similar to: Problem getting TDM400P clone card to go off-hook and dial

Hi, I am trying to implement 911 funtionality in my PBX. A call should drop if all lines are busy. Here is my context nineoneone from extensions.conf [nineoneone] exten => s,1,Set(SET_EMERG_FLAG=0) exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten => s,n,Set(EMERGENCY=1,g) exten => s,n,Set(SET_EMERG_FLAG=1) exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these extensions.conf lines: exten => s,1,SetVar(SET_EMERG_FLAG=0) exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten => s,n,SetGlobalVar(EMERGENCY=1) exten => s,n,SetVar(SET_EMERG_FLAG=1) exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM}) exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} =

Hi, I'm struggling to get busydetect to work. I'm using asterisk 1.2.1 and a digium TDM04B (4 port FXO) card. I've set busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf and i've modified zondata.c with a busy setting of 620+480, 300/200 which is the busysignal received from Korea Telecom. Asterisk isn't detecting the busy signal and doesn't hangup.

Hi, Running Asterisk CVS-Head latest on a Dual P3 800 1Gb Ram. For some odd reason now that I have the asterisk box almost to the stage I want it, I hit a problem. I have a te405p in the system, Zap/g1 is connected to the telco as an ISDN 30, Zap/g4 is connected by ISDN Primary Rate to an Ericcson BP250 phone system. When calls come in on g1 they go straight through instantaneously to the

Hi, We have a Ericsson BP250 Phone system setup witht he following configuration Telco <-> Asterisk E405P <-> BP250 The system seem to work perfectly on 1.0.9 for a very long time but there is some functionality we wanted to take advantage of in the 1.2 version branch so we upgraded. Currently running Asterisk 1.2.4 Zaptel 1.2.3 (noticed 1.2.4 is out will upgrade next

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

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

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 ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

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

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]

# 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

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

Hi, I've searched and searched on the possible problems. If anyone can help me that would be great. Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) B8ZS/ESF CRC4 error count: 4750 E-bit error count: 5023 Timing slips: 72 1 TE2/0/1/1 Clear (In use) (SWEC: MG2) 2 TE2/0/1/2 Clear (In use) (SWEC: MG2) 3 TE2/0/1/3 Clear (In

The update worked like a charm! Hold music is as cheesy as ever! Thanks much, this list is a life saver! Dan ------------------------------ Message: 2 Date: Fri, 18 Mar 2005 09:16:59 -0600 From: Eric Wieling <eric@fnords.org> Subject: Re: [Asterisk-Users] Redhat 9 Music on hold To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>

Hi, I've seen this USB product from Sangoma : http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/usb_fxo.html Is it working ok ? Is it easy to integrate it with Asterisk ? How would you rate your experience with it ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 :-