similar to: Timing Slips CRC & E-Bit Errors - Asterisk - Trixbox 2.8.0.4

# 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"

Dear All; I am afraid from IRQ misses: 1 The ISDN E1 was working fine on the machine, the electrical disconnected and then the Red Allarm. I checked the dahdi and I found that I have to reinstall dahdi again and I did. But still not becoming UP. The output of the cat /proc/dahdi/1 is following (I am afraid from the IRQ misses: 1, so if it is a problem what is the solution)? [root at CC

Hi, Running 1.4.26.1 here. I have installed TE420B card in my server, and followed the appropriate steps (as far as I know to configure it). This TE420B is connected to a CLEC (T1s), so I am using pri_cpe as singalling type. When I dial out, I get this message: Dec 4 11:37:31] WARNING[27983]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0

I keep getting a red alarm when trying to setup asterisk to use my TE420B EC. I only have a blank context setup in my extensions.conf as I haven't started to config that until I can clear this red alarm. I don't have physical access to the server, so I can't go reseat the modules/card/ethernet cable, though I have hands on location that have done this a couple times

I am having problems getting a Nicherons TDM400P wildcard clone to dial out. Everything appears to be configured correctly, but although I see call progress, it never seems to actually pick up the phone. (The following is a test of 911 emergency, where I substitute 811 [repair service] as the actual number dialed.) *CLI> -- Executing [911 at from-internal:1]

The card you installed has FXO or FXS modules in it ????? are you getting your lines directly from the telco co??? Doug D On Mon 23/08/10 8:37 AM , Cassius Smith cassius at cassius.org sent: * -----Original Message----- * From: Todd Reese * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion * To: asterisk-users at lists.digium.com [3] * Subject: [asterisk-users] Dahdi

Hi, On a Asterisk 1.8.12 system working OK for months (>100k calls proceed), users are complaining for bad audio. My setup is: PSTN <--E1/PRI ---> Asterisk <--- E1/PRI---> Siemens HiPath <---E1/PRI ---> PSTN asterisk -rx "dahdi show version" DAHDI Version: SVN-trunk-r10414 Echo Canceller: HWEC asterisk -rx "pri show version" libpri version: 1.4.12 A

Hello, My first attempt to get dahdi running on 1.4.28... with a Rhino 8 port modular card and a single FXS module. Got the Rhino card installed and the machine sees it: root at pbx:/etc/dahdi# dmesg | grep rcbfx [ 71.985309] rcbfx 0000:04:00.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21 [ 71.985440] rcbfx 1: Rhino PCI BAR0 50100000 IOMem mapped at ffffc90008d7c000 [ 71.985504]

[18884732963 at from-fax-machine:... - your call is hitting the from-fax-machine context - yet your 'fax' exten is in the from-pstn-4 context. See the "[2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c: Fax detected, but no fax extension" line. When Asterisk detects an incoming fax tone - it tries to automagically route the call to the 'fax' extension in the SAME

Hello I'd like to configure Asterisk so that... 1. I ring it from my cellphone with CID number displayed, just to notify Asterisk that I wish to make a call 2. Asterisk waits until I hang up, calls me back, and prompts me for the number I wish to call 3. Asterisk puts me on hold through Flash(), which is apparently the equivalent of hitting the R key on European handsets 4. Asterisk calls the

I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine

Some recent issues I had with hardware seem to come back to not understanding two very similarly named files: /etc/asterisk/dahdi-channels.conf /etc/asterisk/chan_dahdi.conf I've modified the chan_dahdi.conf to work now, but it would appear all I needed to do was include dahdi-channels.conf in chan_dahdi.conf and the problem would not have persisted? Is it me or is that a bit Monty Python?

Hello, I'm facing problem where as whenever there are incoming call from pstn, there will be no audio coming in. User at the other end also could not hear my voice. This happens few days back. Im using asterisk 1.6.1.2 with dahdi tool 2.2.0. I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and asterisk 1.6.2.5. However, it does not help at all. My current config as follows :-

Hello, I have an Asterisk box with a digium TDE205p card. The problem is that I have several "Goto(s-${DIALSTATUS}" sentences and while the call is trying to find a free channel to establish the call, I get a little noise in each "Zap/... is proceeding passing it to Zap/..." line. My configuration is: Digium card: TE205P Asterisk Version: 1.4.21.2

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

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

Hello, I have been struggling with the configuration of this card on my box. I have a T1 line and I am trying to setup asterisk with it. I followed all the instructions and I still see a blinking red light on the card. I use fedora 12. If everything is fine should I see a green line when I plug in the T1 line ? I want to isolate the issue so I di not start asterisk. When I run asterisk I get

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:

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.

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