[Asterisk-Dev] stable chan_sip borked

[dbruce at bananatel.com]

It appears that the problem with the patch that OEJ submitted is in the new 
parse_ok_contact function. Basically, the function is parsing the "sip:" 
qualifier out of the contact before storing it in pvt->okcontacturi. This 
causes the ACK that is generated to be non-compliant. To fix it, within 
parse_ok_contact, change:

***************
 /* Make sure it's a SIP URL */
 if (strncasecmp(c, "sip:", 4)) {
  ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying 
to use anyway\n", c);
 } else
  c += 4;

 strncpy(pvt->okcontacturi, c, sizeof(pvt->okcontacturi) - 1);

***************

to:

***************
 strncpy(pvt->okcontacturi, c, sizeof(pvt->okcontacturi) - 1);

 /* Make sure it's a SIP URL */
 if (strncasecmp(c, "sip:", 4)) {
  ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying 
to use anyway\n", c);
 } else
  c += 4;
***************

(sorry... I can't provide a unified diff... I back-ported the fix into an 
older copy of CVS...)

----- Original Message ----- 
From: "Matt Hess" <mhess at livewirenet.com>
To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
Sent: Monday, December 20, 2004 12:39 AM
Subject: [Asterisk-Dev] stable chan_sip borked


> The latest stable version of chan_sip (1.510.2.27) breaks almost all of
> our stuff.. cisco ata devices no longer work for audio.. most of our
> voip terminators also spit back a lot of errors.. reverted back to cvs
> -D2004-12-10 and things were happy again.
>
> just fyi.
>
>


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