similar to: Detecting answer with an analogue card

Hello, I'm following the thread "Asterisk@Now vs Trixbox", and I have a similar question: if someone is going to install Asterisk, FreePBX and A2Billing, should you advice him/her to use Trixbox ... or a custom "step by step" installation on a distribution of his/her choice? Thanks Stefano

Hello, I've discovered that in Italy ISDN lines can be programmed to generate a "billing pulse" every n seconds (it dipends from the pricebook). The pulse has these figures: frequency .................................................................... 12 kHz ? 1% level .......................................................................... 200 mVrms on 200

Hello, which kind of method could you use to inhibit long distance calls to _some_ extensions? Is there a way to do it with freepbx or you have to do it manually in the config files? I wouldn't like to set a route password, because that is not confortable for the pbx operator. I just would like the operator being able to call whatever number, while the extensions should only be able to

Hi, I'm trying to connect an HDL F10 device for a friend living in Brazil to the TDM2400 on his Asterisk server. That device should behave like a normal doorbell and it is if connected to an analog PBX. I connected to the TDM2400 and everything works fine except for one thing: when the called party hangs up his phone, the F10 HDL device does not hang up. I'm not brazilian and not

Hi List... I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing:

We have a recently deployed server with a new TDM2400 card that will not put dialtone or audio on FXS ports after the physical server restarts (though they will ring if called, there's just no audio on the line if the phone at the other end picks up). The symptom can be resolved by stopping Asterisk, restarting DAHDI, and then restarting Asterisk. So far, this has happened on both times the

Does anyone know if a TDM2400 will fit into a Dell 2850? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com -------------- next part -------------- An HTML attachment was scrubbed...

Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a

Hi, I want to do so that extension 501 will always use trunk1 for outbound calls and 502 will use trunk2 for outboud calls. How do I do this. Right now all extensions use the same trunk for outbound calls. -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, is there someone who knows if I can use AOC for billing in Asterisk? I mean: let's say I have an external SIP device that produces AOC data. This device connects me to the telco network. Can Asterisk, if connected via SIP with this device, collect AOC data and put it in the CDR records? If not, which is the right way to use AOC for billing? Thanks a lot Stefano Corsi -- Stefano

i'm trying to set up outboud caller ID on 3 polycom ip 500 phones. here is how i have them set up. each phone is using 2 lines, all registered to * phone 1 line 1 is registered using the DID, and the next 5 lines reg to 03152 03153 03154 03155 03156 from my understanding, the way the polycom configs are set, i have to put the 03152 in the reg.x.address field. and it seems that is where

Hi all, Years ago, I was pretty sure attempting to use two TDM400p cards in one machine was recommended against by Digium ... probably because the cards couldn't hack it, and/or interrupt problems etc I have seen some posts recently that seem to indicate it is in fact possible these days thanks to some updated firmware perhaps? ..... I just need to have two in the server because the 4 ports

Hi all, I have a TDM2400 full of FXS cards and a TDM800 full of FXO cards. If I have a fax machine on the FXS port dialing out through asterisk on the TDM800 FXO, should I be expecting any problems? Or should this just work as expected? (ie, flawlessly with the asterisk box essentially transparent to the whole operation). I am doing it this way to allow many faxes and modems to share a dial

now i''m read a lot of knowledge paper about traffic control in linux because now i''m try to develop and research about traffic shaper so can everybody tell me about my topic ... Is it can do for real???? my topic is " now, many organization use traffic shaper for control traffic and bandwidth because don''t want something example some mac address, ip address,

I would like to hear from anyone good or bad as what their experience has been in recent times with STABILITY of current builds of Asterisk and drivers for TDM400P. The sort of configuration is: 6 incoming POTS lines. ie. 2 TDM400P cards. I am not concerned with: price points, or the advantages or disadvantages of using POTS vs ISDN technology, but simply RELIABILITY & stability of the

I have one TDM2404E digium card on asterisk box, after configuring the zaptel and zapata configuration files, I am getting these errors when reloading asterisk: ast_unregister_indication_country: Removed default indication country 'us' setup_zap: Ignoring signalling setup_zap: Ignoring answeronpolarityswitch unable to recognize channel 13-5 what is the reason for that? Thanks,

Hi, >From a chat with Kevin Fleming during last Astricon Europe in Paris, I understood that Digium was about to support TDM bridging with TDM2400 board. This feature had to be introduced with 1.4 release. Is my understanding correct ? If correct, would it be reliable enough to bridge 14400b/s fax calls though I know, fax were not officially supported or shall I use Sangoma boards for that ?

does TDM2400 work on 3.3v pci slot? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070211/ac7d0c79/attachment.htm

Is there a way I can change the ring cadence on FXS lines on a system using a Digium Wildcard TDM2400 card? I recently deployed a new phone system, and the customer has a few POTS phones in areas where they don't have data network services, so we're using the FXS lines to provide dialtone at those outbuildings. The old phone system would ring those phones with a short ring-short

Hi, I'm having trouble with dialing out on analog lines. Asterisk can't seem to detect "answers". I have two zap groups. Group 1 is connected to an external analog PSTN provider. This group seems to work fine, especially with "answeronpolarityswitch". Group 2 is a group of "GSM gateways", ie. devices that host SIM cards so that you can dial from any PBX