[asterisk-users] REALY strange issue with making calls biside 2 phones

[C F]

what does sip show peers say?

On Tue, Jul 12, 2011 at 12:40 PM, Matiss Jekabsons <matiss at jekabsons.lv> wrote:
> Thats my issue, i hope someone could suggest something:
>
> Phone A -> Phone B
>
>
>
> == Using SIP RTP CoS mark 5
>
>    -- Executing [000001 at default:1] Dial("SIP/000000-00000076", "SIP/000001")
> in new stack
>
>  == Using SIP RTP CoS mark 5
>
>    -- Called 000001
>
>    -- SIP/000001-00000077 is ringing
>
>    -- SIP/000001-00000077 answered SIP/000000-00000076
>
>    -- Locally bridging SIP/000000-00000076 and SIP/000001-00000077
>
>  == Spawn extension (default, 000001, 1) exited non-zero on
> 'SIP/000000-00000076'
>
>
>
>
>
>
>
> Phone B -> phone A
>
>
>
>  == Using SIP RTP CoS mark 5
>
>    -- Executing [000000 at default:1] Dial("SIP/000001-00000078", "SIP/000000")
> in new stack
>
> [Jul 12 19:08:35] WARNING[2965]: app_dial.c:2039 dial_exec_full: Unable to
> create channel of type 'SIP' (cause 20 - Unknown)
>
>  == Everyone is busy/congested at this time (1:0/0/1)
>
>    -- Executing [000000 at default:2] Hangup("SIP/000001-00000078", "") in new
> stack
>
>  == Spawn extension (default, 000000, 2) exited non-zero on
> 'SIP/000001-00000078'
>
>
>
> --
> --
> Best regards
> Matiss Jekabsons
> Procerto Ltd.
>
>
>
>
> --
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