similar to: Check for a phone off hook

Hello. After doing much web searching and searching archives of this mailing list, I see that my question has been asked at least 6 separate times but no answers have been attached. In a nutshell, is there a way for a SIP phone to easily hook flash a ZAP analog trunk mid-call? (This is important when trying to make use of features on a PSTN analog line such as call waiting, call forwarding,

Asterisk version: 1.2.0-beta1 -OR- CVS HEAD Hardware: Generic X100P clone connected to Panasonic KX-TA624-5 extension port The problem happens IF AND ONLY IF: - the Panasonic is set to use double rings - the X100P is set to answer immediately (I'm using DISA here) It does not happen consistently; sometimes Asterisk behaves normally. When the "strange state" happens, DISA usually

Hello fellow * users, I've been experimenting like a madman lately with asterisk, and I just love it. Just reading this list and asking a few questions here and there has helped me out a great amount. Not to mention the excellent resource we call the Wiki. I have searched for answers to the two following questions, but couldn't find anything, it is entirely possible i was just

If Asterisk is configured as a simple answering machine replacement with the X100P connected to PSTN line. No FXS ports in the Asterisk machine. Standard phones are connect in parallel with the X100P like you would a regular answering machine. Can Asterisk detect that a phone has been picked up and cancel the outgoing message and/or voice recording? What about if the phones are connected to

Hi there, I have a x100p card in an asterisk box. Does anyone know if it's possible to do a hook flash / recall on an active zap channel? On what I'm trying to do... >From an ordinary analogue pstn telephone I can call someone, press recall, call someone else, press recall 3 & presto we're on a three way chat, with me only using one line - using the telephone company's

Hello, I am seeing this error: Jan 24 20:02:53 WARNING[1175660480]: Ring/Off-hook in strange state 6 on channel 3 on one of my phone lines that comes into my channel bank. It only happens on one line, and it still seems to work. Also, I have had two different channel banks installed (ADIT 600 & Adtran TA 750) and see the same error message on incoming phone calls on that line. I don't

I have the need for a slightly odd * configuration for testing purposes. I have a working * setup with SIP softphones, VoIP trunks and a single X100P clone for PSTN access. The PSTN line I'm using for testing is also in use by other folks. For incoming calls, I'd like to set is up so that * functions as a voicemail backstop on this line. This much is working fine. For outgoing, I'd

Hello- Sorry if you see this again, I am new to this list and accidentally put this message in another thread. What I want to know is, is there any way to send a hook-flash signal from a cell phone (and then have Asterisk pass it up to the PSTN?) I suspect we need an example. I have an X100P to the PSTN and an S100U to an Analog Phone (ext. 3100) Let's say my PSTN number is 555-1296.

Hi all. I`ve installed a X100P on my box and is working well with incoming and outgoing calls as a trunk with one PTSN line. I want to connect the X100P to my Panasonic 616 EASA-PHONE as an internal extension to permit to users to make calls to SIP devices from analog phones, the problem is when I dial the ext number where the X100P is connected I get busy tone. What config I need to change to

Hi, I'm trying to get my X100P to Dial the following sequence to gain access to speed dial numbers on my Norstar PBX that the X100 is connected to... [FLASH] [*] [0] [22] (where 22 is the speed dial number) But so far I've had no luck, with the following extension:- exten => 922,1,Flash(${DIALOUTANALOG}) exten => 922,2,Dial(${DIALOUTANALOG}/*022) exten => 922,3,Congestion

Not sure if anyone experienced the same - or if anyone ever connected a POTS phone to the "Phone" jack on an X100P card. The POTS phone rings normally when the FXO receives a call. The POTS phone can also make outgoing calls when FXO is not holding the line. This is desired. But if a call connected to the POTS phone lasts longer than a couple of minutes, Asterisk would receive

Hi. I have a x100p card installed on my asterisk box... my zapata.conf file includes the following lines: [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 echocancel=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 Basically, the zapata.conf file generated by make samples. Then in my extensions.conf I have this: [default] include => demo And demo is

Is this supported? I have a very simple setup where I have 2 X100P cards and a TDM10B. The TDM10B is connected to a phone that has a digital answering machine built into it. If I make an inbound call on either X100P interface it gets transferred to the TDM10B interface. If I let it ring the TDM10B interface answers the call and the greeting message of the answering machine starts. Then shortly

I am currently attempting to get Asterisk working with our existing NEC IPK 192 system. The * box has a Wildcard X100P clone card in it and is hooked to an analog port on the NEC system. The analog port is shared between an elevator phone and a credit card machine. I have one SIP extension configured in * which the softphone on my Windows PC is registered to. With the softphone I am able to

What we would like to see happen or emulated is that if someone makes a call via our SIP provider to a PSTN number that is actually busy that we get an actual BUSY tone at the telephone. In our test case this is a PAP2-NA SIP device It would appear that when we call the far end (PSTN phone number) that is busy we do not get any busy indication at the user end (originating telephone on our

I am seeking to implement the following behavor: When a headset on phone1 is picked up, phone2 rings right away, without any need to dial numbers on phone1. Is this possible to implement? ScriptHead -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060120/0892441d/attachment.htm

f it is on a zap device look at immediate=yes if on SIP check on configuratio of phone. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Script Head Sent: Friday, January 20, 2006 12:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] How to have a phone

On Fri, Jun 19, 2015 at 2:14 PM, asterisk <asterisk at solutionengineers.com> wrote: > > Hi, > > Long story short - I have an ancient Britsh Telecom phone attached to my > Asterisk PBX via Dahdi. It works beautifully, receiving calls, and the call > quality is excellent. However, dialling out is impossible, as Asterisk > consistently mis-reads the number of pulses the

Dear asterisk-users, I'm trying to make the windows messenger 4.7 work with Asterisk. I have a X100P. I want to make phone calls from messenger and through * to a PSTN regular phone. After I read all the FAQ and mailing list archives messages about MSN, I can not let it work. My configure files are modified following the FAQ. The * is as fresh as yesterday's version. I can register

Hello, I recently purchased a Wildcard X100P FXO Card primarily as a learning tool and to provide myself with voicemail and a few other perks that are better than an answering machine. I followed the setup in the FAQ, and it works, just not well. The sound is very choppy and staticy whenever a call is placed. Sometimes it clears up, and sometimes it degrades very badly. I also recently