.3-5, dahdi-2.1.0.4-7 ) and OpenVox d210e connected to telco provider (Euro ISDN). Here is my /etc/dahdi/system.conf: # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 # Span 2: TE2/0/2 "T2XXP

Hello list i have an issue with my dahdi_channels.conf in span 1 when i use it like below i can do my outband calls without issue ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel => 17-31 context = default group = 63 but when i add in channel 1-15 like: channel => 1-15,17-31 i receive all

Hello, I'm running Asterisk@home 2.5 asterisk 1.2.4 zapatel 1.2.2 libpri 1.2.2 on a Dell Poweredge 2850 (1 CPU) with a TE210P I have 2 t1 circuits using NFAS with dchan on 24 and no backup dchan. I am able to receive inbound calls on all channels and can only make outbound calls on channels 25-48. Attempting to make an outbound call on channels 1-23 results in congestion.

hello list i have installed 2 diguim card 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

# Span 1 span=1,1,0,esf,b8zs bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2 span=2,2,0,esf,b8zs bchan=25-47 dchan=48 echocanceller=mg2,25-47 # Span 3 span=3,3,0,esf,b8zs bchan=49-71 dchan=72 echocanceller=mg2,49-71 # Span 4 span=4,4,0,esf,b8zs bchan=73-95 dchan=96 echocanceller=mg2,73-95 # Global loadzone = us defaultzone = us Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1"

Hi, at work we have hundreds of machines, and for various of reasons, their hostnames (with domain) do not reflect their physical location. This means that for host "a.bb.cc" i have to go through jump host "jump1.bb.cc", and for "c.bb.cc" i have to go through jump host "jump2.bb.cc". which jump host should be used can be deduced by IP, but it looks that

Hi, I have problem using Asterisk.I have isdn-pri and openvox d110p card in my computer.They are connected with RJ-45 (1,2,4,5 pins to the card and all pins to the isdn done by telco workers). I got green led on isdn which is sign that isdn is working and that is connected to openvox, right ? I compiled newest versions of libpri zaptel and asterisk and had no problems during that. When I started

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