[asterisk-dev] BUG? Asterisk V10 SIP Message To: non numeric IP (mobile1.xyz.com) fails
mjordan at digium.com
Thu Feb 13 08:01:58 CST 2014
On Thu, Feb 13, 2014 at 3:28 AM, Johan Sandgren <jsa at svep.se> wrote:
> [Feb 12 15:13:59] VERBOSE chan_sip.c:
> <--- SIP read from UDP:22.214.171.124:5060 --->
> MESSAGE sip:mobil1.xyz.com SIP/2.0
> Via: SIP/2.0/UDP 126.96.36.199:5060;branch=z9hG4bK-3f138a53
> To: <sip:mobil1.xyz.com>
> From: <sip:188.8.131.52>;tag=7a82b127
> Call-ID: 857d4ed8 at 184.108.40.206
> CSeq: 245 MESSAGE
> Max-Forwards: 70
> User-Agent: CareIP 7813409 v220.127.116.11
> Content-Type: application/scaip+xml
> Content-Length: 138
> [Feb 12 15:13:59] VERBOSE chan_sip.c: No matching peer for
> '18.104.22.168' from '22.214.171.124:5060'
> [Feb 12 15:13:59] VERBOSE chan_sip.c: Looking for s in sipmessage
> (domain mobil1.xyz.com)
> [Feb 12 15:13:59] WARNING 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
(1) It looks for a peer that matches what sent the MESSAGE request, in
this case, sip:126.96.36.199. 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'.
(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
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
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.
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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