similar to: automatic fax detection in asteriskathome

I'm using asteriskathome 2.5. I'm using 2 spa3000 for dialing-out. I configured a trunk for each one with maximun channels=1 and an outbound route that includes both trunks. When a second outgoing call is placed, Asterisk tries to place it in the same that is already in use resulting in a busy tone. ?What can be the problem? -- 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'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

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

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

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

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:

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

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 all ! Let's say you have about 6 small different companies sharing the same E1 / Asterisk server, and every company needs its own fax number. Since they don't really need fax machines, what would be the most cost-effective way to handle this (keeping fax-privacy at its best) ? Is there a way to configure Hylafax or sth & one modem behind an ATA-186 to email faxes to different

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

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

At the same site, DTMF recognition is functioning badly, sometimes duplicating digits and sometimes totally missing others. We have checked already /proc/interrups, there is no interrupt being shared. Our zaptel has .. span=1,1,0,ccs,hdb3 On zapata we have ... switchtype=euroisdn signalling=pri_cpe relaxdtmf=no (yes doesn't seem to help) -----Original Message----- From:

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...

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

How to read a xml file? I have this XML source: ------------------------------------------------------- <?xml version="1.0" encoding="UTF-8" standalone="yes"?> <fichas> <ficha> <nombre>Gabriel</nombre> <apellido>Molina</apellido> <direccion>Alfredo Vargas #36</direccion> </ficha>

Hi All, We are just starting to deploy a new PRI IVR system, and the incoming calls sometimes get random short 'bips' while navigating our IVR menu. Any hint on what this can be? Best regards, PauloHM

Felipe, gracias por tu interés. La sesión quedó grabada y se colgará en breve en la página de R-es. Ya avisaremos cuando esté colgada. Un saludo, Aleix El 10 de octubre de 2013 00:01, Felipe Vargas Reeve <kaiser756@gmail.com>escribió: > Esta fue una gran iniciativa y tenía muy presente el tema relacionado a > la transmisión por internet, pero por motivos de trabajo no pude

In the attached plot, how do I move just the X-label(Time) to the right. Basically, I just want to do a par ( adj = 1) on just the X-label, not all my labels. Any ideas how to accomplish this? Thanks, Tony Tony Vargas Cisco Systems Engineering Computing Services (408) 525-4113 tvargas at cisco.com -------------- next part -------------- my.yugo.Apr.2003.other.info <-