search for: batphones

...e R4T1, as well as 2 PRIs from our telco. The CB24s are for all internal analog phones. Most of the phones are setup in "batphone mode", which is "immediate=on" in the DAHDI config. They are set up this way because we are an outgoing call center, and the context that the batphones go to a database table to pull the phone number they are calling. Along with this new hardware, we changed to a new "server" (just a Dell E520 workstation with 4 gigs of RAM and 2 250 gig SATA drives software RAIDed) with the following software: Ubuntu 9.04 Asterisk 1.6.1.4 Asteri...

Guys. Anybody using sipuras 2002 knows if there is a way to make the phones connected to it to autodial an extension when the phone is picked up? For example, if the phone is on a police booth (building entrance) and you want the guys to just pick up the phone and make the phone auto dial the receptionist extension without the guys having to dial anything (ala batphone). Is this possible with

Does anyone know of a way to make a standard analog phone plugged into an FXS port do something other than get a dialtone when you pick it up? For example, if the phone should automatically ring someone or play a greeting when picked up without having to enter an extension? - Robert -------------- next part -------------- An HTML attachment was scrubbed... URL:

My setup is this: Analogue phone attached to a Linksys PAP2 | Asterisk | VoIP provider I have put the PAP2 in 'batphone' mode where when you pick it up it immediately dials the 's' extension in the pap2_incoming context in Asterisk, where asterisk answers the call and does a DISA(no-password, internal). I do this because it means I can centralise all of my dialplan logic in

I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Asterisk. My hope is that I can let Asteirsk handle the entire dialplan, including dial tone generation. What would my context in extenstions.conf look like for this sort of dialing. More accurately, how can I get Asterisk to generate the dial tone on

How can I let asterisk immediately dials a trunk when off hook?

Hello For users who 1) don't have a QoS-capable ADSL router and 2) would like to run Asterisk with a couple of SIP trunks, I was wondering what hardware is recommend to run any of the main open-source *WRT projects to which Asterisk has been ported: (http://en.wikipedia.org/wiki/List_of_wireless_router_firmware_projects Thank you.

This might be a newbie question... I'm using a SIP trunk and trying to get DID line information on an incoming call. All I hear is a nice lady saying 'Zero' - then the call continues... Any suggestions? thanks Todd exten => s,n,Set(DIDID=(<${FROM_DID}>)) exten => s,n,SayNumber(DIDID) or exten => s,n,Set(FROM_DID=${EXTEN}) exten =>

My analog card, uses a PCI slot and a 12V power connector, is configured with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS ports but I can't dial out from them. Is extensions.conf where I need to make changes? [root at robin asterisk]# cat chan_dahdi.conf [trunkgroups] [channels] [phone](!) usecallerid = yes hidecallerid = no callwaiting = yes usecallingpres = yes

Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm running Asterisk@home 2.5 asterisk 1.2.4 zapatel 1.2.2 libpri 1.2.2 on a Dell Poweredge 2850 (1 CPU) with a TE210P I have 2 t1 circuits using NFAS with dchan on 24 and no backup dchan. I am able to receive inbound calls on all channels and can only make outbound calls on channels 25-48. Attempting to make an outbound call on channels 1-23 results in congestion.