[asterisk-users] Dial weirdness

Mark Michelson mmichelson at digium.com
Mon Jan 26 15:43:49 CST 2009


Bruce Ferrell wrote:
> I'm seeing this response to SIP calls originated in the following manner:
> 
> Dial(SIP/${EXTEN}&SIP/{$DID},30,r)
> 
>     handle_response_invite: Re-invite to non-existing call leg on other UA.
> 
> The response is from the second part of the dial.  What exactly does it 
> mean and how can I fix it?
> 
> Thanks in advance
> 
> Bruce

First of all, it may just be a transcription error on your part, but the 
variable in the second part of the Dial statement should be ${DID} instead of 
{$DID}.

That message you see on the console probably means that Asterisk has received a 
481 response to the INVITE it has sent out. Apparently, whoever is receiving the 
INVITE thinks that it is a re-INVITE that belongs to an established SIP dialog, 
but then it can't actually find the dialog to which the INVITE belongs. This 
seems like it is an incorrect interpretation by the remote end since Asterisk 
generates a new callid, new from-tag, and has no to-tag on each initial INVITE 
it sends out when starting a call. It may be helpful to look at a packet capture 
from a failed attempt. It may be that whoever is sending back the 481 is sending 
a reason for it, or it may be that there is something obviously malformed in the 
SIP requests being sent by Asterisk.

Mark Michelson



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