[Asterisk-Dev] stable chan_sip borked
Andrew Lindh
asterisk at ntplx.net
Mon Dec 20 02:34:26 MST 2004
That seems to help the current CVS...thanks!
>From: <dbruce at bananatel.com>
>To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
>Subject: Re: [Asterisk-Dev] stable chan_sip borked
>Date: Mon, 20 Dec 2004 02:19:02 -0700
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>
>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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