similar to: asteriskathome maximun channels per trunk

How is working the automatic fax detection? I'm making tests in asteriskathome and the ivr plays, the fax sends little bips but asterisk don't detects it as a fax. (for testing I routed one caller id to the ivr). -- Alejandro Vargas

I saw on here where there was an asteriskathome site where I could sign up for the mailing list. However when I bring up that site I just get a blank page. Is www.asteriskathome.org the correct address? Bob

i will be installing asterisk; before that i understood i need RHEL3 or CentOS and i downloaded asterisk@home1.0 already int this download page http://sourceforge.net/project/showfiles.php?group_id=123387 i have seen Download asteriskathome-1.0.iso and Download asteriskathome-1.0-md5sum.txt please anyone explain __________________________________________________ Do You Yahoo!? Tired of

dear kind helper, i would like to know how to find the annual maximun for a table that basicly looks like this: date time measurement1 measurement2 measurement3 mm/dd/yyyy hh:mm:ss m1 m2 m3 there are about 9000 measurements for each year, which makes it quite large... i already tried to subset all rows for a year, to find the maximum within these choosen rows, y <-

I want to do inbound routing of calls comming from sip trunks. Is there a way to force the DID that comes from a trunk that does not have DID support? (something like using the outgoing caller-id for the trunk?) My problem is this: I've got several sip trunks (SPA3000). I want to have an IVR in all but one of them, the one that is connected to a cellular adapter. In this line I want to let it

dear r-helpers i've got a table that in extracts looks like this: V1 V2 V3 V4 V5 1 01/01/1975 00:00:00 125.837 3.691 296.618 2 01/01/1975 01:00:00 124.799 3.679 281.307 3 01/01/1975 02:00:00 111.607 3.536 281.307 4 02/24/1976 11:00:00 21.602 2.555 93.893 5 02/24/1976 12:00:00 27.804 2.623 93.893 6 02/24/1976 13:00:00 26.105 2.604 114.716 7 10/18/1977

Hello, I am trying to make the following... Can someone tell me if it is possible? Is someone willing to do it from an asterisk@home box? 1. I send an http request to asterisk@home box. Ex: http://asterisk@home/call.php?phone=0033102030405&code=12345 2. Application will call phone number 0033102030405 (using a sip provider) 3. Application will play a pre-recorded voice prompt 4. Application

Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. Do you know any kind of phone that let me do that? Which is the maximun number of extensions your phones can monitor and which models phones

I'm configuring a sip trunk. My problem is if I configure the sip device to dial to a sip phone, it works ok but when I dials to "s" or "7777", asterisk picks up the call immediatly and places it's own ring tone instead of waiting until one of the extension configured for answer the call picks up. Is there a way to avoid it? Is it a problem of the sip trunk? Should I

Hi, I have cisco 2600 with fxo card, can we use it to connect to asteriskathome as SIP trunk..?, so we can use for incoming and outgoing trunk. Please need your help. -mendro-

?having problems with installing asterisk@home i downloaded the asteriskathome-1.5.iso file from asteriskathome.sourceforge.net link & burned it on a cd but it is not booting what seems to be the problem hoping for a quick reply -------------- next part -------------- An HTML attachment was scrubbed... URL:

As far as I can make out the root password for the ISO download is supposed to be epping or EPPING depending upon which version you are using. I've downloaded an ISO image from the following link but neither passwords seem to work :( http://ovh.dl.sourceforge.net:80/sourceforge/asteriskathome/asteriskathome-0.6.iso any one know the password for this one? -- Regards Phil -- This

Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) #make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64

Sipdiscount has replaced their asterisk servers for another thing. Then, no more iax. Ok, but I can't make calls using sip also... I'm getting a "forbidden" error when using sip1.sipdiscount.com. Anybody got it working? -- Alejandro Vargas

We have replaced the simple contact management system in Asterisk@Home with SugarCRM a full CRM system. This might seem like over kill for a home PBX but Sugar has the best contact management we have seen. With click to dial functionality and the ability to import data from other contact managers it’s a great fit for Asterisk@Home. We have also added new version of the usual Asterisk software AMP

Hi!I've 2 asteriskAtHome; How can I copy one database where are put all the sip authentificated registration to another one database on one other asteriskAthome so I've always the same Sip registrated and if one linux falls down I can run the other one without problems? Which files must I copy?then..I'll use a ssh scritp for this, I want only know which files I must copy...

I've a problem. I've some spa3000 and spa2100. Asterisk 1.2.4. Prefered codec g711u in both. Calleng from a fxs of spa2100 to the fxo of spa3000, all works ok. Then I call from a sip phone configured for using g729, to the fxo of spa3000, it also works ok. The problem is that after this, when, making again a new call from spa2100 to spa3000, spa2100 receives only white noise. I suspect a

Hi, Problem: When talking to someone (from pstn) and this person is not talking loud, the voice is cut by Asterisk. I tried increase RxGain but it changed nothing (was talking louder but voice still cut.) I use XLite as soft phone. I think this is probably a codec setting... but how do I check that on server side? I just don't know what to do. All works fine (asteriskathome) but I always

Hi I have installed the http://asteriskathome.sourceforge.net/ with a Digium card with no problems, very good ! Now I want to install my Billion PCI ISDN card (HFC based) in TE mode. I get a little confused with Isdn4Linux, ZapHFC HIAX and the need to install Capi ! Please suggest best and easiest approach ? Thank you ! HB Norway

Hi all, Newbie quest here. I have 3 PSTN lines (2430-2432) setup by the CLEC in a hunt group coming in to a TDM04B and 3 Grandstream gxp-2000's (say a,b and c) and Asteriskathome installed. I want all phones (a,b,c) to be able to take calls from the hunt group. Does this mean I must make 9 SIP extensions on * ? 3 for each IP phone? and group them in trtiplets (ring groups?)