[asterisk-users] adding a second extension

[Stephen Reese]

I also tried downgrading to version 1.4-current but that didn't help.

> Oh, typo, but that still didn't cure it....
>
> Successful call from from 101 to 102
>
>  == Using SIP RTP CoS mark 5
>    -- Executing [102 at default:1] Dial("SIP/101-08220318",
> "SIP/102,20") in new stack
>  == Using SIP RTP CoS mark 5
>    -- Called 102
>    -- SIP/102-08221a78 is ringing
>    -- SIP/102-08221a78 answered SIP/101-08220318
>    -- Packet2Packet bridging SIP/101-08220318 and SIP/102-08221a78
>  == Spawn extension (default, 102, 1) exited non-zero on 'SIP/101-08220318'
>
> Failed call from 102 to 101
>
>  == Using SIP RTP CoS mark 5
>    -- Executing [101 at default:1] Dial("SIP/102-08221a78",
> "SIP/101,20") in new stack
>  == Using SIP RTP CoS mark 5
>    -- Called 101
>    -- Got SIP response 400 "Bad Request" back from 68.156.63.118
>    -- SIP/101-0821e110 is circuit-busy
>  == Everyone is busy/congested at this time (1:0/1/0)
>    -- Executing [101 at default:2] Hangup("SIP/102-08221a78", "") in new stack
>  == Spawn extension (default, 101, 2) exited non-zero on 'SIP/102-08221a78'
>