similar to: asterisk box after an analogic pbx

Hi All, i'm experiencing a strange problem on sip channel. Sometime appens that the sip client ring as if it recieves 3 calls at the same time from the same number, even if thre is only a single call. I'm experiencing that both on the softphone sjphone and on the sip phone Grandstream GXP2000 an on two asterisk box one asterisk v 1.0.7 the second asterisk 1.2.16 I have not idea where to

Hi All, Is there someone who have used a Dell 1600n as fax machine? Any information or suggestion is welcome. Thank's in advance Best Regards Accursio Avona

Hi all, I'm experiencing a problem with meetme i can't resolve. This is my scenario: A iax client, say IaxComm, make a call through a zap channel. When it answers it is tranfered to a conference room. Then the iax client make a second call though a second zap channel, at the other side there is an IVR. Iax client send some dtmf to the IVR then it transfers the IVR to the previos

Guys.. Im new to asterisk but is it possible to simulate a dialtone for example, in other PBX when you pick up the phone you can hear a certain dialup, which is the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is this possible? __________________________________________________________________ Anton Krall

Hi, I'm planning to buy a new ISDN-PBX (I hope this is the right term for an ISDN phone system). I would also like to connect it to asterisk. As far as I know there is no ISDN card where I can connect an ISDN-Phone to directly working together with asterisk (please correct me if I'm wrong). So what I was thinking of doing was to get a regular ISDN PBX and add a second internal S0 bus

once u press 9 is there a way to make it so it restores dial tone, like most pbx's do? so dial tone , 9, dialtone, then ur local num -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you -------------- next part -------------- A non-text attachment was scrubbed... Name: not available

Hello all. I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Ie, pick up NEC handset, dial

Hi. I'm getting new lines for using with Asterisk. In my Telco they said I could choose between Analogic lines and RDSI lines... I've already bought a TDM400P with FXO modules. Can you give some hints on the differences between RDSI and normal Analogic lines? Would I have problems for using a RDSI line with the TDM? Any other issue in general? Thanks in advance, RODOLFO --- avast!

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

<DIV><FONT face=Arial size=2>hello All,</FONT></DIV> <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV> <DIV><FONT face=Arial size=2>our Asterisk pbx is sitting behind a normal analog hardware pbx, we have to dial 9</FONT></DIV> <DIV><FONT face=Arial size=2>to take an outside call through the hardware pbx, our

Sorry to post such a newb set of questions but I have been hammering about trying to get Asterisk running on FC2 machine reading everything available (I think that is what stuffed me, shouldn't have read it all :-) ). Config FC2 running Asterisk 1.0.0, with the h323 compiled in and installed correctly. Amazingly enough I have everything compiled correctly and installed. I am running a

Hi All, I got a doubt about something I want to do with asterisk. I have this office (site a) with only a Panasonic analog PBX and another office (site b) with an Asterisk Box with an ADIT 600 . I want to interconnect both via IAX. Is it possible to put a new asterisk box in site a without the channel bank and put a card (FXS or FXO???) and connect it to the pbx as a CO line ? What

hi .. i require to interface asterisk to a 60 line analog PBX in a hotel. I was thinking of giving Asterisk a couple of PBX lines interfaced through cards, and then place outgoing calls through SIP/H323 and a DSL connection. analog extension lines <--> analog pbx <-->asterisk <--> SIP --> termination I do not need incoming calls to the lines. My question is this : if I

I'm having trouble with a WX100USB adapter and a Siemens Gigaset cordless phone. If I select fxols as a signalling method, calls are being disconnected. Usually after about 4 minutes, and asterisk just says that the phone has hung up. If I choose fxogs, I immediately get a LINE IN USE message on my phone and I can't even get a dialtone. If I choose fxoks, it mostly works, but sometimes

Hello, I''m fairly new to Ruby on Rails. I have a model called Report and I''m trying to create a text file that is saved in a /public/report directory. I''ve had a look at the ruby api but I can''t seem to get this working. The current code I have is: #report.rb class Report < ActiveRecord::Base after_save :write_file def write_file if @file_data

Hello All, Yep I'm a newbe. I'm just started to play with asterisk. What I have Redhat Fedora Core 2 (New install) 3 X100P cards. I installed zaptel-1.0.3 libpri-1.0.3 asterisk-1.0.3 Where should I start?? -- Thanks, David -- This message has been scanned for viruses and dangerous content by KE6UPI, and is believed to be clean. KE6UPI thanks MailScanner for their support. Please

Hi, I'd like to build a personal PBX connecting 4 or 5 analogic phones with a ADSL line and I'd like to know what is the right card I need I visited digium site and I think TDM400 could be the right choice but I cannot understand how it works...I think it has 4 slots where 4 modules (FXS or FXO) can be inserted. How many cards do I need to connect my ADSL line to 5 phones? I think I

i need to know something in the zaptel configuration as it seems i can configure detecting the busy tone and hangup after number of busy tone counts, that was great but the problem is sometimes the pstn line has no dialtone and when i try to make call it continue dialing while not having a dialtone! while it should say "all lines are busy/congested" how can i configure that?? i already

OK, I have a Asterisk @ Home 1.0.7 server with two Digium TDM400 cards, one 4 port FXO and one 4 port FXS. When I plug an analog cordless phone into the TDM40B card and setup a ZAP extension, the phone rings in and you can answer just fine. The weird part is when you try to get dialtone from the cordless handset it rings all of the other extensions... One other thing to mention is that the

Hello, I created a new application for myself called Dialtone() by modifing res/res_indications.c file. It can be used as such: exten => s,4,Dialtone(30|${CALLERIDNUM}) exten => s,5,Playback(time-exceeded) exten => s,6,Goto(s|1) It will stutter if you have new voicemail and you have passed the mailbox number as I did above. It will stop dialtone the moment you press a key