[Asterisk-Users] SIP Retransmission

[Matteo Brancaleoni]

I can confirm that.
After 10/15 seconds, outgoing sip calls are
destroyed. 
Incoming sip calls works ok.
That "destroy" timeout restart if any action
is taken during the call, ie a dtmf
press.

Verified and repudicible with snom100, messenger, cisco phone.
Matteo

Il sab, 2003-03-29 alle 01:24, Luke Howard ha scritto:
> Latest CVS breaks outgoing SIP calls for me after a second or so
> of audio (if that).
> 
>   -- Executing Macro("SIP/515-Office-b922", "iconnecthere|18006822878|60") in new stack
>   -- Executing Dial("SIP/515-Office-b922", "SIP/18006822878 at iconnecthere|60|r") in new stack
>   -- Called 18006822878 at iconnecthere
>   -- SIP/iconnecthere-31cc answered SIP/515-Office-b922
>   -- Attempting native bridge of SIP/515-Office-b922 and SIP/iconnecthere-31cc
>   -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 203.13.32.85
> == Spawn extension (macro-iconnecthere, s, 1) exited non-zero on 'SIP/515-Office-b922' in macro 'iconnecthere'
> == Spawn extension (local, s, 1) exited non-zero on 'SIP/515-Office-b922'
> 
> -- Luke
> 
> -
> 
> --
> Luke Howard | PADL Software Pty Ltd | www.padl.com
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-- 
Matteo Brancaleoni <mbrancaleoni at espia.it>
Espia - Emmegi Srl