[Asterisk-Users] SIP Retransmission

Mark Spencer markster at digium.com
Sat Mar 29 09:09:12 MST 2003


try turning off re-invite.

Mark

On Sat, 29 Mar 2003, Luke Howard wrote:

>
> 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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