similar to: Re: Three way calling on outgoing FXO line (Martin Pycko)

I was wondering if anyone is able to use the three way calling features from their telco on the incoming FXO lines to transfer a caller back out to say a cell phone. I am currently moving from a Talkswitch to the Asterisk PBX and one nice feature they have is after 4 rings or so I can have the call transferred to my cell phone using the same line it came in on with three way calling. Just

I have an adit 600 with an fxo card connected to a digium T1 card. If I try to make an outgoing call and the T1 cable is disconnected, asterisks returns congested like it should. But, if the adit 600 is connected to the T1 card, the adit 600 immediately "answers" the call even if there are no physical lines attached. I even removed the fxo card and the adit 600 still

Hi all, We have an analog line from telco, on which 3-way calling is subscribed to. This line is plugged into an FXO module on a tdm400p. If an incoming call comes in on this line, can */zaptel send Flash to telco via the FXO module? If it could, then an incoming call could be 'transfered' to a cell-phone, for example, with a single analog line. (where 'transfer' is really

Hello, I'm planning to convert my phone system to Asterisk, as I've outgrown my TalkSwitch system. I have a few questions for experienced * users, most of which can be answered yes/no. Current Setup: - Talkswitch 48NLS (4CO/8Ext) phone system. - One CO line, two Vonage lines, one Voicepulse line connected to phone system - A third Vonage line directly connected to a fax machine - A

Do you have threewaycalling and transfer set in your zapata.conf? Here's mine (four TDM400's, seems to be working so far). I didn't do anything in my extensions.conf for any of these features (what confused me at first is the t and T options of the Dial application in extensions.conf are for transfers via the # key), when you flash you get another dialtone that works just like the

I tried putting this as the ata's dailplan: *St4-|#St4-|9|^9t4>$.- this is sip.conf [ata2001] type=friend username=ata2001 secret=SoMeSeCrEt host=dynamic context=fromata canreinvite=no and this in extensions.conf [fromata] ignorepat => 9 exten => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) exten =>

hello everybody , i got a test setup with lanphone 101 connected to * and a welltech fxo 3802 also to * the extensions are configure so that i can dial from the lanphone to the fxo. although once on the FXO and having the dialtone, no of my dtmf dialing is being processed on the FXO. It keeps giving me the dialtone constatly. Everything is configured as outband DTMF (we

My home asterisk system has failed the wife test (still too much echo with the current hardware... my voice seems okay but when she talks she complains of echo), but I'd still like to be able to use it to send and receive fixed line sms messages to and from my mobile. My x100p card has 2 ports on it, a 'line' port and a 'phone' port, so there is provision there to plug the

Hi *, now for a long time i am trying to set the outgoing callerid, without luck. I am here in Germany, my asterisk has a pri interface connected to a PMX installed by Telekom. All telephone calls are preselected to EcoVoice. I am using asterisk 1.2.7.1, zaptel 1.2.5 and libpri 1.2.2. A week ago we tried with a device able to simulate a telephone system so send out a callerid, and that

Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-----[*]----[mediant2k]-----[Avaya_PBX]------e1-----[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give

Anyone interested in this feature? I have a version 0.1 patch, which is currently against 1.2.25-bristuffed, but which should port trivially to almost any version. I am away until Tuesday 8th April, but if there is enough interest, I will open a "new-feature" ticket and upload the patch to the bugtracker so that more capable programmers can laugh at it ;-) It should work reasonably on

Does anyone know a good (and stable) voip gateway product with 4 ports (2 fxo and 2 fxs), with the following requirements: * being able to connect analog phones to the FXS ports, and communicate over SIP with an REGISTRAR/PROXY server (SER in our case). * being able to connect the FXO port to local office PSTN network, and dial to that office pstn number and getting an internal dialtone, or

I am trying to configure the FXO port on a Sipura-3000 for use with Asterisk. When I connect to the Sipura to dial out on the PSTN line connected to the Sipura's FXO port, it gives me the dialtone of the PSTN line and then I can hear the DTMF for the number I dialled beforehand. It does work but the customer perceives this delayed second DTMF feedback as "unprofessional" and the

Hi Guys, i've successfully installed the current bristuff, everything works fine, exept one thing: As soon as i pick up the phone that is connected to the bri-card, asterisk jumps into extension s of the context that is specified in zapata.conf, if i have immediate=no then i hear the normal dialtone for about 1/10 of a second. Any hints what i'm doing wrong? If i dial the number

My system (Mandriva2006) crashed recently due to hardware difficulties. on reinstalling wine I get the following message when trying to run the program TreePadSAFE, which used to work OK Any Idea how this could be fixed? T.I.A. Carlton. [carlton@localhost TreePadSAFE]$ wine treepadSAFE.exe ALSA lib pcm_dmix.c:802:(snd_pcm_dmix_open) unable to open slave fixme:winspool:WINSPOOL_EnumPrinters We

Hello All... I have searched in the archive and also followed Zara's instruction on getting incoming calls to work with Asterisk...but I still can't get Asterisk to answer incoming call on Linejack's LINE port. I attached a phone set to the PHONE port, and telco line to the LINE port on the Linejack(ISA) card. I have downloaded, compiled and installed the newest driver for

I'm having trouble with incoming calls that start to ring again a few seconds after the local user hangs up. I get "Starting simple switch" on the FXO port, so it's not just a matter of mistakingly initiating a 3-way call on hang up or something. Picking up the ringing telephone will get a CO dialtone. This happens, say, 50% of the time for incoming calls, never for outgoing

Since the irc channel wasn't any help, I will try posting my problem here. I have two TDM400Ps less than a week old in a PC. All of the FXS ports work fine, and all of the FXO ports worked fine up until thisafternoon. If I try to dial in, I get a busy signal, if I try to dial out, all I hear is a very scratchy, very crackly dialtone. If I swap the first FXO module with the second on the card,

Hi, I got back from two weeks away and appear to have lost audio on my tdm411 fxo. Everything was working properly when I left. I checked the logs, config files and can't see any problems, the zap channels and modules are all loaded properly, asterisk starts without probs and everything looks sweet on the colsole with -vvvvvvvvc when I make calls, but I just don't hear a dialtone or

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