similar to: did from sip trunk

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'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'm trying to get working a spa3000 with asterisk. My problem is I cant get wroking the incomming calls (I installed the lastest firmware). My problem is asterisk is rejecting the authentication from the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I think I placed the username and password correctly... Sip.conf says this: [linea2] username=linea2 type=peer secret=1111

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

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

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

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

Can Asterisk @ home receive incoming call using a external modem? Thanks Gidean Chan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060315/213e9820/attachment.htm

Hello all ... mY first ever post in here. I am bit or (full) confused on what this program does.is it useful if i have a alcatel pabx system.And i can bill my guests for their call charges etc.. can i use it on calling another computer on the network via Ethernet card.Ihave already read the Documentation,But if any one could clear me up on the above things. how can i call a regular PSTN landline

Hi All, I have two Asterisks, one on the LAN that handles the internal calls with a PSTN interface and one on the DMZ with a public interface. I would like to route SIP calls from the internal users to the Internet over IAX2 to the DMZ and onwards. All users have soft phones so they would enter sip:someuser@somevoip.org to get a connection. I would like to avoid having number prefixes to dial

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. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten => 123,2,Dial(SIP/3000,,wW) ; wW allow one-touch recording During the call, I press *1 but it records nothing. David Morrow

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

Hi all, I just set up a small asterisk box at home and it works as expected, I have some relatives in USA, I usually use skype to speak to them, is there anyway to connect skype and asterisk. I'd like to use skpe as an extension or a channel with asterisk. Thanks in advance for any suggestion. adriano. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there

More an echo algorithm question than a purely asterisk one... I have the following setup: Handset - PAP2 - Asterisk - SPA3000 - Telco And no matter what I do, I get echo on a call routed out via the PSTN when I talk into the handset, in the order of a hundred ms (my estimate, could be wildly inaccurate!). Echo will occur also when I have a handset plugged into the phone port on the SPA3000

Hi, These days I run into something quite odd. I have an A@H that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset <--> (FXS)SPA3000(FXO) <--> PAP2 And set the Line1 dialplan on the SPA3k to '(<:@gw0>S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with