search for: undrhil

...ftphone since it only requires one port to be open into my firewall, but kiax is the only one I've found for free. Once I get Asterisk up and running properly, I'll probably buy a couple of licenses for a good quality softphone, but until then, free is the best for me. :) So, any ideas? Undrhil

..., I'm starting over at square one and I am going to download plain-jane Asterisk and get it running on a Knoppix HD installation... I hope. :) Anyway, this has been a brief (trust me, brief is good!) introduction of myself to the group. I'm sure I'll be asking lots of questions. :) Undrhil

I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten => s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the

I have asterisk running more or less ok but I would like to turn off call waiting and be selective about the incoming sip connections. This is running asterisk 1.2.8 with a fxs and fxo card and a configured voip (sip) line. Currently I'm using freePBX 2.1.1 to configure asterisk. Problem 1) if someone is on the phone already and another call comes in for an already engaged extension I

...asterisk to auto-dial an extension when I go off-hook on another one. For instance, I have two SIP phones, Brian and Susan. When Susan goes off-hook, it should automatically ring Brian, without any other activity taking place. If Brian goes off-hook, it acts like a normal extension. Any ideas? Undrhil

...I configure the boxes to use each other's lines as local? In other words, let's say Site A has Phone1 for a 1FB line going into it on an FXO port. Site B has Phone2 for a 1FB line going into it on an FXO port. Is there a way to configure Site A to use Phone2 from Site B and vice versa? Undrhil

Asterisk handling My Skype Calls This is for me, once more, Asterisk as the Future of Telephony. Today I've integrated my Skype Account as SIP extension in my * Box. This has been possible using "Uplink Skype to SIP Adapter", available for free at http://www.nch.com.au/skypetosip/index.html . Main features that any one can easily integrate into Asterisk: - Route skype incoming

...n the dial-pad through the VoIP line. Is Asterisk capable of doing on-the-fly transfers like this? I'm sure I'll think of more scenarios eventually. I work in the telecom industry and I like that Asterisk can do what our top-of-the-line phone system can do at a fraction of the cost. :) Undrhil

...ons before buying anything. I know that sounds like an odd way to do things: research it before buying it, but what can I say? :) Besides, this is all still theorhetical for me. I'm still working on getting a FXO card for my home phone line so I can get this all configured for testing. :) Undrhil

Hi, voicemail are working ok, I receive message as attach via email. My question is : how can the user call asterisk and listen to his voicemessages ? thanks Victor

...from the public internet, is it possible for someone with a SIP client which is not on my Asterisk box to be able to dial something like (myexten@mydomain) and get an extension on my box? If so, what would I need to do to enable it or does it come that way "out of the box", as it were? Undrhil

...conf and the FXS port rings and I can answer it. However, I cannot dial out through my Asterisk box on it. I need to get this part working before I even think of trying to put my dial tone on the FXO port. So, has anyone use one of these and might have some kind of documentation for it? Thanks Undrhil

Hi All, I'm looking for a good voip hardphone that has a decent set of the "regular" features (conference, 2 lines, etc) thats reliable, has decent quality, and isn't too pricey. Does anyone have any suggestions? Thanks in advance. Derek -- Derek Fedel

Does anyone know where to find an example or able to provide an example of how to do the following: When asterisk answers a call... Ask for number to dial...then dial that number? I am basically dialing into the asterisk box and then wanting it to take the digits I enter and dial them on an outbound zap trunk... I basically am just not sure how to have asterisk accept the digits and then use

I am trying to perform a "rollover" when the primary number is busy. This is coming from a POTS line. Apparently I need call waiting on the POTS line as I get immediate busy from the FXS if I don't have it. So my question is this. I have an Aastra 480I CT. The call forward when busy here seems pretty straight forward. Choose the mode as busy enter the extension in the forward number

I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip

I am trying to modify a fairly complex digital receptionist dialplan that has a number of included contexts. Right now the system is not announcing the extension that the caller attempted to dial, so callers get confused when they think they dialed a valid extension but asterisk didn't pick everything up. I would like to have the system announce the entension that they attempted to dial in

Just a Question. I would like to know if TDMoE follows specifiaciones of TDMoIP RAD protocol that says that there is a compression of 16/1 when you do TDMoIP. Manuel Marin Garcia TRANSTELCO S.A. DE C.V. Campos Eliseos 9050 B4 – Cd. Juárez, Chih. 32452 - México Oficina: +52 656 692 11 09 – Fax: +52 656 692 1112 - Celular: 915 727 6141 http://www.transtelco.com.mx

Hi, Could somebody help me this SIP trasport? I'm receiving SIP "invite" with CLI of calling party from the SIP gateway, aster that my IVR has to answer the call. sip.conf: ========= [general] port = 5060 bindaddr = 0.0.0.0 context = incomingsip videosupport=yes ; Turn on support for SIP video disallow=all ; Disallow all codecs allow=g729

The USB core was completely rewritten in 2.6, and as such the functions that ztdummy depends on do not exist in 2.6. I get the feeling that these changes are too much to easily fix ztdummy, so I don't expect to see it working on 2.6 any time soon (if ever) I made some small changes to zaprtc to work on 2.6 and I have MoH and Meetme functions working fine in my lab. For production I would