[Asterisk-Users] sending traffic to LiveVoip

[Ed Greenberg]

I have several DIDs (working well) with LiveVoip and I just signed up for 
some outbound minutes. Unfortunately they did not send connection 
instructions.

I tried:
exten => 
_1NXXNXXXXXX,2,Dial(IAX2/userid:password at 217.160.244.186/${EXTEN}|60|s)

but I get the error
Feb 19 15:14:09 WARNING[21453]: chan_iax2.c:5546 socket_read: Call rejected 
by 217.160.244.186: No authority found
    -- Hungup 'IAX2/217.160.244.186:4569/1'

If I change the dialing string to
exten => 
_1NXXNXXXXXX,2,Dial(SIP/userid:password at 217.160.244.186/${EXTEN}|60|s)

I get:
Feb 19 15:15:18 WARNING[21453]: chan_sip.c:1398 create_addr: No such host: 
217.160.244.186/14082098516
Feb 19 15:15:18 NOTICE[21453]: app_dial.c:749 dial_exec: Unable to create 
channel of type 'SIP'


Unfortunately, LiveVoip does not reliably answer technical support 
questions on the weekend. I have one in, but no response as yet.

Is anybody sending traffic to LiveVoip, and what is your dialing string?

(remember to edit out your user id and password :)

</edg>