[Asterisk-Dev] work-around for stuck SIP channels

[Luca Spada]

Hi there,

I was suffering the following problem with stuck SIP channels (it happends with asterisk 1.0.7 and the lastest CVS-HEAD-05/24/05-20:44:59):

sip show channels
82.48.126.223    (None)      2eb141f241b  00101/00000   unknow
82.48.126.223    (None)      1f16e9e8119  00101/00000   unknow
82.48.126.223    (None)      2eb141f241b  00101/00000   unknow
82.48.126.223    (None)      1f16e9e8119  00101/00000   unknow
82.48.126.223    (None)      2eb141f241b  00101/00000   unknow
82.48.126.223    (None)      1f16e9e8119  00101/00000   unknow
82.48.126.223    (None)      2eb141f241b  00101/00000   unknow
82.48.126.223    (None)      1f16e9e8119  00101/00000   unknow
82.48.126.223    (None)      2eb141f241b  00101/00000   unknow
82.48.126.223    (None)      1f16e9e8119  00101/00000   unknow
82.48.126.223    (None)      2eb141f241b  00101/00000   unknow
82.48.126.223    (None)      1f16e9e8119  00101/00000   unknow
... lots of them

and after some hours asterisk crashed for "too many open files".

I did some debugging and discovered that the offendind IP was sending malformed "Call-ID:" header:

[...]
Call-ID: 2eb141f241b71efb79e2a9e37545e146 at 82.48.126.223eq: 857 REGISTER
[...]

It seems like it joined "Call-ID:" and "CSeq:" in one string, eating 3 chars. The funny thing is that this malformed Call-ID are coming from another asterisk. I'm making some investigations to discover what version of asterisk the user is using.

Asterisk doesn't seem to process righ this kind of "Call-ID" strings causing the above problem.

I made the following patch in order to skip this kind of malformed Call-ID.

chan_sip.c:

search for "sip_pvt *find_call", before:

        if (ast_strlen_zero(callid)) {
                ast_log(LOG_WARNING, "Call missing call ID from '%s'\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr));
                return NULL;
        }

add:

        if (strchr(callid,' ')) {
                ast_log(LOG_WARNING, "Malformed call ID '%s' from '%s'\n", callid, ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr
));
                return NULL;
        }

Mark:

I think you should better check "callid" in "create_addr" before tokening it into sip_peer.fromdomain.

Thank you for asterisk ;)

bye


Luca "skyluke" Spada - luca.spada at ngi.it - Fondatore e Amministratore Delegato
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