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I strange error messages when using mgcp and ata186 . This session is simply dial into 600 demo extension - echo test ... Handling request 'NTFY' on aaln/1@10.0.1.19 Transmitting: 200 29 OK to 10.0.1.19:2427 -- Endpoint 'aaln/1@10.0.1.19-1' observed '0' -- MGCP Asked to indicate tone: on aaln/1@10.0.1.19-1 in cxmode: sendrecv Posting Request: RQNT 306

I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk. Can I dial from asterisk into ata, then indicate phone number playing tone (use DISA feature at panasonic) and connect to any analog phone connected to panasonic ? I think some of Playtones application within Dial application can help me. But I don't know how. -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia

Hello everyone, Pleace can any one tell me how I can install an USB driver into Wine. I want to connect to my burglar alarm with "Comlink". But "Comlink" only works with Windows and the USB driver is not checked for windows. When I install Comlink on a windowsmachine everything works. please can anyone help me Boswolf

I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and asterisk as pbx. I need feature called as 'three way calling' or 'transfer with consultation'. Registering,calling and 'blind transfer' work fine. Is this feature provided by sip clients or by asterisk itself ? What I have to configure in ATA and what keys I have to press on my phones ? Three way calling is

Does asterisk work with CCM as gateway ? When I trying call asterisk,I totally can't hear any sound. When call ohphone - works good. 10.0.1.219 is CCM, 10.0.1.207 asterisk. Trace messages here : -------------------- == New H.323 Connection created. -- Received SETUP message... == Setting up Call -- Calling party name: [5001,] -- Calling party number: [5001] -- Called party

Hello all, I'm new to the list, but use VoIP and * for a little while now. Running Asterisk 1.0.2 on debian linux I'm facing the following problem: I've got two Fritz!Box Fon Adapters (kind of ATA's) with two hardware phone connectors each. So I'm trying to set up a PBX with four internal (SIP) phones. One box has fon1+2, the other fon3+4. When I start up *, everything

Hi, I've got attended (superivised) transfer working with a handful of SIP phones, connected via different ATA's to an Asterisk CVS-D2005.05.28.22.00.00-07/12/05-20:47:08. pingu*CLI> show features Feature Default Current ------- ------- ------- Pickup *8 *8 Blind Transfer # ** Attended Transfer