Deactivate the old one. Activate the new one. IF the phone is registered send it its profile and tell it to reboot (or reboot the phone manually).
The phone should auto detect the change and load the firmware accordingly. There's no sipx magic here, just phone bootrom logic as long as its activated. ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: [email protected] <[email protected]> To: [email protected] <[email protected]> Sent: Thu Jul 15 16:36:58 2010 Subject: Re: [sipx-users] Polycom IP 430 won't register remotely Ok. We are making progress. It APPEARS as though the phone is now communicating with the sipX via FTP. I deleted and recreated one of the remote phones in sipX, changing the line configuration from what it used to be (so I could actually SEE a change from the old configuration). The configuration was indeed updated. The PROBLEM is that it isn't downgrading the firmware. I uploaded the 3.1.3c file in Devices > Device Files. I setup a new "profile" basically using the 3.1.3c file and 4.2.2 bootrom and just deactivated the 3.2.3 "profile". Ideas? On Thu, Jul 15, 2010 at 3:56 PM, JOLY, ROBERT (ROBERT) <[email protected]>wrote: > > Ok. So. I'm trying to remotely re-provision the phone and > > downgrade the firmware. I figured i'd try TFTP first since I > > know that works in the local office. So I forwarded port 69 > > on the local router to the internal IP of my sipX. Went into > > the network setup menu on the phone and defined > > "myhostname.no-ip.org" as the boot server, using TFTP. I > > rebooted and I'm getting an error that the phone can't > > connect to the boot server. Any ideas? > > Yes, TFTP does not go across NATs and firewalls unless you have a TFTP ALG > on your router (and few do). This is due to the fact that TFTP uses an > ephemeral port for sending down the file to the client. You will have > much > better luck using FTP. > > > > > > > On Thu, Jul 15, 2010 at 6:39 AM, Tony Graziano > > <[email protected]> wrote: > > > > > > I may have spoke too soon... > > > > > > I see you can load dd-wrt on the actiontek... > > > > > > > > http://www.dd-wrt.com/wiki/index.php/Supported_Devices#Actiont > > ec <http://www.dd-wrt.com/wiki/index.php/Supported_Devices#Actiontec> > > > > > > > > <http://www.dd-wrt.com/wiki/index.php/Supported_Devices#Action > > tec> Question is what revision and is it in the supported > > devices list or on the HCL list, only rev a,b,c,d are > > supported it seems. > > > > > > On Thu, Jul 15, 2010 at 5:58 AM, Tony Graziano > > <[email protected]> wrote: > > > > > > You might find your internal firewall may have > > an issue as well passing audio if it cannot do symmetrical > > NAT (which I doubt). > > > > So once you get DNS setup where the phones will > > register you should try making a call and if you get no (or > > one-way audio), you should stop and address the firewall piece. > > > > If it were me (and it is not), I would see if I > > could set the verizon modem to "bridged" mode and put a > > compatible firewall in. I say this because in looking through > > the fios devices of that model, I see nothing which > > encourages me that this is something that has the guts to > > work in the way that is needed, but is fine for a remote site. > > > > On Thu, Jul 15, 2010 at 3:55 AM, Paul Scheepens > > <[email protected]> wrote: > > > > > > Gary Luca <[email protected]> wrote > > on 15-07-2010 05:02:40: > > > > > > > Ok. So I just got a whole TON of > > replies with questions to answer. Here goes: > > > > > > Nathaniel... > > > > > > DNS/DHCP at the remote site are > > handled by the Westell ADSL2 modem. > > > It deals out just IP, Subnet mask, > > gateway (itself) and DNS (also > > > itself). DNS relays to what I > > imagine are Verizon DNS servers. The > > > relavent DNS records on the local > > network (in Microsoft DNS Server) > > > are as follows: > > > > > > spadafora4senate.com > > > mal-pbx (A) 172.16.17.45 > > > spadafora4senate.com (NAPTR) > > [2][0][S][SIP+D2U]<Reg Exp>[_sip._ > > > udp.spadafora4senate.com.] > > > spadafora4senate.com (NAPTR) > > [2][0][S][SIP+D2T]<Reg Exp>[_sip._ > > > tcp.spadafora4senate.com.] > > > > > > As far as I know you can skip the > > NAPTR's, I have been running without them for years. > > > > > > > > > > _tcp.spadafora4senate.com > > > _sip (SRV) [200][1][5060] > > mal-pbx.spadafora4senate.com. > > > _sips (SRV) [300][1][5060] > > mal-pbx.spadafora4senate.com. > > > > > > _udp.spadafora4senate.com > > > _sip (SRV) [100][1][5060] > > mal-pbx.spadafora4senate.com. > > > > > > > > > These are the DNS records you use on > > the central site, that's why your Polycoms work locally. > > You need something similar on the > > remote site, that's what's missing. > > > > > > > Doug... > > > > > > Thank you for the link regarding FTP. > > Do you know of any place > > > where I can find detailed > > instructions on how precisely to configure > > > sipX for whatever it needs in order > > for FTP to work for phone > > > provisioning? On the local network I > > use TFTP, which I don't mind > > > because it is a secure network. For > > the REMOTES i'd rather use FTP > > > as it incorporates authentication. > > > > > > Tony... > > > > > > I actually thought it WAS a DNS issue > > for a while. I looked through > > > the wiki and everything I could find > > about DNS considerations for > > > sipX and nothing mentioned anything > > about external DNS (that I could > > > see). So, being fairly new to the > > SIP world, I assumed the only > > > resolution that had to be done by the > > remote phone was in regards to > > > the Outgoing Proxy. I figured it > > just contacted the Outgoing Proxy > > > and that acted as the middle man for > > everything, not requiring the > > > phone to have to do any other DNS > > resolution or anything. The > > > proxy, being on the local network, > > would resolve everything to the > > > local DNS and just make everything > > work for the remote user. It > > > seemed to make sense, but my limited > > knowledge of the inner workings > > > of the relationship between the > > remote, the proxy, and the registrar > > > left me without a DEFINITE answer. > > > > > > So at that point, with the phones > > just NOT registering at all, it > > > was clear that it could be ANYTHING. > > My sipX config could be wrong. > > > I could have screwed something up > > with the NAT traversal features. > > > It could have been a DNS issue. It > > could have been a phone issue. > > > So I had to narrow it down. That's > > when I installed x-lite. And > > > within like a MINUTE, I had it > > connected to sipX and able to make > > > and receive calls. There was NO VPN > > involved or anything. Just > > > straight over the internet through a > > NAT on both ends. Here are the > > > settings I used: > > > > > > General > > > User ID: x703 > > > Domain: spadafora4senate.com > > > Password: ********* > > > Display name: Gary Luca > > > Authorization name: [blank] > > > > > > Register with domain and receive > > calls: [checked] > > > Send outbound via: Proxy - > > Address: myhostname.no-ip.org > > > > > > Get x-lite working without "Send > > outbound via Proxy". > > By enabling "Send outbound via Proxy" > > x-lite will register via the A record of myhostname.no-ip.org. > > If "Send outbound via Proxy" is > > disabled x-lite will try to resolve the SRV records for > > spadafora4senate.com (or was that spada4a4senate.com ;-) > > This is also the method the Polycom's will use. > > > > Now you need to create the > > corresponding SRV records somewhere so that x-lite and > > polycoms on the remote site can register via SRV records: > > > > _tcp.spadafora4senate.com > > _sip (SRV) [200][1][5060] > > myhostname.no-ip.org > > _sips (SRV) [300][1][5060] > > myhostname.no-ip.org > > > > _udp.spadafora4senate.com > > _sip (SRV) [100][1][5060] > > myhostname.no-ip.org > > > > This can be done by defining the SRV > > records on your internet dns-server (ISP) > > or setting up a DNS at the remote site > > or ... I don't know your setup enough to give the best advise. > > > > > > > Dial plan: #1\a\a.T;match=1;prestrip=2; > > > > > > Topology > > > IP Address: Use local IP address > > > STUN Server: Discover server > > > Enable ICE: [unchecked] > > > > > > Manually specify range: [unchecked] > > > > > > I think this is the port range (don't > > have x-lite running), normally the port range > > is limited to keep the firewall rules > > simple and not too open. > > It is normally limited to ports > > 30000-31000, a smaller range is also possible. > > > > > > > > > > Presence > > > Mode: Peer-to-peer > > > Poll time: 300 > > > Refresh interval: 3600 > > > > > > Transport > > > Signaling transport: UDP > > > > > > Advanced > > > Reregister every: 3600 seconds > > > Minimum time: 20 seconds > > > Maximum time: 1800 seconds > > > > > > Enable session timers: [unchecked] > > > > > > Send SIP keep-alives: [checked] > > > Use rport: [checked] > > > Send outgoing request directly > > to target: [checked] > > > > > > So if what I assuming about the proxy > > being the middle man is wrong > > > and what you are saying about the > > remote needing to resolve the SRVs > > > itself is correct, then I have > > absolutely NO idea why x-lite is > > > working. But it is. If you want to > > email me directly (outside of > > > the list), I'll even set you up with > > a test user on my system so you > > > can configure x-lite or any other > > manually configured phone (soft or > > > hard) and evaluate the results. > > > > > > Dale... > > > > > > The phone doesn't list in the > > Registrations page. On the phone > > > interface, the little "phone" icon > > next to each of the two line > > > buttons is "hollow" indicating that > > the line did not register (not > > > sure how familiar you are with the > > Polycom display). Calls to the > > > phone fail to make it ring. > > > > > > Thank you all. Let me know if I can > > clarify anything further. I > > > look forward to your thoughts > > > > > > -G > > > > > > > > > > > > -- > > > Gary J. Luca Jr. > > > > > > 781-333-8087 > > > http://www.linkedin.com/in/garylukes > > <http://www.linkedin.com/in/garylukes> > > > > > > > _______________________________________________ > > > sipx-users mailing list > > [email protected] > > > List Archive: > > http://list.sipfoundry.org/archive/sipx-users > > <http://list.sipfoundry.org/archive/sipx-users> > > > Unsubscribe: > > http://list.sipfoundry.org/mailman/listinfo/sipx-users > > <http://list.sipfoundry.org/mailman/listinfo/sipx-users> > > > sipXecs IP PBX -- > > http://www.sipfoundry.org/ <http://www.sipfoundry.org/> > > > > _______________________________________________ > > sipx-users mailing list > > [email protected] > > List Archive: > > http://list.sipfoundry.org/archive/sipx-users > > Unsubscribe: > > http://list.sipfoundry.org/mailman/listinfo/sipx-users > > sipXecs IP PBX -- http://www.sipfoundry.org/ > > > > > > > > > > > > -- > > > > ====================== > > Tony Graziano, Manager > > Telephone: 434.984.8430 > > > > sip: [email protected] > > > > Fax: 434.984.8431 > > > > Email: [email protected] > > > > LAN/Telephony/Security and Control Systems Helpdesk: > > Telephone: 434.984.8426 > > > > sip: [email protected] > > > > Fax: 434.984.8427 > > > > Helpdesk Contract Customers: > > http://www.myitdepartment.net/gethelp/ > > > > > > Why do mathematicians always confuse Halloween > > and Christmas? > > Because 31 Oct = 25 Dec. > > > > > > > > > > > > > > -- > > ====================== > > Tony Graziano, Manager > > Telephone: 434.984.8430 > > sip: [email protected] > > Fax: 434.984.8431 > > > > Email: [email protected] > > > > LAN/Telephony/Security and Control Systems Helpdesk: > > Telephone: 434.984.8426 > > sip: [email protected] > > Fax: 434.984.8427 > > > > Helpdesk Contract Customers: > > http://www.myitdepartment.net/gethelp/ > > > > Why do mathematicians always confuse Halloween and Christmas? > > Because 31 Oct = 25 Dec. > > > > > > > > > > > > > > -- > > Gary J. Luca Jr. > > > > 781-333-8087 > > http://www.linkedin.com/in/garylukes > > > > > -- Gary J. Luca Jr. 781-333-8087 http://www.linkedin.com/in/garylukes _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
