similar to: TDM400P: flash on analog phones doesn't work

I am having trouble getting transfers to work when a zap channel is part of the call. I have a couple SPA-2000's and some X100P cards as my setup. This is what I'm trying: Dial number from phone: -- Executing Dial("SIP/206-2c61", "Zap/1/#######") in new stack Currently on call: -- Called 1/####### Press flash to place call on hold with SPA-2000: --

I've been trying to get fax reception to work using an SPA-2000 to ring the fax machine or modem that's taking fax calls. I was curious if anyone else had tried something similar, and if so, had any luck getting it to work reliably. I've been able to get it to work, but it isn't reliable. (Pages/lines of black dots result more frequently than not.) The incoming lines are FXO

Hi, I recently installed a gentoo on a PC, purchased a TDM400 and installed asterisk. Kernel is 2.6.11-rc3 First, there seems to be a problem with my system: I must put the pci=routeirq to make the card detected. The modules are loaded ok (dmesg output) Zapata Telephony Interface Registered on major 196 ACPI: PCI interrupt 0000:00:08.0[A] -> GSI 16 (level, low) -> IRQ 16 Freshmaker

When I reload the config, I see this error in the CLI. However, I don't see what I have done wrong: == Parsing '/etc/asterisk/zapata.conf': Found May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 1, FXO Kewlstart signalling May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured

On Jun 20, 2004, at 9:16 PM, asterisk-users-request@lists.digium.com wrote: > I am new to Asterisk, but though that I would need calls being > answered in > different contexts. > > How can I direct one line to a given context and the other one to > another, > or is there a better way??? > I use this at the end of my zapata.conf file to do exactly what you're

Hi, I have a running asterisk 1.2.1 (bristuffed) with a TDM400 Board. Connected to 1 port is am ordanary Fax Machine. Everything 'seems' to work, however receiving faxes is very unreliable. Sometimes I receive a normal page, without problems. Sometimes half of a page and the rest is scrambled, but most often, I receive nothing and the other site reports a Fax error... The Fax machine

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.

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

I'm replacing a Merlin for a client and they have a PagePal Intercom that I would like to reuse. Here is what I know about it: It has a screw-down wires that goto rj-11 (This was told to me over the phone) that went into one of the Merlin ports. I tried bring it up with fxo_ks and fxo_ls (assuming it was analog and autoanswered) but no luck. I would be happy to replace if anyone knows of

Hi, I facing a problem with flash hook. When ever I do a flash hook to place an extsing call on hold, the call gets disconnected. The debugs on Asterisk shows that 'on hook event detected' when I press the flash button on the phone. The setup is like this Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD and configured for ISDN PRI lines. Analog phones come

This is, hopefully, just a case of brain fade. With zapata.conf and zaptel.conf in place, asterisk loaded, no dial plan and all LEDS on the card lit, I get no dial tone, plugging an analog phone into ports 1 or 2, only a buzz and click. zaptel.conf - defaultzone=us loadzone=us fxoks=1,2 fxsks=3,4 zapata.conf [channels] signalling=fxo_ks language=us context=phones-1 group=0

Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128

Hi all, I'm trying to install a TDM400P card, and I need some help. Please, see below... after dmesg command: [root@voip root]# dmesg via82cxxx: board #1 at 0xD800, IRQ 5 Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 3 for device 00:09.0 PCI: Sharing IRQ 3 with 00:10.1 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1:

I had to return my TDM11B because it put the PSTN line 'off hook' the moment I plugged it in and wouldn't hang it up. The new card seems to work because I can actually make an outgoing call from the FXO port to my cell phone, so I'm pretty happy about that. But Asterisk doesn't recognize incoming calls from the PSTN. If I dial my home phone from my cell phone asterisk

Hello, everyone After having spent several time to look for any solution for my problem, I decided to write here. Here is the problem got a Digium X101 FXO card and Digium TDM400P alias freshmaker (1 fxs module on it on first port ) in my asterisk box. The X101 works perfectly. The problem comes from the TDM400, when i plug a tested working phone on it , i get no dialtone; the goal is to

I have an intermittent problem with the one FXS line that I have. On most calls, the first ~5 seconds of the call has a loud buzzing noise on the line. After 5 seconds or so, it fades off to nothing, and the sound quality is great. Searching for "buzzing" on the list doesn't give a whole lot to work with. The buzzing happens on calls that are routed over both my FXO line and

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]

Hi to all, I have been using Wildcard TDM400P with four fxs modules and it was working fine now I have added another Wildcard TDM400P with four fxs modules . So there are total 8 ports for 8 hard phones. I have modified following configurations In /etc/zaptel.conf loadzone=us defaultzone=us fxoks=1-8 in /etc/asterisk/zapata.conf context=headoffice signalling = fxo_ks

> -----Original Message----- > From: C F [mailto:shmaltz@gmail.com] > Sent: Tuesday, January 11, 2005 4:38 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] internal caller id on analog phones > connected tozap > > How are the analog phones connected to * ? this is where the setting > should be. They're connected to

Run /usr/src/zaptel/ztmonitor 32 -v And adjust your gains in /etc/asterisk/zapata.conf accordingly. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of bam Sent: Friday, May 07, 2004 3:35 AM To: asterisk-users@lists.digium.com