Yves, I have a SoundStation IP 6000:
My sip.conf [1006] type=friend context=sip-phone call-limit=1 trustrpid=no callerid="Conference Room3" <1006> disallow=all allow=ulaw allow=alaw username=1006 secret=secret1 dtmfmode=rfc2833 host=dynamic mailbox=1000 nat=yes canreinvite=no Asterisk server IP XX.XX.42.16 (Public) Client IP: 192.168.1.56 I would look for typos in your configuration! -Motty From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, December 22, 2016 6:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom SoundStation IP 6000 does not register Do you have any LLDP or CDP enabled anywhere ? 2016-12-21 19:50 GMT+01:00 Victor Villarreal <mefhigos...@gmail.com>: Hi Yves, Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC of the phone. Maybe with the snom this not happen because your switch don't see the MAC of the Snom as a "supperted IP Phone". 2016-12-21 13:59 GMT-03:00 Yves <yves...@gmx.de>: sorry... typo.... the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am 21.12.2016 um 15:13 schrieb Mauricio Tavares: On Wed, Dec 21, 2016 at 7:50 AM, Yves <yves...@gmx.de> wrote: Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves I am a bit confused: is your problematic phone's IP 192.168.0.13 (what the error log is reporting below) or 192.168.1.13? Am 21.12.2016 um 13:34 schrieb Mark Wiater: Yves, Didn't you say that AsteriskServer: 192.168.1.211 SIP-user: 165 ? On 12/21/2016 4:24 AM, Yves wrote: . It is sure for 100% that there is no firewall or something else mangeling in between... another Hardphone works as expected using the same Netzworkcable on the same Networkplug with UDP on Port 5060... This other hardphone, what IP does it have? 000050.848|cfg |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask 255.255.255.0 The line above suggests to me that your phone and your asterisk server are on a different network, there has to be something that routes between those two networks. Often what routes, can firewall. 000122.941|sip |4|03|Registration failed User: 165, Error Code:480 Temporarily not available Mark -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- GnuPG Key ID: 0x39BCA9D8 https://www.github.com/mefhigoseth ...:::[ God Rulz ! ]:::... -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
