[asterisk-users] Outgoing SIP calls dropped after 30 seconds.

[Steve Totaro]

Usually, calls terminating at 30 seconds is a sure sign that you need to add
an Answer() in your dialplan.  Try dropping that in before you dial out.  I
have seen this so many times and Answer() has always fixed the issue.  The
magic number is 30 seconds.

Depending on if you use your CDRs for anything, especially billing, you may
need to figure a way around that, since even if a call rings out, the CDR
will reflect Answered.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)



On Fri, Nov 7, 2008 at 10:38 PM, Grey Man <greymanvoip at gmail.com> wrote:

> To get to the bottom of it I'd recommend determining why the ACKs are
> not getting through to Asterisk rather than trying to work around it.
> I'm actually suprised Asterisk terminates the call by default when it
> doesn't get the ACK to it's 200 Ok response that must be new for
> 1.4.22 as I haven't seen that behaviour in earlier versions. In my
> opinion it's unwarranted behaviour, if Asterisk is getting RTP then it
> should leave the call up irrespective of whether it gets an ACK or
> not.
>
> From the original SIP trace the ACK does not appear to be arriving at
> your Asterisk server at all. Try doing a packet trace on the network
> segment where the calling SIP agent is and see where it's trying to
> send the ACK to. My guess would be your firewall is incorrectly
> handling the SIP messages. Generally it's very bad news to use an ALG
> or firewall to mangle SIP packets as they almost always get it wrong.
>
> In your case there is a Record-Route header in the response so the ACK
> request should be being sent to that address. Perhaps your firewall is
> not correctly mangling that to allow the request to find its way back
> to your Asterisk server.
>
> Record-Route: <sip:216.82.224.202;lr;ftag=as3ed791f3>
>
> Regards,
>
> Greyman.
>
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