similar to: asterisk dialplan to share fax line

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 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 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 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 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 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 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 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

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 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 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 All, I seem to be having a problem with all my VSPs. When I am registering with them I don't seem to be passing my port number. This problem causes other users the inability to call my VoIP number with the VSP. My VSP showed me what they are seeing. I have changed my useragent to be: Linksys/SPA941-4.1.15 Linksys/SPA941-4.1.15 Contact sip:1234321234@aa.bb.cc.dd with no

I tried this with no luck on the machine, seems the fax did not like even ulaw. Ended up just putting it inline, but of course, in a multi-line setup it could be a problem. Just a word of advice, get another line for it if you can. I fought for about a year with my pingtel ip phone over a wireless connection, now I got serious and got a normal line about 5 months ago. Much better off now...

Hi, I have Grandstream GXP-2000 connected to Asterisk, and Asterisk has trunk to VSP for PSTN calls. When ever I place local PSTN call, the landline doesn't hang up right away (40 sec), when I hang up the GXP-2000. The GXP-2000 seems to have problems making international calls as well. Where it hangs up soon as the other party picks up. I have used different IP phones, VSP's and etc.

Hi All, I think I have missed something as I am resisted with 4 VSPs and I can not dial in using any one of them using the corresponding VoIP numbers assigned with the VSP. I can make outbound calls to another VoIP number to the same provider. The weird thing is that I have a DID with a VSP and I have that working fine but try using the associate VoIP number and nothing happens. When

Hi My setup is Internet -> firewall -> asteriskbox -> spa3102a -> spa3102b the spa's can talk to the firewall directly. The firewall does NAT. The current asterisk flow for outgoing calls is phone => spa3102 => asterisk => vsp and vis versa for inbound calls. can I use re invite for outbound calls such that the spa3102

the caller is out his money anyway when you call any phone and voicemail kicks in, although i think on a payphone they give you a 2 or 3 second window to hang up. Suggest you implement i'm here / i'm away dialplan logic or set the do not disturb button that way when someone calls and the guy is away it hits voicemail right away and the caller can hear this and still have the 2 or 3

Hello All, For some reason my asterisk server is not registering a port number with my VSPs. This is causing problems where people are not able to dial in from any of my SIP or IAX VSPs. I do have one VSP that has hard coded my IP and port so I can get incoming calls but this still leaves a problem with my other VSPs. Hose can I get asterisk to register my IP and port? I have been

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

All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch,