Christopher J. Wolff
2003-Sep-29 13:50 UTC
[Asterisk-Users] RE: SIP i.e. Is something broken?
Is it safe to assume that a fresh CVS build will not have the SIP translation problem described? Regards, Christopher --__--__-- Message: 11 Date: Mon, 29 Sep 2003 12:45:40 -0700 To: asterisk-users@lists.digium.com From: "Ernest W. Lessenger" <ernest@oacys.com> Subject: Re: [Asterisk-Users] Is somthing broken? Reply-To: asterisk-users@lists.digium.com At 12:33 PM 9/29/2003, you wrote:>Can you clarify any / find me on IRC? (irc.freenode.net/#asterisk/kram)Just FYI: I had similar problems for a while, and then I completely scrapped my CVS directory and did a CVS CHECKOUT (instead of an update). That solved the problem. --Ernest
Christopher J. Wolff wrote:> Is it safe to assume that a fresh CVS build will not have the SIP > translation problem described? > > Regards, > Christopher > --__--__--> > Just FYI: I had similar problems for a while, and then I completely > scrapped my CVS directory and did a CVS CHECKOUT (instead of an update). > That solved the problem. >I "checkout" rebuilt from CVS last night about 10:00 pm, and filed a bug report at that time. It isn't *all* SIP calls for me, btw; my ATA186 works just fine, but my Budgetone won't work with the "broken" code. . . It's as described in other mails--I can receive calls on the Budgetone but when I make them the RTP part is broken and the calls cut off the second they're set up. B. -- This message has been scanned for viruses and is believed to be clean. Scan engine v4.2.40 for Linux. Virus data file v4294 created Sep 18 2003 Scanning for 80178 viruses, trojans and variants.
Christopher J. Wolff wrote:>Is it safe to assume that a fresh CVS build will not have the SIP >translation problem described? > >Regards, >Christopher >--__--__-- > > >If you do try the current CVS and UA's behind NAT work please let me know.. later..
Thanks. That worked -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Thorsten Lockert Sent: Tuesday, 30 September 2003 11:43 AM To: asterisk-users@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@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users