similar to: FXO Line not Hanged up

Dear All, I've got a weird problem with my asterisk box which has fxo interfaces (TDM400). Well, the problem is that the interface answers the call, but no caller id is being received. Also, sometimes this error happens: fsk_serie made mylen < 0 Any idea what is going on? Thanks, Kaveh __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has

I had the same issue at one stage although it was with call files - check what your WaitTime is. Mine was set to 5 which means 5ms so only half a ring and then it would hang-up. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of fhunter Sent: Thursday, May 12, 2005 1:58 PM To: 'Asterisk Users Mailing List -

Never mind. The Asterisk@Home documentation is incorrect the echotest is in *43 and it works fine. -----Original Message----- From: fhunter [mailto:fhunter@survivorsoft.com] Sent: Thursday, May 12, 2005 4:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Incoming calls picked-up then simply hanged-up Something else I have noticed when

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 have been looking for the answer to this question for a while. Google-ing and reading the archives of Asterisk-Users has not enlightened me. It seems that this question has been asked many times, and many times it has gone unanswered. I have call waiting and three way calling on my PSTN line from Verizon (the local telco). This is connected to a FXO port on a TDM400P. I also have

Thanks Rich, I have an SPA-3000 laying around, so I will attempt to set it up in a little more conventional manner (although your method looks like a winner for a home test PBX). Would you mind posting or PM your current config to me, maybe screenshots if you PM. If I start with that it will take less time to get to the point where the SPA-3000 is a true FXO-FXS gateway for *. I will be happy to

Hi, I have HT488 with it's FXO connected to Israeli PSTN (bezeq) when dialing to that PSTN line asterisk see gets the call and direct it to the right extension but if the extension doesn't answer and the dialer is hanging the call the extension will keep on ringing. I'm not an expert but it seems like my asterisk doesn't recognize the hangup signal from the HT488 -or it's the

Voxilla.com has a great config wizard for the SPA-3000 and * http://voxilla.com/spa3kasterisk.php I took the output from this wizard and dumped it on my test box with an SPA 3000 (with some mods to match my * contexts) and everything worked great. Calls from the PSTN to the spa3000 are routed to dialplan #8 on the spa3000, which dials * Both the FXO and FXS port register with * The SPA3000 is

Hi all. I am unable to answer calls coming into asterisk over PSTN. (UK) I want to have a call answered by my TDM400P/FXO module and forwarded to a sip phone. When I make a call from the PSTN to the BT line installed on my FXO module the sip phone rings however, when i pick up the call using the sip phone, the incoming call is not answered/routed by asterisk. As a result the sip phone is left

I am currently having our new office wired up with 8 PSTN lines. The guy asked me if he could wire it up such that one line had two phone numbers. I bought a Sangoma A200 with 8 FXO ports, but now I'm wondering if all I needed were 4 FXO ports. Is it possible to set up Asterisk with 2 numbers per FXO? Thanks for any help, Andrew -------------- next part -------------- An HTML attachment

Hello, I'm trying to configure Asterisk and my Handytone 488 to pass incoming calls coming over PSTN through the FXO port to Asterisk, which will process the calls with voicemail, or some such service. I point the Handytone 488 FXO port configuration to 192.168.0.2 (the machine that is running Asterisk) and have configured a catchall extension to receive the call: [from-pstn] exten =>

For some reason, this didn't go through the first time, maybe because I had JUST signed up. Hello, I'm trying to configure Asterisk and my Handytone 488 to pass incoming calls coming over PSTN through the FXO port to Asterisk, which will process the calls with voicemail, or some such service. I point the Handytone 488 FXO port configuration to 192.168.0.2 (the machine that is running

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

Hi, every one I'd like to introduce some new feature of our products. mg3000-r fxo gateway provides more feature to work with asterisk. 1.play asterisk ivr with no interuption. when the mg3000-r received call from co line, it wouldn't conect instantly.instead, it start call to asterisk ivr first,when the ivr ready, it connect the co line. this feature make user feel friendly.

Hello, I am facing hangup delay problem using FXO (X100P) card with the following scenario: PSTN Phone--------->BTTB PSTN Gateway------->Asterisk BOX-FXO-------->SIP Endpoint When anyone make call to my Asterisk-BOX FXO, they gets IVR to press their desired SIP extension number. If calling party disconnect the his/her call before pick up the call by the called party, then called

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.

Just installed the new 4-port FXO card and moved two pstn lines from existing x100p cards to ports on the FXO card. All zapata.conf entries that were functional on the x100p's were copied to the new FXO channels (including callprogress=no). Observations thus far: 1. asterisk will spontanously decide a pstn call has arrived, and ring the sip phone designated in the dailplan. Verified

I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN

Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an

As far as I can tell, after discussing this matter with other asterisk users in my area, my telco _does_ provide disconnect supervision.. It seems that the problem is actually related to the Sangoma A200 card I'm using, as two other people both using this same card have expressed the same problem.. Are there any other users on this list using the Sangoma A200 FXO port card, and experiencing