similar to: Echo on TE121B with hardware echo module

Hi all, I just posted this question before last week. Meanwhile after upgrading Asterisk 1.0.7-BRIstuffed-0.2.0-RCg to 1.0.8-BRIstuffed-0.2.0-RCh the same problem occurs, but seems to be more seldom. Attached is now the output of "zap show channel" . - I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns OctoBRI running in NT-mode.

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

I new in asterisk world so, please, forgive me if I say something stupid. At least, and after a lot of tryes, the isdn card seems to be registered: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Automatically generated pseudo

I've been having trouble with making outbound calls to my TELCO from a TDM400 card (FXS KS signalling) after upgrading from 1.6-beta9 to 1.6.0. The problem is completely intermittent. When it fails, I get this message: [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) At some point, it starts working, but I don't know what

Hi, We've got a dedicated T1 with two trunks running into our ILECs selective router for 911. Split out of the T1 into two MF CAMA trunks on ILEC side. I'm trying to use asterisks e911 signaling, but I'm having trouble with the dial command. (== Everyone is busy/congested at this time (1:1/0/0)) I'm missing something and I'm thinking it has to do with the hookstate

Hello all, I have an asterisk box running in a customer with Hylafax, iaxmodem, asterisk 1.2.18. The service can receive faxes, from a lot of fax machines, but there are a couple of them that asterisk Hylafax cannot complete. This calls arrive the asterisk box, asterisk detect that this calls are fax, asterisk answer the call, and then Hangup the call. But hylafax do not receive nothing. When

I have a TDM400P with fxo and fxs modules, and local extension calls work fine but I can not dial-out. I am getting both problems on FreeBSD and Linux, and plugging a regular telefon set works fine. Any ideas of what I am missing? thanks micko Zaptel Configuration ====================== Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default)

Hi I have the X100P card which as to sockets (LINE - for fxo line ) and PHONE (to connect a n analog line) This card is setup as fxs_ks I was getting dial tone but suddenly no Dial Tone....Help appreciated.... When I try to route the Call using - Dial Zap/1) to this FXO Line I get this error: ------------------------------------- linux*CLI> -- Executing

Hi, I've a busy server where one E1/PRI span has been behaving oddly. Here's the span: Primary D-channel: 16 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 Looks good, eh? But every

I am running Asterisk 1.0.9 and the latest bristuff in combination with a HFC isdn card, connected to a BRI interface. For some reason, I am not able to have it dial out (see below). It exits with DIALSTATUS=CHANUNAVAIL. One thing that may be misconfigured is that it says: Signalling Type: PRI Signalling. How do I change this? Or is this the correct value? I posted my configs below. ------

Hello, I have a strange problem with one of my Zap channels. A user told me that he was in a voicemail system during a call, hit the Flash button, and the call hung up. Now we get no dialtone on the phone hooked up to the channel. Here's the status of the channel: jmartin at rogue:~$ sudo asterisk -r -x "zap show channel 7" Parsing /etc/asterisk/extconfig.conf Channel: 7 File

Hi guys, This has been bugging me for days and I can't figure out what's happening. Using the same channel on DAHDI, I can't make a call after previous one being hung up by remote side. However, it works like it should if my end hangs up the call. To make things interesting, DAHDI, or astribank rather, seems to "reset" itself after receiving an incoming call on that DAHDI

Im having a problem getting the 2 cards working together. I added the TDM with 3 FXO and 1 FXS board. The TE405p is working fine. THe TDM registers and identifies the FXO's and FXS but cant get them to go off hook. I hope someone can help. Asterisk seems to THINK its dialing it sends manager status as dialing, and it just sits there with no tones, on the phone dialing out. If I dial,

Hi, I have problem with X100P clone card.I can not force it to work under Asterisk 1.4.27.1 and DAHDI Version: 2.2.0.2. I looked over and over on configuration and could not see any mistakes. Here are relevant configuration files. /etc/dahdi/system.conf > fxsks=1 > echocanceller=mg2,1 > loadzone = hu > defaultzone = hu > dahdi_tool > OK Wildcard X100P

I have echocancel=yes in zapata.conf but when I do a zap show channel 1, I notice echo cancellation is turned off. I followed the article that talks about the order in which the statements need to be in zapata.conf to get echo canceling to work: http://lists.digium.com/pipermail/asterisk-users/2005-June/110615.html But it is still not working. Does anyone know how to get echo

Hi List, I have configured TE121PF card in E1 mode. I am using asterisk 1.6 and freepbx 2.7. My card staus turn in to green and looks like sync with the service provider. My service provider is BSNL - India. I have one toll free number for incoming and one land line number for out going calls. Problem : If i am calling to the toll free number, i am getting the ring but that call is

Hello everyone, I have a Atcom AX-1600P(1) card with a FXO module and I can't configure it. I have four extension with this PJSIP settings: --- /etc/asterisk/pjsip.conf --- [transport-udp] type=transport protocol=udp bind=0.0.0.0 [6001] type=endpoint transport=transport-udp context=from-internal disallow=all allow=ulaw auth=6001 aors=6001 direct_media=no rtp_symmetric=yes force_rport=yes

In order solve my incoming caller ID problem, I upgrade the dahdi to version 2.6.1 from version 2.4.x. After upgrade, I found the echo cancellation doesn't working (I'm using Digium AEX800B PCI Express card). I can hear my self talking on the phone. How to solve this? I think I need to recompile dahdi 2.6.1 with OSLEC support? how? [root at callcenter ~]# dahdi_cfg -vvv DAHDI Tools

I am unable to dial out over a Wildcard TDM400P. This was working previously, so must have messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX 2.5.2.2. When I dial, I see: -- Executing [s at macro-dialout-trunk:19] Dial("DAHDI/1-1", "DAHDI/g0/9239220,300,") in new stack == Everyone is busy/congested at this time (1:0/1/0) It looks like the

Hi All, I am installing a Zaptel Quad E1 card with ISDN PRI in Brazil and I?m with this problem: Asterisk CLI : Mar 27 01:49:57 WARNING[2612]: chan_zap.c:7143 zt_pri_error: PRI: !! No channel map, no channel, and no ds1? W hat am I supposed to identify? Mar 27 01:49:57 WARNING[2612]: chan_zap.c:7143 zt_pri_error: PRI: !! Unable to add IE 'Channel Identification' == Restart