similar to: Analogue card recommendation

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 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 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 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 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 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 seeing this in my syslog [235900.797660] dahdi: Registered tone zone 0 (United States / North America) I am in Australia so I would want to set them to AUS zone I have got this though options wctdm24xxp opermode=AUSTRALIA thanks -- "See, we love -- we love freedom. That's what they didn't understand. They hate things - we love things." - George W. Bush

Hi I have a site that has asterisk install with a tdm410 one port is connected to a pstn that is used as a backup outbound line when/if the internet/voip is unavailable. Currently my dial plan for this line is to ignore it, I just basically do a s,1,noop s,n,wait (60) s,n,hangup what I would like to do is if I call from a certain number be given an option to run some commands or for the

Hi Everyone, I want to fine tune the Rx and Tx gain on an analogue Sangoma card by dialing into another server that is running on Sangoma PRI card (both services on Bell network). [mwatt1004khz] exten => s,1,Answer exten => s,n,PlayTones(1004/1000) exten => s,n,Wait(300) If I match the Rx/Tx numbers on both sides by monitoring "ztmonitor X -vv" am I right with my theory of

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

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

Hello, I have two TDM2400 card with some 40 FXS modules and 4 FXO modules. I would like use analogue lines for outboud calls. How is it possibile to detect ANSWER? - answeronpolarityswitch does not seem to work in Italy - call progress does not give safe results, sometimes calls get billed, sometimes not Am I forced to buy an ISDN adapter or could there be a solution (maybe tweaking some

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

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 Everyone, I am trying to find the issue of dropped calls in the middle of the conversation. The system is Elastix. Anyway to know which party hangup the channel in case of Asterisk 1.4 and Sangoma analogue cards? (this is not PRI) Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dears; I am facing a problem in disconnecting the calls, it is related to the rtptimeout (disconnecting if there is no RTP packets from both sides). My Asterisk Box is behind NAT but there is a static real IP address at the ADSL router. We call from the Mobile to the PSTN analogue numbers which are connected to Asterisk Analogue card (the telephone lines are analoge), and then we dial the

On Wed, Feb 17, 2016 at 8:14 AM, Mitul Limbani <mitul at enterux.in> wrote: > Sangoma 50 port FXS Thanks. Will I now stack 20 boxes in order to achieve the 1000 FXS lines? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/0d4c2800/attachment.html>

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

Thanks Harry. I will check and revert. I hope it work perfectly with asterisk. Regards On Wed, Feb 17, 2016 at 8:32 AM, Harry McGregor <hmcgregor at biggeeks.org> wrote: > Hi, > > For analog, I really like telco grade channel banks. > > I would recommend the adit 600, there is a good market on Ebay, and you > can do 48 channels per adit 600, with 2 T1 interfaces. Having