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

[Jerris, Michael MI]

> From: asterisk-dev-bounces at lists.digium.com 
> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Luca Spada
> 
> 
> 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;
>         }
> 

Can you please post this issue to the bugtracker at bugs.digium.com.