similar to: Asterisk, tdm card and BT line:

Displaying 20 results from an estimated 1000 matches similar to: "Asterisk, tdm card and BT line:"

2006 Apr 26
1
Problem with a TDM-400P
(Sorry of this appears in the list twice, but I wasn't sure if it was blocked or not) Hi there, I'm having a problem with my TDM-400P which has been working like a charm up until very recently. It started to fail last week, and so I was hoping someone could illuminate me with some information as to why. Its configuration is as follows: ------------ FXS (green) module is in position 1,
2009 Nov 23
0
TDM400P alarm state
I'm having real problems with my connection to BT, it is a home line, but after a while it sets an alarm and only a restart of asterisk resets it could some one look at the below configs and suggest any changes to make this more reliable Thanks for your help Robb asterisk version 1.6.1.10 dahdi version SVN-trunk-r7445 dahdi_scan active=yes alarms=OK description=Wildcard TDM400P REV I
2009 Jul 03
0
DAHDI CDR problem
Hello gang, We just got MaBell to turn on our callerid. I tested the capability with a southwest bell box and a plain phone, so I know the line is sending the signal. I'm running Asterisk SVN-branch-1.4-r204834 using a TDM400P card. Here is my dahdi_cfg -vv output: dahdi_cfg -vv DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.3 Echo Canceller(s): MG2
2005 May 31
0
Re: Asterisk-Users Digest, Vol 10, Issue 234
Hello All I'm using asterisk 1.1.X and MFCR2 lib version 0.03pre2. when i call to E1 (connected with asterisk), chan_unicall don't detected event incoming call and show error. error messages: *CLI> Warning, flexibel rate not heavily tested! Rx CAS bits 0x9 [ 10000/ 0/ 0] Line unblocked -- R2 Channel 4 unblocked Rx CAS bits 0x9 [ 10000/ 0/ 0] Line unblocked -- R2
2005 Jan 27
1
TDM-400P + CallerID
Hi, I'm just starting out with Asterisk, in combination with a TDM400, filled with 2 FXS on channels 1 and 2, and 1 FXO on 4. Having just started, all I want right now is to be able to answer incoming calls on a phone connected to channel 1. The trouble is the caller id. I have caller id enabled on my line, my phone supports it, and when I connect the phone directly to the line, it works.
2005 May 27
0
Re: Asterisk-Users Digest, Vol 10, Issue 215
Hi All i'm using sangoma card. connected to E1, my wanpipe file as #================================================ # WANPIPE1 Configuration File #================================================ # # Date: Fri May 27 00:25:04 GMT+7 2005 # # Note: This file was generated automatically # by /usr/sbin/wancfg program. # # If you want to edit this file, it is # recommended
2010 Jul 29
2
Disconnect supervision tone detection
Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI
2005 Oct 12
0
X100P callerid ETSI - caller*ID failed checksum
Dear All, I am a newbie about asterisk. I have 1x X100P card 3x Sip phone I got aware of problem, after I saw the caller id on my sip phone. I noticed that if I receive a call from GSM Operator A, I can see caller id. But any other operator, I got no caller id, even my direct PSTN service operator. So at that moment I was using *1.0.9. than I changed to asterisk@home 1.3(1.0.9). I got same
2006 Oct 31
1
Asterisk does not bridge zap channels on outgoing calls
Hello... I have a big problem with asterisk. Every time i make a call asterisk does not bridge the zap channels. The zap channel from which i'm calling remains in state:ring and applicaton:dial and the zap channel with the external line configured remains in state:dialling an Application:AppDial. Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None) Zap/9-1 int_omg 09399 5 Ring
2005 Sep 12
1
Can't pickup inbound calls with TDM400P Fxo
Howdy, 1 x TDM400P card with 1 x fxo module. 1 x BT Pots line. Location - UK Calls work fine outbound but i'm unable to pickup the inbound calls. Asterisk debug: Asterisk -vvvvvvvvvvcg *CLI> -- Starting simple switch on 'Zap/1-1' -- Executing Wait("Zap/1-1", "1") in new stack -- Executing Answer("Zap/1-1", "") in new stack
2005 Feb 28
1
Zap channel calling back after hangup (due to polarity CID detection)
Today I received a TDM11B (1 FXO and 1 FXS) and got it installed just fine. I bought the card mainly to get caller ID to work properly in Sweden, and that works just fine. However, if the called or calling party hangs up after I hangup my SIP channel, polarity CID detection kicks in and dials a couple of signals to my incoming context. This happens with Asterisk 1.0.6 and CVS-HEAD. I have tried
2005 Sep 07
1
TDM400P not detecting hangup and not hanging up.
Can anyone suggest where I might begin looking for an answer please? I have just installed a TDM400P (2x FXS and 1x FXO modules installed) The first problem is that it does not seem to be able to detect if the remote party has hung up when a call comes through on the FXO. For example, if someone calls in, and then hangs up at any time after it starts ringing, Asterisk carries on as though the
2007 Jun 26
0
No CID on Zaps - TDM400
I'm running Trixbox 1.2.3 with 2 TDM400s (FXOs). With Trixbox out of the mix and a regular phone connected I get the CID fine yet Trixbox shows 'unknown': dialparties.agi: Caller ID name is 'unknown' number is 'unknown' dialparties.agi: Methodology of ring is 'ringall' Here is my Zapata.conf if it helps: ############################# ; ; Zapata telephony
2005 Mar 08
1
TDM22B in the UK on BT
Hi I am having problems getting my card to hang up properly when a remote party hangs up the line. I know i have to use the busydetect stuff but it doesn't seem to be working. It is a BT line and my zapata.conf is as follows: [channels] language=en rxgain=0.0 txgain=0.0 immediate=no echocancel=yes echocancelwhenbridged=yes echotraining=yes signalling=fxo_ks context = dialphone channel =>
2006 Feb 10
0
TDM - Analog Trunk - CallerID question
Hello list. I have a question about how to read the incoming calls' callerid on an FXO interface of a TDM 400 analog card; (it's one of those RED modules). Now -may this is the complexity adding step..- I have a GSM gateway attached to this FXO thing; incoming calls are processed as they should. But both when peeking on the CLI, as well as in the phone display I do not see the caller id.
2006 Jun 23
1
SIP -> PSTN calls not connecting properly
Hi, I've got a problem with my asterisk set up which has been going on for a while (months). I'm currently running 1.2.7.1 on a gentoo box with the topology below: +-----+ PSTN ---------+ * +------------- Service Provider (wctdm400p) +-+-+-+ IAX | | | | FXS --+ +-- SIP (cisco 7940)
2011 Jan 24
2
Outgoing FXO calls have no audio with callprogress=no
My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes
2005 Jul 21
0
Busy Extensions
Here is the output. These are Panasonic KX-TG2564's. Does something need to be set for the phones? I can call out fine, but all of the extensions seem to be busy. Starting simple switch on 'Zap/5-1' -- Executing Macro("Zap/5-1", "exten-vm|200@default|200") in new stack -- Executing SetVar("Zap/5-1", "FROMCONTEXT=exten-vm") in new stack
2008 Feb 28
1
Problems with setting up Zaptel
Hi all, I've just got an OpenVox A400P card with 1 FXO and 1 FXS module and I am just trying to get it working. But no luck as of yet. In /etc/zaptel.conf I've set the following options: fxsks=2 fxoks=1 loadzone=se defaultzone=se And in /etc/asterisk/zapata.conf I've not sure what to set exactly. For example, under [trunkgroups] what to specify there? Under [channels I set something
2009 Dec 30
2
CID not working.
Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0