[Asterisk-Users] RE: SIP i.e. Is something broken?

[Thorsten Lockert]

To roll back only the affected stuff for SIP negotiation, I would recommend:

	make update
	cvs update -j 1.181 -j 1.179 channels/chan_sip.c

Note that the second line should only be executed *once*.  Once this is
fixed
in CVS, you should *remove* channels/chan_sip.c to make sure the changes
done
by the second line above are removed again.

By using this recipe you will be able to run the latest and greatest version
of Asterisk, and still have working codec negotiation for SIP.  Note that
this
negotiation problem only happens with certain devices, including
Grandstream.

Thorsten