Danny: I added
codecprobe=yes did a restart and got the same error. I changed it to "codecprobe=no" and got the same error. Is it possible that the error is similar to this: http://www.nabble.com/ekiga-registration-in-asterisk-td15816700.html The reason I ask this is that my sip debug logs show: From: "asterisk" <sip:[email protected]>;tag=as183adb5e To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.d07f Asterisk is putting itself out as "10.1.1.40" which obviously is my internal LAN IP and not something that can be reached by ekiga.net. I've used both stunaddr=stun.ekiga.net and externip=71.xxx.xxx.xxx (my router's IP address). I double checked my router. I'd not updated my IP address for the UDP port forwarding since installing my new server (that is running asterisk). But, even after making sure 10000-20000 is being forwarded to my box, I don't see a single packet being logged by my firewall (either accepted or blocked--I fiddled with logging to try both) from ekiga.net's ip address. Also, junctn.net's sip connection is working no problem. I'm stumped. :( Any ideas out there? Thanks for the help so far...Still plugging away... Bud On Fri, 2008-12-19 at 17:00 -0600, Danny Nicholas wrote: > Check you codecprobe settings > http://lists.digium.com/pipermail/asterisk-dev/2006-October/024101.html > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of obitori junk > Sent: Friday, December 19, 2008 4:55 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Problem configuring Asterisk as client for > ekiga.net -- NAT problem > > Danny: > > I restarted asterisk both times after making changes to the sip.conf in > response to Eric's suggestions. > > I'll send you my original post with sip debug info. I ran sip debug > again. The logged info is identical to what I was getting before I > changed the codec allow/disallow settings... > > I think this is an Asterisk behind a NAT configuration problem, but if I > could be wrong. > > Bud > > > On Fri, 2008-12-19 at 16:38 -0600, Danny Nicholas wrote: > > Have you done a sip set debug, then sip reload? Do you have a range of > > 10000-20000 open in your firewall? Asterisk will "poke out" through 5060 > > but has to get a random response back in the 10-20K range (you can narrow > > this) > > > > -----Original Message----- > > From: [email protected] > > [mailto:[email protected]] On Behalf Of obitori junk > > Sent: Friday, December 19, 2008 4:28 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Problem configuring Asterisk as client > > for86.83.214.233 > > ekiga.net -- NAT problem > > > > I tried both "disallow all /// allow ulaw" and "allow all". Neither > > worked. Ekiga.net is not accepting asterisk as a client. I keep > > getting the "600 not acceptable error". What seems odd to me is that > > the NAT column shows an "n" for both ekiga.net and jnctn.net, but I have > > "nat=yes" for both. Thanks for the suggestion...I'm still open to > > ideas. > > > > Regards, > > > > Bud Roth > > > > sip.conf contains: "allow all" > > OUTPUT--> > > > > Name/username Host Dyn Nat ACL Port > > Status > > ekiga/budzhaus 86.64.162.35 N 5060 OK (102 > > ms) > > 110 (Unspecified) D 0 > > UNKNOWN > > 109 (Unspecified) D 0 > > UNKNOWN > > 108 (Unspecified) D 0 > > UNKNOWN > > 107 (Unspecified) D 0 > > UNKNOWN > > 106 (Unspecified) D 0 > > UNKNOWN > > 105 (Unspecified) D 0 > > UNKNOWN > > 104 (Unspecified) D 0 > > UNKNOWN > > 103 (Unspecified) D 0From: "asterisk" > > <sip:[email protected]>;tag=as183adb5e To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.d07f > > UNKNOWN > > 102/102 10.1.1.20 D 5061 OK (1 > > ms) > > 101 (Unspecified) D 0 > > UNKNOWN > > jnctn/obitori 66.227.100.20 N 5060 OK (22 > > ms) > > 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0 > > offline] > > [Dec 19 17:20:31] NOTICE[2478]: chan_sip.c:7517 sip_reg_timeout: -- > > Registration for '[email protected]' timed out, trying again (Attempt > > #2) > > -- Got SIP response 606 "Not Acceptable" back from 86.64.162.35 > > > > sip.conf contains: > > > > disallow all > > allow ulaw > > > > OUTPUT--> > > victoria*CLI> sip show peers > > Name/username Host Dyn Nat ACL Port > > Status > > ekiga/budzhaus 86.64.162.35 N 5060 OK (97 > > ms) > > 110 (Unspecified) D 0 > > UNKNOWN > > 109 (Unspecified) D 0 > > UNKNOWN > > 108 (Unspecified) D 0 > > UNKNOWN > > 107 (Unspecified) D 0 > > UNKNOWN > > 106 (Unspecified) D 0 > > UNKNOWN > > 105 (Unspecified) D 0 > > UNKNOWN > > 104 (Unspecified) D 0 > > UNKNOWN > > 103 (Unspecified) D 0 > > UNKNOWN > > 102/102 10.1.1.20 D 5061 OK (1 > > ms) > > 101 (Unspecified) D 0 > > UNKNOWN > > jnctn/obitori 66.227.100.20 N 5060 OK (23 > > ms) > > 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0 > > offline] > > victoria*CLI> > > > > > > On Fri, 2008-12-19 at 15:18 -0600, Eric "ManxPower" Wieling wrote: > > > obitori junk wrote: > > > > I am experiencing a "606 not Acceptable" error trying to set up an > > > > Asterisk server as an ekiga.net client. My server is behind a > firewall > > > > with NAT routing. I have googled this problem and read about Asterisk > > > > feeding its local ip address to ekiga.net. That seems to be my > > > > problem. > > > > > > In my experience "Not Acceptable" errors happen because the two > > > endpoints cannot agree on a codec. Try allowing all the codecs in your > > > softphone and in Asterisk sip.conf [general] do a disallow=all and an > > > allow=ulaw. I suggest you do this in [general] when testing because > > > it can sometimes be hard to make sure that a peer/friend/user entry is > > > actually matching the incoming call. Once you get it working you can > > > refine it. > > > > > > You could be having a NAT issue too, but I don't think it is related to > > > your 606 error. > > > > > > _______________________________________________ > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
