similar to: fxs without fxo

I'm using happily this cheap phones, but I still have a little problem. Configuring the phone is extremely easy on * and I've a couple of them perfectly working, except when i try to call some toll-free number (in italy 800xxxxxxx ). If the number called is an IVR system, often with GrandStream (but also with Cisco 7905.h323) it's impossible to make the menu choices via the Dialpad.

Greetings, I need if possibile an explanation on how to manage the incoming callerid for an incoming call. Let me explain the situation: We have two different companies in this office that shares the same PBX (* box). Each company have its own number for the incoming calls. What i'd like to implement is something that, depending on the incoming line that is involved in the call, plays a

I need to configure an * box to connect to a primary * box which is attached to a PRI in order to make calls using both the same E1 connection. I know the best solution is to use IAX (altough i could connect the IP phones on the remote site directly to the main * box that is VPNed with the remote one). I've some doubt on how to config IAX to work in this situation. I think that: On the

Greetings. Today I've installed a fresh new E100P on a EuroISDN PRI. It seems to work well, accepting calls, but, when I start *, I have the screen flooded with this message: PRI got event: 8 (or 6) If i look into /var/log/asterisk/messages i've this Error: File chan_zap.c, Line 5341 (zt_pri_error): PRI: Read on 35 failed: Unknown error 500 Repeated indefinitely. Also, i've a

I need help creating a channel for my phone device (Quicknet PhoneJack). I have installed and loaded the driver and phone devices listen in /dev (phone0 - phone15). [phone.conf] mode=dialtone format=slinear device => /dev/phone0 fxoks=2 ;Quicknet PhoneJack [extensions.conf] ... exten=>_NXXNXXXXXX,1,Dial,Phone/phone0 ... When I try to make a call, I get the following output: Executing

I've to project and build a fresh new box with * on. Basically, i'll have this situation: The office is connected to the phone carrier with a PRI. I need to let users continue to use their analog phones in a office, and an IP-phone based solution on the remote office, that will call using outside using the PRI on the first one. PRI---> Asterisk ---> Analog phones |

After a long working evening yesterday, now my * box place and receive calls with H323,SIP and ISDN line. Calling from the office to an outside line, happens: - If I call a mobile number and the called answers, all goes ok - If I call a number at home/office, and it's answered , all goes ok - If I call a toll-free number with an IVR system, nothing happens: it continues to ring indefinitely

Hi all, I bought a Sangoma A200 card from an online supplier and explained exactly what I wanted, 3 incoming phone lines to PBX and a life line (some where to connect a standard BT phone to the PBX incase the power goes, making the BT phone ring). I was told to order 1 x FXS module (2 FXS ports) 2 x FXO modules (4 FXO ports) However being a complete noob, I have connected the 3 lines to the

If you have an FXS port and would like to attach a PSTN analog line to it this device would do the job by converting the FXS port to FXO. It's a small external device. Works well with VOIP FXS and other FXS interfaces. Interface: 2 RJ11 Jacks (one for the FXS port and one for the PSTN outlet.) Cost: $35.00 with USPS regular mail included. Power: 9 to 20V dc power supply (Not

Please explain the relationship between modules from the driver (wctdm), the /etc/zaptel.conf file and zapata.conf. Specifically, if I have a FXS module 0 and FXO module 1, what should be used in zaptel.conf and what should be used in zapata.conf? Then finally, in extensions.conf, what is the Zap channel for dialing out? Zap/? % dmesg Module 0: Installed -- AUTO FXS/DPO Module 1:

Hey All, I'll be configuring an asterisk box to be the voicemail server to an old Merlin system which had an octel 100 voicemail server that is now dying. My question is simple: do I need to stick an FXO card in the asterisk box? My logic is that if the Merlin Magix system is actually generating electrical current, then I would need to have an fxo card. Is this correct?

I'm looking for a zaptel type device with one (or more) FXO and one (or more) FXS port. Basically this guy would sit in-line of your phone line (PCI card). Any suggestions? TDM400 would be overkill. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050515/d66e6335/attachment.htm

Thanks. Is there a way I can log into the Merlin Magix to determine that? How else do I tell? ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Germann Sent: Monday, February 05, 2007 8:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users]

I just received a Wildcard TDM400P by FedEx yesterday. I noticed that the FXO/FXS modules use connectors similar to Ethernet. Now, i want to connect the TDM400P to the PSTN connector in the wall, and also to a regular analog phone. Both the PSTN conn and the phone use smaller connectors, typical for analog phones. I searched the "official" docs and the Wiki, there's good

Hello everyone, I'm trying to set up a Asterisk server. I have two cards - one is an BeroNet BN2S0 with two ISDN lines (4 channels): http://www.adcomtec.com/webstore/beronet_bn2s0.php?cat=90 and a Rhino R8FXX with one FXO module and two FXS: http://www.voipsupply.com/product_info.php?products_id=2940 I would like to set up an Asterisk server with 8 phones, which will share the phone

Hello, I've recently purchased Adit 600 with 3FXS and 1FXO to be connected to my * server via T100P card. This is the output of "status equipment" command in the Adit600: For some reason the FXO card is seen as FXS, why? Is it ok? On the card it is written "FXO". Regards, Shlomi Bachar -------------- next part -------------- An HTML attachment was

we have got asterisk 1.0 (over 1 yrs old) version and very old zaptel version. That code is working only with 8 or less ports (accumulative) on digium fxs/fxo cards (2 cards with 4 ports each). the questoin is, what if we want 12 ports?..well, really, i don't understand the limitations? is it simply zaptel driver code fix? or kernel fix? or technology limitation? donno any tips would

We're having issues with the FXO/FXS ports on our Digium TDM cards sporadically. I'm wondering if anyone else has had these problems, or if anyone can provide guidance diagnosing or fixing the issue? The symptoms are that the FXO and FXS ports "stop working", usually after 2-4 weeks of server uptime. When this happens, sending a (SIP) call to an analog phone on an FXS port

Does the voicetronix card work with Asterisk? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of woody+asterisk@solutionsfirst.com.au Sent: Monday, February 02, 2004 11:06 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum > -----Original Message-----

Rich Adamson <radamson@routers.com> wrote: > >> I have been investigating on this list and found that faxing is not >> reliable between Zaptel >> cards and that Digium does not support nor >> recommend fax over the TDMXXB interfaces. >> >> Is this true ? Will fax not be reliable enough for my customer ? > > True... the TDM card (or its drivers)