search for: chuks

Displaying 6 results from an estimated 6 matches for "chuks".

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2005 Feb 21
2
Why can't I make inter IAX calls between 2 Asterisk servers
...iv><FONT size=2></FONT>&nbsp;</div> <div><FONT size=2>A's extension.config</FONT></div> <div><FONT size=2></FONT>&nbsp;</div><FONT size=2> <div>[internal]</div> <div>exten => 3000,1,DIAL(IAX2/chuks:password@192.168.1.107/3000)&nbsp;</div> <div>exten => 3000,2,congestion</div></FONT><FONT size=2><FONT size=2> <div>include -> from-iax</div> <div>&nbsp;</div> <div>[from-iax]</div></FONT> <div>exten...
2005 Feb 20
7
bridging iaxtel calls to PSTN
...to make a call to local US number (where A is located) from B, using iaxtel. Can anyone please help me? All I have seen so far is just making calls from A to B and vice versa using the iaxtel 1700 number, but I haven't seen any examples of how to bridge the iaxtel calls to PSTN. Help please. chuks. NB: I don't mean toll free number, I mean just local dialing.
2005 Feb 20
3
Digium TDM400P has RJ45 interface, how to connect to analog phone RJ11?
...card, I also intend to bring in an analog line into the RJ45, so i am still left with the same question....how do I go from the RJ11 standard analog to the RJ45 on the TDM400P card? I'd appreciate any response.</div> <div>&nbsp;</div> <div>thx</div> <div>chuks</div>
2005 Feb 21
1
why can't I make toll free calls via IAXTEL
...es loop, and the only output I see is a "timeout on Zap/1-1" and it tries the whole thing again...and goes on forever, and the call never goes through. Is there anything wrong with my configuration above?</DIV> <DIV>&nbsp;</DIV> <DIV>thx,</DIV> <DIV>Chuks.</DIV></FONT></BLOCKQUOTE>
2005 Mar 21
4
Can't hear the caller
Hi, I've got a strange issue, that I haven't found addressed on the wiki. My asterisk box is behind a firewall which routes udp/tcp requests on 5060 and 8000 to asterisk. When I make a call from a Zap or SIP extension on the inside of the firewall to any Zap or SIP extension on the inside of the firewall, everything works find. I have access to voipjet, and when I place a call
2005 Mar 15
0
RE: can't hear anything on my side during a SIP call
...e. Can anyone assist? I believe I have some asterisk configuration set wrong. What's the minimum i need (configuration wise) to successfully make a SIP call? NB: my extensions.conf for outgoing is defined as exten => 9NXXXXXXXXX,1,Dial(SIP/001${ENTEN:1}@voipuser thx, chuks.