similar to: system cmd + fax line

Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the motherboard is too small. I have read up about a PWR2400b and it seems to use 2wire pin, I am guessing to connect to P8 just behind the molex connector on the tdm410. can any one here confirm this, or have any info on the

Hi I was wondering if any one has used these cards, I am looking at this as a replacement for the tdm410, I have some issues with installing the tdm410 in a small case because of the power plug being at the end of the board. I am in australia seems like we have a different setup for out fxs voltage, any one in oz using this card ? Thanks Alex -------------- next part -------------- A non-text

Hi I am in the process of installing a new box and using dahdi. I have a tdm410 + hardware echo canceller. I have just read in the read me for dadhi that VPMADT032 support has been removed and unlike with the zaptel stuff i could just download and install the firmware I can't with dahdi what is the best way forward to recompile with hardware echo canceller support. Thanks Alex

Hi I am having some problem forcing my tdm410 to alaw over ulaw, I have 1.6.1.0 asterisk (debian i486) dahdi 1:2.2.0 built with the hardware echo canceller firmware /etc/asterisk/chan_dahdi.conf alaw=1-4 but I have this in the general section, before any channel definition dahdi show channel 1 Default law: ulaw even when i have a call going it is still ulaw Thanks Alex

Hi I have a fxs (tdm410 ) connected to a pstn that is primarily used for faxing, it is meant to be a just in case line. How do I tell asterisk to ignore the line completely - ie don;t pick up when it rings ? Alex -- "I will have a foreign-handed foreign policy." - George W. Bush 09/27/2000 Redwood, CA -------------- next part -------------- A non-text attachment was scrubbed...

Hi I have an account with mynetphone (australia), which gives me two voip (sip) accounts, which i used to have connected to a spa9000. this is behind a firewall, so on the spa9000 I would listen on another port apart from 5060. so on the firewall 5060 would go to voip1 and 5061 to voip2. I moved to asterisk (+tdm410) and the machine was also the firewall and I had no problem - well atleast it

Hi I am in the process of move a company from pstn to an asterisk setup. They had 2 pstn lines - only really needed a max of 2 previously. Now I have installed a tdm410 to handle the cross over from pabx to voip handset. this has been done, the tdm is now just used to provide a backup pstn line - only used as a last resort for outgoing calls - as its shared with a fax line. I use 2 voip

Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (>5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to

Hi I am trying to setup asterisk at home, I have 1 in bound VSP (I have a register cmd setup for that in asterisk). At home I have a cordless phone with 2 line capability - I currently have 2 spa3102's in place to handle the 2 lines ( I am in the process of buying tdm410 to handle to handle this and the backup pstn line). I also have 2 laptops setup with soft sip phones. What I would like

Hi I had a working system, until recently - its asterisk 1.6.1 from debian - not the lastest as the last doesn't seem to work. but somebody who rang me said my voice mail announcement was all stuttery. so i dialed my voicemail box and its really stuttery... so I have done a reboot and its just as bad, now I am not sure what to check to try and get this working again ..... Alex

Becasue of all the issues with fax over voip, we want to use pstn for our fax machine, but not dedicate a line just to fax. I'm thinking of having asterisk answer the pstn line, check for fax tones, and route appropriately. In zapata ( chan_dahdi ) set faxdetect=incoming then the dial plan would have [incoming-pstn] exten => fax,1,Dial(DAHDI/1) ; the fax machine exten =>

Hi everyone! I have a Digium Card TDM410 But, it appear for me this massege chan_dahdi.c:8061 handle_alarms: Detected alarm on channel 3: Red Alarm But my line is ok! But sometimes it back sig_analog.c:3807 analog_handle_init_event: Alarm cleared on channel 2 But it again back to red alarm. What can be happen? My lines is all ok! But when I put on Digium Card TDM410 is very inconsistent

Humm, if I put a filter in this lines, maybe back? 2016-01-05 12:36 GMT-02:00, Ryan, Travis <RyanT at oscarwinski.com>: > >> -----Original Message----- >> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- >> bounces at lists.digium.com] On Behalf Of Vitor Mazuco >> Sent: Tuesday, January 05, 2016 9:21 AM >> To: Asterisk Users Mailing

I am having a issues with adding a analog card to my dell 2800. I already have a t1 card installed and running fine but when I install the analog card asterisk will not start (ztcfg fails). I have determined it is because of a IRQ problem and have decided to get a new server. Can anyone suggest a server grade setup that supports this? I would rather not buy a machine that will be unstable. I

Hello, Trying to set up res_fax_spandsp. Based on https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in my extensions.conf: exten => 306,1,NoOp(Fax transmission) same => n,Set(FAXOPT(gateway)=yes) same => n,Dial(DAHDI/3) ----->FXS port to fax machine same => n,Hangup() Call flow Im trying to pull out is as follows: Zoiper -->

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

Hi all, Recently I have instaled one Digium TDM410 on my Asterisk. After instaled , I can do outgoing calls but I cant receive calls. I receive the following messages: chan_dahdi.c: Got event 2 (Ring/Answered)...[Sep 14 11:24:44] NOTICE[2654] chan_dahdi.c: Got event 18 (Ring Begin)...[Sep 14 11:24:44] WARNING[2654] pbx.c: Channel 'DAHDI/4-1' sent into invalid extension 's' in

Hi, Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a "fully-compatible" Openvox product...but is it really "fully-compatible"? Any experience about Openvox products (card and zaptel versions, etc...)? Thank you! Giorgio.

I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO modules. When I plug one PSTN-line into a FXO-port I am able to receive calls on this line and I can also make calls from an internal SIP-phone to the external PSTN-network. Still I am bothered about something that appears on the CLI when I do a reload chan_dahdi.so : asterisk*CLI> reload chan_dahdi.so -- Reloading module

> Subject: Re: [Asterisk-Users] Booting error - Unable to specify channel 2: > No such device I just went through all this as well. The best thing to do IMHO is to try to find a way to manually assign IRQ in the BIOS. Also, and this is what I didn't see at first, some slots SHARE IRQ. Avoid this! If you are not using USB at all, turn it off in BIOS if possible. Tell the BIOS