similar to: Problem with Asterisk + TDM410 FXO

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

Did Linksys really use Asterisk for the SPA9000 software? -- Andres Technical Support http://www.telesip.net

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 there, I was wondering if anyone has successfully used Asterisk as a dedicated Analog PSTN gateway to take the place of, for example, a Mediatrix 1204 or an 8 port model? Basically, I am thinking of using a Linksys SPA9000 as the PBX and just need an Analog PSTN gateway for 4 to 8 FXO lines. It does not sound like the Mediatrix 1204 does a very good job and I figure I can build a much more

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

I have a TMD400 card installed in a PC with one fxs (installed in slot 2) and two fxos (installed in slots 3 & 4).? fxos work fine but I am unable to get a dial tone for any devices connected to the fxs.? I have correctly connected the power supply to the card and I have even tried moving the card from slot 1 to 2 on the board. Below is from the console when I try to route a call from FXO on

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

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

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

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?

I get the following errors when starting Asterisk. == Parsing '/etc/asterisk/zapata.conf': Found Feb 19 21:14:35 WARNING[10440]: chan_zap.c:920 zt_open: Unable to specify channel 1: No such device Feb 19 21:14:35 ERROR[10440]: chan_zap.c:6860 mkintf: Unable to open channel 1: No such device here = 0, tmp->channel = 1, channel = 1 Feb 19 21:14:35

Hi All, Can Asterisk dial extension which resides in the PABX? (eg. 2000) Sip Phone <-----> Asterisk <------> ATA (FXS) <------> (CO side) PABX <-----> Extension (eg. 1000) (2100 & 2101) can my sip phone call to pabx extension 1000? What will be my dial plan? I know I can connect to 1000 by

Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story ........ " This is with regard to the setup which you can find at the "Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our

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

As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;

http://voip-info.org/wiki/view/Easy+PABX With Easy PABX you can create your own virtual PABX online in just minutes. Easy PABX is based on Asterisk and best of all - it's completely free. Regards thorben.dk

I am installing one asterisk, to establish connection with my PABX Siemens, in ISDN, link went up normally, also I obtain to internally call the branches the PABX, normally, but when I try to dial for the PSTN, through pabx with the command exten = _ 19xxxxxxxx, 1, dial(zap/g2/${EXTEN}, 30) asterisk, reports me the following error: -- Executing Dial("SIP/8110-a729",

Hello all! I'm VERY new in using VoIP. I'm looking for any tip or trick to connect a physically PABX behind an Asterisk-System(or similar) via an SIP to Analog- or ISDN-Converter. The point is, I _need_ to deliver calls to extensions in the connected PABX directly (in ISDN-speech "DDI" (DirectDialIn)) without intervention of an operator. Is this technically possible, and if

Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil

Hi- I'm totally stumped configuring my TDM400P with one FXS and one FXO module. Before I got the FXO module, I used to have an X101P, and everything was working very well. Now * doesn't seem to recognize the FXO channel. I've searched the wiki and the list archives. Stock Debian 3.0 stable installation. Any advice? Thanks. -- David Here's my configuration: modprobe zaptel