the reason this is happening is because you have qualify = yes for the extensions you are trying to use. This makes the asterisk box send out the "OPTIONS" to the extension. Basically it is a form of keepalive. If the softphone dioesn't know how to answer this, then asterisk doesn;t get a reply, and assumes that the extension is down. So you can only make outbound calls from it. If you look at tthe asterisk info page (Trixbox) you will see "UNREACHABLE" for the status of the extensions. Just simply turn i t off and you will be all set...

the reason this is happening is because you have qualify = yes for the extensions you are trying to use. This makes the asterisk box send out the "OPTIONS" to the extension. Basically it is a form of keepalive. If the softphone dioesn't know how to answer this, then asterisk doesn;t get a reply, and assumes that the extension is down. So you can only make outbound calls from it. If you look at tthe asterisk info page (Trixbox) you will see "UNREACHABLE" for the status of the extensions. Just simply turn i t off and you will be all set...

Can i use sip communicator for a conference call? ( one to many or many to many ) client?
or i need a server(?) to make a conference call?
Does S.I.P. provides interface for conference call?

:)

Hello. I have the same problem.
I can establish calls in both directions but when SIP Communicator is conneced to asterisk, it immediately brings up a strange error dialog box message which looks like this:


SipCommunicator's SipManager didn't know how to handle the message OPTIONS in the current c ontext! (See Details)

Has anyone solved this?
Thanks

Hello. I have the same problem.
I can establish calls in both directions but when SIP Communicator is conneced to asterisk, it immediately brings up a strange error dialog box message which looks like this:


SipCommunicator's SipManager didn't know how to handle the message OPTIONS in the current c ontext! (See Details)

Has anyone solved this?
Thanks

beccon do you fix your problem?

Hi..

I could get it to work but I'm only listening one-way voice. I mean, others can hear me, but I can't hear them. Is this have to do with asterisk?

Cheers

Has anyone successfully run SIP-Communicator at an Asterisk system? I can establish calls in both directions but when SIP Communicator is conneced to asterisk, it immediately brings up a strange error dialog box message which looks like this:<br><br>

<i>SipCommunicator's SipManager didn't know how to handle the message OPTIONS in the current c ontext! (See Details)<br><br>

OPTIONS sip:192.168.0.1:24075;transport=udp SIP/2.0 <br>
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK02215330 <br>
From: "asterisk" <sip:asterisk@192.168.0.1>;tag=as6faf003b <br>
To: <sip:192.168.0.1:24075;transport=udp>;tag=-1239673351 <br>
Contact: <sip:asterisk@192.168.0.1> <br>
Call-ID: 710f3ebd16d4fa970235f07a530eacdc@192.168.0.1 <br>
CSeq: 102 OPTIONS <br>
User-Agent: Asterisk PBX <br>
Date: Mon, 03 Oct 2005 10:52:05 GMT <br>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER <br>
Content-Length: 0 </i><br><br>


Unfortunately this box keeps returning annoyingly every 20 seconds, making the use of
sip-communicator with almost impossible.<br>

Conrad

i want the package voip.router for java