search for: facility_reqs

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to get callers forwarded to by mobile phone when they dial a certain digit. In my extensions.conf I have defined the following: [279xxxx] exten => s,1,SetLanguage(de) exten => s,2,Wait,5 exten => s,3,BackGround(demo-congrats) exten => s,4,Goto(boksa,#,1) exten => 3,1,VoiceMail,u1 exten => 4,1,VoicemailMain exten =>

>>0.4.0? who is working on chan_capi? long time with no updates from. >>I'm rewriting the chan_capi logic. >> >> > >kapejod already did 0.4.0-PRE1, which I use as base for cleanup, fixing and >new features. > >What exactly are you rewriting? If someone else is working on chan_capi, we >should synchronize our work and ideas. > > yes

Hi, although setting up voicemail for busy and unavailable should be easy, things aren't working the way they should in my configuration (asterisk 1.2.14 bristuffed): Here's the relevant part of the extensions.conf: exten => 56830976,1,Answer() exten => 56830976,2,Dial(SIP/hbaumgart,20,tr) exten => 56830976,3,VoiceMail,u76 exten => 56830976,4,Hangup exten =>

I've the following setup : sip phone -> ser (auth and routing) -> asterisk with capi isdn when I call a pstn number everything works fine, but I can't hear anything till the called answer. this is the output from a test call : -- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack -- Executing Dial("SIP/2.7.184.61-08152880",

Hi, I've got Asterisk configured and working (sort of) with an Eicon Diva Server 2M ISDN card (connected to S0 bus of another PBX). This * box is on a 'live', non-nat IP address. I also have a couple of budgetone phones, one behind NAT and one not. When I place an outgoing call, I get the following messages: -- Executing Dial("SIP/filbert-9876", "CAPI/288:333") in

In the capi.conf file, there is a bridge option that allow to "native bridging (CAPI line interconnect) if available", and I found this in the capi-user mailing list : -------------------------------- I suggest you put bridge=yes into each interface. Then, when Asterisk bridges two channels, it looks for the possibility to do a native bridge (call the bridge code of the channel

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

Hi all, does anyone have capiECT working? I can't get it to work - please see my logfiles below. While using an * CVS version of late September and chan_capi-0.2.5 (I guess), it worked!!! (I know, never change a running system ... should've backuped ... etc.) Here's the setup: NT----AgfeoISDNPBX----AgfeoISDNPBX----AnalogPhones in between the two PBX I attached my * server with a

Hello, I tryed out spandsp with libtiff-3.5.7 and with Asterisk from CVS. I tryed to receive a fax on a CAPI channel. Finally I got a file with 8 byte length (/tmp/testfax.tif). How can I do next? Thanks in advance, Thomas ps: what are hardware requirements for sending/receiving faxes? -- creating pipe for PLCI=0x101 msn = 2258589 > sent ALERT_REQ PLCI = 0x101 --