similar to: Three way calling on outgoing FXO line

I guess what I meant to ask was for a way to do it from within extensions.conf. Using either the Dial command or if there is another method to do the three way calling. >Press flash on your phone (asterisk will intercept that) and then when you >have a dialtone press *0 then asterisk will send the flash to PSTN line. > >regards >Martin > >On Thu, 28 Aug 2003, Carlton J.

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

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

Hello, In setting up * for my company's office and remote employees, I have a question about how to log one username into * as either a SIP account, or IAX account. For example, we will be using SIP phones in the office locally to the * server, however some employees travel, and want to use IAX (as it's much friendlier with firewall/proxy setups than SIP) clients on their notebooks.

Are there any plans to incorporate the running of Asterisk as a non-root user into the current CVS? There is nothing in Asterisk that requires root access as far as I know and this would solve the vmail.cgi script permissions problem.

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

I never did see this get to the list. Tim Litwiller wrote: > For my church school we need a way to connect 3 room phones, 1 office > phone and 2 phone lines. > so I need a device or several that i can connect to 2 pots phone lines > and at least 3 plain old wall phones. I'll donate a sipura 941 for > the office. > > What would be the best product to get 2 fxo ports

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

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

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

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

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

Hello. I am going to be putting together my first * system using FXO/FXS interfaces. All the systems I have set up thus far have been pure VoIP setups. The system I need to set up should have 3 FXO interfaces and 1 FXS interface, as well as several SIP phones. I have noticed people complaining about Digium's TDM cards - are these isolated incidents or are these cards unreliable? I intend to

I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. /carmi

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 integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only

I've been looking around for a small external sip fxo gateway, sending emails to possible vendors, etc, and can not seem to come up with anything that fits. Suggestions anyone? (No channel bank & T1 card suggestions, please. I've also just completed an eval of the Mediatrix 1204 which does not support the requirements.) The market between two fxo pstn lines (pair of x100p's) and

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 =>

On 12/14/19 11:29 AM, Greg Troxel wrote: > sean darcy <seandarcy2 at gmail.com> writes: > >>> There is also the ObiHai OBi202 with an OBiLine, which provides an FXO >>> port remoted over SIP. (I am not sure if this is discontinued.) >> >> "FXO port remoted over SIP"? >> >> I have an analog phone system. I can use the obi202 to