similar to: CallerID in Australia & Analogue PSTN Phone System

I'm trying to get the X100P card working in AU. So far I have managed to get it to handle incoming calls from the PSTN and have managed to eliminate pretty much most of the echo. My big problem is getting the outbound calls to work. When I get ZAP to dial out it won't connect and I get what I think is the Congestion signal - like a busy signal but with what appears to be a 10db

Is it possible to run the Festival command in the same manner as the Background command so that it can be interrupted by caller key presses? -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you choose Linux; when you want a system that just works, you choose

Will Wait(n) still listen for DTMF input from the caller after there has been a Background(some-message) prompt, or do I need to use Background(silence/n) to still listen for DTMF? -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you choose Linux; when you want a

Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you choose Linux; when you

Can anyone _recommend_ a downloadable OSS softphone that _works_ under Linux and is compatible with Asterisk. So far I have tried kphone and linphone and had problems with both, and I am still waiting to hear back from the X-Lite beta folks. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just

Surely there has to be one soft phone that works under Linux. I've tried: kphone - it sometimes complains about the need to release the sound device linphone - sssssssslllllllllooooooooowwwwwww iaxcomm - needs some strange widgets various others - either only supplied as binaries, or just plain don't work, or won't compile. Is there just one out there that is guaranteed to work with

When I try to start up zaptel, whilst running ztcfg, I get the following error: Jan 1 10:48:18 bu ztcfg: ZT_CHANCONFIG failed on channel 2: No such device or address (6) My /etc/zaptel.conf is: fxsks=1 fxoks=2 loadzone = au defaultzone=au Channel 1 is a X101P card connected to the PSTN and channel 2 is a S100U box driving an analogue phone. The zaptel kernel module gets loaded OK as does the

When I make a call from a SIP phone to a speaking extension on *, such as one that speaks digits or similar, when I monitor * in very verbose mode I can see it running through the routine associated with the extension, but I am getting no RTP data stream back to the phone. Does the machine housing * need a sound card? Does it need OSS or ALSA modules installed? What actually generates the RTP

I have a situation where I need to know which Zap channel an incoming call is on, so that the call can be answered appropriately when a SIP phone displays the channel. These Zap calls are coming in over PSTN and don't have caller ID. As far as I can make out my SIP phones (WuChuan HOP-1002) display the user part from the SIP "From:" header as the second line on the display. If the

Does anyone know why the "n" priority in the dial plan is not recognised in 1.0.6 It seems strange to me that it should be so. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you choose Linux; when you want a system that just works, you choose

Hi, Are there any resellers of phones that can take power over ethernet in Australia? All I can find for sale online is the BT-10[12], which is cheap but not featureful enough, and the Snom 190, which is about right, but neither of them support PoE. I'm particularly intereseted in the Snom 220 with the keypad expansion for our receptionist. Although, could you make a PoE split-out cable

For some thime now I have been trying to get some sense out of Caller ID on PSTN lines in AU and have been getting no where. Now, at last, I seem to be getting something, even though it is an error message (line 5 below). All and any guidance would be welcome. -- Starting simple switch on 'Zap/1-1' Jan 10 09:36:38 NOTICE[18384]: chan_zap.c:5514 ss_thread: Got event 2

Has anyone had problems with Call Waiting signals causing Zap channel or bridging hangups in AU. I was on a call the other day (Zap channel to PSTN) and the call suddenly hung up on my side. I dialled the calling party and got the call again, it seems that the bridge had dropped and that the other party had not lost the connection. As soon as I got the bridging again the other party mentioned

Hi. I need help with my brand new Voicetronix OpenLine4 board that I installed into Asterisk. After building the Linux device driver and inserting the module, I modified the /usr/src/asterisk/channels/chan_vpb.c file to uncomment the US settings and comment out the Austrailian ones. I made the appropriate entries for routing in vpb.conf and extensions.conf.... All appears to be well, except

I've just replaced a X101P card with a brand new TDM400P card (specifically TDM421B). I do have the molex plug attached. kudzu removed the config for the X101P OK, but didn't find the TDM400P lspci does not show the card ?? Bung card ?? How susceptible are these cards to XRays, as it has been thru AU customs and might have been thru the scanner. -- Howard. LANNet Computing

In AU we have a number of different dialtones defined for various purposes. >From indications.conf: au <ringcadance> 400,200,400,2000 au dial 413+438 au busy 425/375,0/375 au ring 413+438/400,0/200,413+438/400,0/2000 au congestion 425/375,0/375,420/375,0/375 au callwaiting 425/200,0/200,425/200,0/4400 au

I think I have got * worked out as far as getting users on a small private network talking with each other, but when it comes to the bigger picture about talking between private networks connected by the Internet then I am getting confused about STUN, SER, SIPPROXY, RTPPROXY, etc. Before I start let me make it clear that I am not looking to drop out onto the public telco network anywhere, not at

I am running * in a development environment, adding functionality as I go. The * box has a X100P card in it which ztcfg enabled as channel 1 with fxsks signalling (fxsks=1). Everything worked fine and I was able to make inbound and outbound calls to/from the PSTN, the only issue being that some exchanges wouldn't handle the dtmf signalling, but I put that down to a peculiarity with some AU

Good day all I'm getting this error while trying to dial out on my asterisk server using a openline4 card "exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872" Please Help me

Hello, Hacker I install VoiceTronix OpenSwitch 12 port PCI Telephone Card, and setting vpb.conf, extensions.conf following My problem is: When i dial to fxo(channel 9-12), it is ok, but when i continue press exten 102, the channel crach with error messages following exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872 Do i ignore some setting for VoiceTronix OpenSwitch12