asterisk users - Jul 2011 - REALY strange issue with making calls biside 2 phones

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.

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.
>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> ? ? ? ? ? ? ?http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> ?http://lists.digium.com/mailman/listinfo/asterisk-users
>