[asterisk-dev] BUG? Asterisk V10 SIP Message To: non numeric IP (mobile1.xyz.com) fails

[Matthew Jordan]

On Thu, Feb 13, 2014 at 3:28 AM, Johan Sandgren <jsa at svep.se> wrote:
> Hi,
>

<snip>

>
>
> [Feb 12 15:13:59] VERBOSE[25824] chan_sip.c:
>
> <--- SIP read from UDP:83.186.238.111:5060 --->
>
> MESSAGE sip:mobil1.xyz.com SIP/2.0
>
> Via: SIP/2.0/UDP 83.186.238.111:5060;branch=z9hG4bK-3f138a53
>
> To: <sip:mobil1.xyz.com>
>
> From: <sip:83.186.238.111>;tag=7a82b127
>
> Call-ID: 857d4ed8 at 83.186.238.111
>
> CSeq: 245 MESSAGE
>
> Max-Forwards: 70
>
> User-Agent: CareIP 7813409 v1.2.4.0
>
> Content-Type: application/scaip+xml
>
> Content-Length: 138
>
>
>

<snip>

>
> [Feb 12 15:13:59] VERBOSE[25824] chan_sip.c: No matching peer for
> '83.186.238.111' from '83.186.238.111:5060'
>
> [Feb 12 15:13:59] VERBOSE[25824] chan_sip.c: Looking for s in sipmessage
> (domain mobil1.xyz.com)
>
> [Feb 12 15:13:59] WARNING[25812] pbx.c: Channel 'Message/ast_msg_queue' sent
> into invalid extension 'mobil1.xyz.com' in context 'sipmessage', but no
> invalid handler
>

It isn't a bug. It is telling you how it will attempt to match the
inbound request:

(1) It looks for a peer that matches what sent the MESSAGE request, in
this case, sip:83.186.238.111. That fails.
(2) Since the request URI is simply a domain and not a destination, it
falls back to looking for an 's' extension in context 'sipmessage'.
That fails.
(3) Now, truly panicking, it looks for the 'i' extension in the same
context. Since you don't have an invalid extension handler, that fails
too.

Despondent, it throws in the towel.

I'm not sure where you thought it would end up, but it certainly tried
lots of different places. And the 'rules' for it doing so are (for
chan_sip, at any rate) relatively consistent with how inbound requests
are matched for INVITE requests as well. chan_sip tries to figure out
who sent the request to Asterisk, and then use that peer definition.
If chan_sip can't find that, it falls back to using a general entry
point.

Also: please don't stay on Asterisk 10. That version is no longer
supported and is no longer receiving security fixes. You should move
to Asterisk 11, which is an LTS release, as soon as possible.

Matt

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org