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

[Master Abi]

Thanks. That worked

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Thorsten
Lockert
Sent: Tuesday, 30 September 2003 11:43 AM
To: asterisk-users at lists.digium.com
Subject: RE: [Asterisk-Users] RE: SIP i.e. Is something broken?


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

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users