[asterisk-dev] [svn-commits] mmichelson: branch 1.8 r368625 - /branches/1.8/channels/chan_sip.c

Mark Michelson mmichelson at digium.com
Thu Jun 7 09:08:56 CDT 2012


On 06/07/2012 03:51 AM, Walter Doekes wrote:
>>> To fix the first problem, we zero out the to-tag seen in the incoming
>>> INVITE. This way, Asterisk, when time to send a response, will send
>>> its generated local tag instead.
>> Which means that the sender of the SIP request can't match your 
>> response to the request. The to-tag is an important part
>> of the dialog-matching and you can't change it like that.
>>
>> Did you test this with a number of devices?
>
> Very valid concern indeed. I was just writing the same thing on the 
> bug report.
>
> This changeset does fix the problem with the sipp scenario from r1918, 
> but that's not a valid test at all.
>
> Regards,
> Walter
Yep, I have re-opened the issue and will pursue a different approach. I 
had a moment before committing the fix where I thought I might put it on 
review board and then thought, "nah this is simple enough." I'll put the 
next potential fix on review board first.

Mark!



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