VOIP Phone Continously "Registering"

Please help.
We use Callmanager 3.2 with a publisher and two subscribers. 90% of phones are 7940's with some 7960's too.
More often these days, when people try to login (using extension mobility) the phone will prompt "REGISTERING" this message is continuously shown.
Replacing the phone will alleviate the problem, but we are building up a selection of problematic phones.
Is there a resolution or something we should be doing on our Callmanager cluster as a method of maintenance etc.
Kindest Regards..
Kai Nicholls

Is it getting the IP address? Is it able to contact the CallManager? Was there any recent CCM upgrade, if so, the firmware for that phone should be available on the new CCM after upgradation. If firmware is already available and inspite of it , it is cycling in the Registering states, then it could be due to the CCM configuration. Try stopping and starting the TFTP and CCM on the Servicabbility and Control Center. This should help it to work.
Cisco IP Phone Model 7960G, 7940G, and 7910 Administration Guide for Cisco CallManager Release 3.3 and later
http://www.cisco.com/en/US/products/hw/phones/ps379/products_administration_guide_book09186a00801d66b3.html

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HT5538 Registering a VoIP phone number to Apple ID

I'm using an VoIP based phone on both my iPad mini and my iMac desktop. I DO NOT have a carrier based iPhone number. So how can I associate my assigned VoIP phone number with iMessage? I can get iMessages using my email Apple ID, but this is way too confusing for other iPhone users who "text" to phone numbers and let iMessage resolve the message path.
In my opinion, Apple needs to either: (1) allow for the verfication & use of non-iPhone numbers in their Apple ID database, or (2) change iOS on the device to allow for the entry of an "Apple ID" into the contact's information. This "Apple ID" entry could then be used for both iMessage, and FaceTime.

Welcome to the Apple Community.
Settings > Messages > Send & Receive.

Branch Office VOIPs do not register.

Hi:
I've been breaking my head on this for a few weeks and nothing seems to be working.
I have three PIX 515e, one at each office.
ALL VOIPs are Polycom 300IP phones.
We have a main office (called PB) with 15 VOIP phones.
We have a branch office (called JAX) with 2 VOIP phones.
We have a branch office (called JADE) with 2 VOIP phones.
All site VOIPs must register with a hosted PBX outside of all three offices (called TN).
All 15 VOIPs at PB are registering and working with TN.
Only one of two VOIPs at JAX is registering with TN.
No VOIPs at JADE are registering with TN.
VPN Tunnels are up and functioning between PB and JAX and PB and JADE. Able to ping both ways and users in both branch sites are able to map folders to our servers.
I have opened UDP 5060 (SIP) on all interfaces. It seems there is initial conversation between TN and JAX and JADE but receiving following errors at both branches.
Pre-allocate SIP for secondary channel blah blah blah and followed immediately with a
Teardown UDP connection blah blah blah
I have attached configs for all three PIX 515e boxes (edited for security).
Could somebody take a gander at this and help me out. I'm at a complete loss.
Thank you so much in advance and have a great day!

Thank you for the feedback and suggestion GTG! I went ahead and posted it on the "security" bb and I'm going to look into SIP inspection.
Can you please MOVE this thread to the Security section and delete the duplicate post you've created?
Here's the link to your duplicate post: https://supportforums.cisco.com/thread/2260989

CME SIP issue - Cisco 7821 phone not registering

Hi
I am having issues with getting a Cisco 7821 phone to register.
Current deployment is with Cisco 6921 phones SCCP registration
SIP integration with CUE
SIP integration with Mitel system
c2951-universalk9-mz.SPA.154-3.M1.bin (CME 10.5)
In flash:
rootfs78xx.10-1-1SR1-4.sbn
kern78xx.10-1-1SR1-4.sbn
sboot78xx.10-1-1SR1-4.sbn
sip78xx.10-1-1SR1-4.loads
The 7821 phone gets IP address but fails to register. Please could somebody let me know why phone is not registering.
Configuration below (10.245.226.132 is CME address) .
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol pass-through g711ulaw
modem passthrough nse codec g711ulaw redundancy maximum-sessions 5
h323
sip
registrar server expires max 600 min 60
options-ping 90
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
voice register global
mode cme
source-address 10.245.226.132 port 5060
max-dn 30
max-pool 10
load 7821 sip78xx.10-1-1SR1-4
authenticate register
authenticate realm all
timezone 22
date-format D/M/Y
voicemail 590
tftp-path flash:
create profile sync 0061443538560005
network-locale GB
voice register dn 1
number 1010
name user1
label user1
mwi
voice register pool 1
busy-trigger-per-button 2
id mac F09E.636E.63F2
type 7821
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username 1010 password 123
codec g711ulaw
no vad
dial-peer voice 391 voip
description *** Auto Attendant ***
destination-pattern 399
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 392 voip
description *** Administration Via Telephone ***
destination-pattern 392
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 393 voip
description *** Extension Assigner ***
service ea out-bound
destination-pattern 393
session target ipv4:10.245.226.132
dial-peer voice 590 voip
description *** Voice Mail Pilot ***
destination-pattern 590
b2bua
session protocol sipv2
session target ipv4:10.245.226.131
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 1 pots
description ** Match all incoming POTS calls **
translation-profile incoming IncomingPSTNcalls
incoming called-number .
direct-inward-dial
dial-peer voice 899 voip
description Call to Mitel
translation-profile incoming Prefix9
translation-profile outgoing rem44
destination-pattern [23]..
session protocol sipv2
session target ipv4:192.168.114.2
voice-class codec 1
dtmf-relay rtp-nte
no vad
interface GigabitEthernet0/0
description *** Connection to Mitel Phone System ***
ip address 192.168.114.5 255.255.255.248
duplex auto
speed auto
interface ISM0/0
description *** Connection to Cisco Unity Express ***
ip unnumbered GigabitEthernet0/1
service-module ip address 10.245.226.131 255.255.255.128
!Application: CUE Running on ISM
service-module ip default-gateway 10.245.226.132
interface GigabitEthernet0/1
description *** Connection to IP Phone LAN ***
ip address 10.245.226.132 255.255.255.128
duplex auto
speed auto
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
ip route 0.0.0.0 0.0.0.0 10.245.226.129
ip route 10.245.226.131 255.255.2
tftp-server flash:apps37sccp.1-4-4-0.bin
tftp-server flash:sip78xx.10-1-1SR1-4.loads
tftp-server flash:rootfs78xx.10-1-1SR1-4.sbn
tftp-server flash:sboot78xx.10-1-1SR1-4.sbn
sip-ua
mwi-server ipv4:10.245.226.131 expires 3600 port 5060 transport udp
registrar ipv4:10.245.226.132 expires 600
gatekeeper
shutdown
telephony-service
authentication credential cmeadmin c4p1ta2012
xml user xmladmin password xmladmin 15
extension-assigner tag-type provision-tag
max-ephones 104
max-dn 299
ip source-address 10.245.226.132 port 2000
auto assign 101 to 105
no service directed-pickup
timeouts interdigit 5
system message CFGS
url services http://10.245.226.131/voiceview/common/login.do
url authentication http://10.245.226.132/CCMCIP/authenticate.asp
cnf-file location flash:
cnf-file perphone
load 7931 SCCP31.9-2-1S
load 6921 SCCP69xx.9-2-1-0
time-zone 22
date-format dd-mm-yy
voicemail 590
max-conferences 8 gain -6
call-forward pattern .T
moh enable-g711 "music-on-hold.au"
web admin system name cmeadmin secret 5 $1$QmIK$46fDKVSudMxzI2bRp/Ef7/
time-webedit
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 298
number 598...
mwi on
ephone-dn 299
number 599...
mwi off

Page 7 of the following link recommends that you use option 150 with the Cisco 7800 series phones and use option 66 if you cannot use option 150
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7821_7841_7861/10_1/english/admin_guide/PA2D_BK_AB3F74DA_00_admin-7821-7841-7861-10_0/PA2D_BK_AB3F74DA_00_admin-7821-7841-7861-10_0_chapter_01.pdf
Dynamic Host Configuration Protocol (DHCP)
DHCP dynamically allocates and assigns an IP address to network devices.
DHCP enables you to connect an IP phone into the network and have the phone become operational without your needing to manually assign an IP address or to configure additional network parameters.
DHCP is enabled by default. If disabled, you must manually configure the IP address, subnet mask, gateway, and a TFTP server on each phone locally.
Cisco recommends that you use DHCP custom option 150. With this method, you configure the TFTP server IP address as the option value. For additional supported DHCP configurations, go to the "Dynamic Host Configuration Protocol" chapter and the "Cisco TFTP" chapter in the Cisco Unified Communications Manager System Guide.
Note
If you cannot use option 150, you may try using DHCP option 66.

SP-3102 - Unable to make VOIP phone calls after switching router

I had to change the router from Netgear (g) to Belkin N+ (n). Changed the SPA-3102 config to DHCP, it picked up an IP and showed the 1st 3 LEDs lit, status Registered. So, good news.
I am able to receive VOIP phone calls, but not able to make phone calls, it won't dial. I installed slogsrv.exe and got the following log: syslog.514.log. Unfortunately, it's chinese to me.
Please help,
Daniel

Try checking any settings on the old router which you might need to configure also on the new router to get VoIP working like port forwarding, dmz or disabling the NAT. The Belkin might have a feature like NAT which behaves differently so you might need to open up VoIP ports or enable NAT mapping and NAT Keep Alive on the SPA.

6760s Voip phone

I would like to set the 6760s ( V 21.005 custom version jblack 50.01 )like a VOIP phone.
I went already to: " Tools-Phone settings-Connection-SIP settings " and i setted and registered a SIP provider in "Options " menu i have also setted it as "Default profile" .
The SIP provider is registered and underscored but the phone does'nt work.
There is any other setting to do or that's all ?
I think that if there is "SIP settings" menu the phone shoud work as VOIP phone like others Nokia mobile phones.
Please, anybody cna give me an help for this topic ?
Thanks.

Never set this up my self but the User guide is not at help at all?
https://www.developer.nokia.com/info/sw.nokia.com/id/4ecbad3f-764e-4095-9d1f-944bb7d21b69/SIP_VoIP_S...
As far as i know, when you have set up the SIP settings you can then under Contacts add a new contact, or edit an existing user and add a SIP number to that contact
If you set up the SIP settings correct.
In the phone go to Menu -> Contacts -> Options -> New contact
Option -> Add detail -> SIP
You should in theory then be able to dial that SIP number and it should work

PAP2T VOIP phone adapter not working

My new Linksys PAP2T VOIP phone adapter will not work. When I install the adapter (I'm sure I have done this correctly), all I get is a slow blinking green power light and a fast blinking green internet light. I get no phone light, and I get no dial tone. Is my PAP2T defective? Or is something else going on? Can you recommend a fix?

The "User Guide" which comes with your PAP2T and you can read or is downloadable from Linksys says the following about the "Power Lite"
Green. The Power LED lights up when the Phone Adapter is powered on and connected to the Internet. It flashes when the Phone Adapter is not connected to the Internet. The LED also flashes when the Phone Adapter is booting up or upgrading its firmware.
It says the following about the "Internet Lite"
Green. The INTERNET LED lights up when there is an active connection through the port. If the LED is flashing, then there is traffic moving through that port.
It says the following about the "Phone Lite"
Green. The PHONE LED is solidly lit when a telephone or fax machine has an active or registered connection to your Internet phone service provider through the corresponding port. It flashes when the phone is being used or is off the hook.
Using logic, I would say the adapter may not connected to the internet and if it is, the adapter failed to register with your voip provider.

RV220W: VoIP phones losing registration and staying offline

Hi all,
VoIP phones connected to a Cloud PBX provider are losing their registration to the Cloud PBX. The router in question is an RV220W.
This started occurring when more phones were added to the site. Phones will boot up, register and then lose their registration within a minute or two.
A larger number of phones of the same model with identical config are connecting through a Cisco 871 router without issues so we know the phones and their configs are good.
SIP ALG is disabled.
The UDP session timeout duration is 120 secs. The phones are set to re-register every minute.
What could be wrong?

TS1559 I have done all of the steps that Apple have said to do, including completely restarting my phone and registering it as a new one, but still my wifi is greyed out, can anyone help me?

I have done all of the steps that Apple have said to do, including completely restarting my phone and registering it as a new one, but still my wifi is greyed out, can anyone help me?

http://support.apple.com/kb/ht5621

CUCM 8.6 - 8961 "Phone Not Registered"

Hi,
I can't get my 8961 to register to my CUCM 8.6. My 7941 registered without a problem. I verified that I correctly entered the MAC Address. My 8961 can reach the TFTP server and get its TFTP file.
Status Messages:
00:41:37 VPN Error :VPN is not configured.
00:41:36 SEPXXXXXXXXXXXX.cnf.xml.sgn (HTTP)
00:41:33 Trust List Updated
00:41:32 Updating Trust List
I verified the traces on CUCM and the phone really gets its tftp file. Those 4 messages loop and phone displays : "Phone Not Registered". In the CUCM Admin page, the registration status says : Unknown..
Does anyone have an idea?
Thank you

The usual issue when SIP phones do not register is when there is no line configured on the device. SIP phones require a line number to be configured if they are not auto registering because the phone has to learn of it's directory number from the TFTP config file so that it knows what to put in the SIP REGISTER message. Double check that there is a line set on the device in CUCM and if there's not add one and the phone should register.

Since upgrading to iOS7 some text messages are not being sent, iMessage on my Ipad and MacBook doesn't have my phone number registered and my reminder colour on my Iphone keeps changing ... among other things

Since upgrading to iOS7 some text messages are not being sent on my iPhone 5, iMessage on my Ipads and MacBook doesn't have my phone number registered and my reminder colour on my Iphone keeps changing ... among other things. Any suggestions to fix these faults?

Make sure People Sending you via iMessage are sending their messages on your PHONE number not on your EMAIL addresses . My sister had the same problem, thats how i managed to solve it!

My iphone 4s was registered to my home laptop, but i done an itunes update to my works computer then plugged in my phone to listen to music now my phone is registered on work computer, how do i reconnect to my home laptop

my iphone 4s was linked to itunes on my home laptop for backups and syncing. at work yesterday i updated itunes to 11 and later on i plugged in my phone to listen to music and my phone registered to work computer and when i plugged into my home laptop it will not reconise it for backups etc, how do i get my phone to register back to my laptop.

you connect it by usb and tell it to sync from itunes under devices

VoIP Phones - Testing Latency, Jitter, and Packet Loss

I am having big problems with my VoIP phone connection and I'll try to lay it out clearly here.
The main telephone system resides at Location A (static IP address - see below - xxx.xxx.206.19), which has a network connection of 50MB down/20MB up (i.e., very fast). The VoIP phone configured for that system resides at Location B, which has a network connection of 10MB down/1MB up (i.e., also fast, or at least fast enough "on paper" for a quality VoIP connection). The LAN at Location A uses an Airport Extreme router, which does not have QOS or EF capability. The LAN at Location B uses a D-Link DIR-655 router which does have QOS that is configured properly to direct all traffic to the VoIP phone's IP address.
The VoIP phone at Location B is having intermittent call quality problems with skipping of words, hollowing out noises, jittery conversations, etc. All the inquiries I've made to the ISPs and phone system manufacturer (ESI) suggest that my base Internet speeds are not the problem.
I'm told, instead, that the problem might be latency, jitter, or packet loss between Location A and Location B. This leads to several questions:
(1)     Is there any Mac software that can test latency, jitter, and packet loss? I've looked at Network Utility and it seems to only measure a few things.
(2)     Does anyone see anything in the following Traceroute and Ping results (done twice from Location B to Location A) that looks problematic to VoIP quality?:
Traceroute:
First run: Traceroute has started…
traceroute to xxx.xxx.206.19 (xxx.xxx.206.19), 64 hops max, 72 byte packets
1 alfirving (192.168.0.1) 0.569 ms 0.363 ms 0.302 ms
2 10.72.28.1 (10.72.28.1) 27.567 ms 18.161 ms 22.288 ms
3 70.125.216.150 (70.125.216.150) 9.841 ms 10.346 ms 9.497 ms
4 24.164.209.116 (24.164.209.116) 11.042 ms 8.298 ms 9.433 ms
5 70.125.216.108 (70.125.216.108) 21.068 ms 20.657 ms 12.045 ms
6 te0-8-0-2.dllatxl3-cr01.texas.rr.com (72.179.205.48) 11.154 ms 11.540 ms 24.495 ms
7 107.14.17.136 (107.14.17.136) 11.994 ms 14.217 ms 15.816 ms
8 ae-3-0.pr0.dfw10.tbone.rr.com (66.109.6.209) 14.566 ms 32.670 ms 15.947 ms
9 ix-0-3-2-0.tcore2.dt8-dallas.as6453.net (209.58.47.105) 11.647 ms 12.260 ms 12.386 ms
10 if-2-2.tcore1.dt8-dallas.as6453.net (66.110.56.5) 10.023 ms 12.285 ms 12.338 ms
11 209.58.47.74 (209.58.47.74) 17.641 ms 16.741 ms 16.372 ms
12 0.ae2.xl3.dfw7.alter.net (152.63.97.57) 11.584 ms 12.315 ms 12.890 ms
13 0.so-6-1-0.dfw01-bb-rtr1.verizon-gni.net (152.63.1.90) 13.812 ms
    0.ge-3-0-0.dfw01-bb-rtr1.verizon-gni.net (152.63.1.17) 18.831 ms
    130.81.23.164 (130.81.23.164) 14.189 ms
14 p14-0-0.dllstx-lcr-05.verizon-gni.net (130.81.27.40) 14.561 ms 13.621 ms 15.544 ms
15 * * *
16 static-xxx.xxx.206.19.dllstx.fios.verizon.net (xxx.xxx.206.19) 23.125 ms 24.136 ms 22.411 ms
Second run: Traceroute has started…
traceroute to xxx.xxx.206.19 (xxx.xxx.206.19), 64 hops max, 72 byte packets
1 alfirving (192.168.0.1) 0.603 ms 0.420 ms 0.324 ms
2 10.72.28.1 (10.72.28.1) 40.494 ms 26.625 ms 14.152 ms
3 70.125.216.150 (70.125.216.150) 9.431 ms 9.660 ms 9.018 ms
4 24.164.209.116 (24.164.209.116) 16.293 ms 12.339 ms 19.252 ms
5 70.125.216.108 (70.125.216.108) 15.801 ms 11.438 ms 12.068 ms
6 te0-8-0-2.dllatxl3-cr01.texas.rr.com (72.179.205.48) 23.221 ms 30.459 ms 17.519 ms
7 107.14.17.136 (107.14.17.136) 14.611 ms 15.696 ms 15.775 ms
8 ae-3-0.pr0.dfw10.tbone.rr.com (66.109.6.209) 17.643 ms 14.812 ms 16.294 ms
9 ix-0-3-2-0.tcore2.dt8-dallas.as6453.net (209.58.47.105) 11.169 ms 12.374 ms 9.849 ms
10 if-2-2.tcore1.dt8-dallas.as6453.net (66.110.56.5) 16.453 ms 12.168 ms 12.384 ms
11 209.58.47.74 (209.58.47.74) 18.015 ms 14.867 ms 16.432 ms
12 0.ae2.xl3.dfw7.alter.net (152.63.97.57) 11.471 ms 11.993 ms 12.395 ms
13 0.ge-6-3-0.dfw01-bb-rtr1.verizon-gni.net (152.63.96.42) 14.077 ms 29.153 ms
    0.ge-3-0-0.dfw01-bb-rtr1.verizon-gni.net (152.63.1.17) 17.962 ms
14 p14-0-0.dllstx-lcr-05.verizon-gni.net (130.81.27.40) 14.629 ms 12.297 ms 12.839 ms
15 * * *
16 static-xxx.xxx.206.19.dllstx.fios.verizon.net (xxx.xxx.206.19) 24.976 ms 22.170 ms 22.376 ms
Ping:
First Run: Ping has started…
PING xxx.xxx.206.19 (xxx.xxx.206.19): 56 data bytes
64 bytes from xxx.xxx.206.19: icmp_seq=0 ttl=242 time=22.814 ms
64 bytes from xxx.xxx.206.19: icmp_seq=1 ttl=242 time=24.621 ms
64 bytes from xxx.xxx.206.19: icmp_seq=2 ttl=242 time=24.711 ms
64 bytes from xxx.xxx.206.19: icmp_seq=3 ttl=242 time=24.109 ms
64 bytes from xxx.xxx.206.19: icmp_seq=4 ttl=242 time=23.336 ms
64 bytes from xxx.xxx.206.19: icmp_seq=5 ttl=242 time=25.644 ms
64 bytes from xxx.xxx.206.19: icmp_seq=6 ttl=242 time=27.755 ms
64 bytes from xxx.xxx.206.19: icmp_seq=7 ttl=242 time=25.135 ms
64 bytes from xxx.xxx.206.19: icmp_seq=8 ttl=242 time=22.443 ms
64 bytes from xxx.xxx.206.19: icmp_seq=9 ttl=242 time=24.635 ms
--- xxx.xxx.206.19 ping statistics ---
10 packets transmitted, 10 packets received, 0.0% packet loss
round-trip min/avg/max/stddev = 22.443/24.520/27.755/1.448 ms
Second Run: Ping has started…
PING xxx.xxx.206.19 (xxx.xxx.206.19): 56 data bytes
64 bytes from xxx.xxx.206.19: icmp_seq=0 ttl=242 time=27.183 ms
64 bytes from xxx.xxx.206.19: icmp_seq=1 ttl=242 time=24.629 ms
64 bytes from xxx.xxx.206.19: icmp_seq=2 ttl=242 time=22.511 ms
64 bytes from xxx.xxx.206.19: icmp_seq=3 ttl=242 time=39.620 ms
64 bytes from xxx.xxx.206.19: icmp_seq=4 ttl=242 time=26.722 ms
64 bytes from xxx.xxx.206.19: icmp_seq=5 ttl=242 time=23.183 ms
64 bytes from xxx.xxx.206.19: icmp_seq=6 ttl=242 time=25.171 ms
64 bytes from xxx.xxx.206.19: icmp_seq=7 ttl=242 time=24.412 ms
64 bytes from xxx.xxx.206.19: icmp_seq=8 ttl=242 time=23.837 ms
64 bytes from xxx.xxx.206.19: icmp_seq=9 ttl=242 time=23.785 ms
--- xxx.xxx.206.19 ping statistics ---
10 packets transmitted, 10 packets received, 0.0% packet loss
round-trip min/avg/max/stddev = 22.511/26.105/39.620/4.713 ms
(3) Any other ideas on what my call quality problem might be, or how I can tweak it? For example, would putting a DIR-655 router at Location A and enabling QOS really make a difference?
Thanks to everyone, and I hope this is not too long or difficult to understand.

Hey thanks for your reply Yeah im only getting 1 ro sometimes 2 bars reception so hopefully the antenna will beef things up but I think it is what it is perhaps.

VoIP Phone licenses

Hello there,
I'm pretty new to VoIP so please bare with me.. Reading about VoIP phones I've found that you have to buy some kind of license no matter which phone you are using : " All Cisco Unified IP phones require the purchase of a phone user license, regardless of call protocol being used. " Please can someone give me a link or enlighten me a little about them.
Questions :
1. what are these licenses, why should I buy them, how much do they cost ?
2. Is this the only license I'll have to buy to use the ip phone ? Or I'll need some kind of software license too?
Thank you

DLUs = Device License Unit
depending on the phone model how many they use, you need to contact pre-sales or licensing for them
if you don't have the DLUs you cannot add phones
the SKU from each phone entitles you to the necessary DLUs for that phone but DLUs can be purchased separately
you also need a node license to activate the CCM service if 5.x and an extra feature license if running 6.x
you can look "licensing" on cco and will get more info
HTH
java
if this helps, please rate

DHCP - Cannot add text option for VOIP phones in OES Linux

While working through this, I solved the issue, but decided to post this anyway as it may help others to find these sorts of errors.
I'm working on migrating from NetWare 6.5sp8 to OES11sp2. Client has Shoretel VOIP phones. Existing NetWare-based DHCP has no problem. Option 156 has been configured to give out the required text information that Shoretel phones require.
Problem is that I could not get the OES11 DHCP to run with that option. Nor could I migrate the existing option over - the Migration Tool (in OES11) says it successfully migrates DHCP, but I cannot start the dhcpd daemon. Error is that it failed, and in the rc.dhcpd.log file I see an error:
LDAP Line 26: unknown option dhcp.Shoretel_Boot.
LDAP Line 26: unexpected end of file
LDAP: cannot parse dhcpService entry 'cn=newdhcpservice,o=LIBRARY'
Configuration file errors encountered -- exiting
If I look in the file (created when LDAP reads DHCP config from eDirectory apparently) dhcp-ldap-startup.log I can see the problem entry at line 26:
option Shoretel_Boot "FTPSERVERS=172.30.43.8,COUNTRY=1,LANGUAGE=1,L AYER default-lease-time 259200 ;
This option does NOT show up in the newdhcpservice option when I look at it in ConsoleOne, or DSBROWSE, or DNS/DHCP Management Console.
This option DOES show up in the DNS/DHCPManagement Console if I look at the DHCP (NetWare) tab and look at Other DHCP Options for some of the configured subnets, but it actually has different text from the above, specifically:
FTPSERVERS=172.30.43.8,COUNTRY=1,LANGUAGE=1,LAYER2 TAGGING=1,VLANID=9
Note that it does not have a " character anywhere in the entry. This option is configured as a Global DHCP text option.
Novell TID 7009464 mentions the issue, though not for Option 156. In that TID there is this:
"Situation #2
Migrate a working DHCP server with DHCP options that are of type "Text" to an OES server.
Load the DHCP server service... it fails to load and gives similar errors to the ones listed above."
Under resolution the TID says to delete and recreate the dhcp service object without the text option and it will load. That doesn't work for me as I still get an LDAP error pointing to the Shoretel_Boot unknown option. (I dare not try deleting it from the NetWare DHCP config and risk breaking the client's phone system).
One of the options in the TID to fix this is to re-enter the data using the DNS/DHCP Management Console - but that didn't work.
Here is the answer:
First, the log files are misleading. The error message points to not being able to read the newdhcpservice object entry - but the problem was elsewhere. In fact the problem showed up in the logs even when there were no option 156 entries at all in any object inside the newdhcpservice or the newdhcpservice object itself. The problem existed in the NetWare configuration of the object for one of the dhcp subnets.
Specifically, there was an illegal character in the text entry for option 156 - the # character was in there, like this:
FTPSERVERS=172.30.43.8,COUNTRY=1,LANGUAGE=1,LAYER# 2TAGGING=1,VLANID=9
If you look at the error log entry for syntax error you can see that the option 156 text stopped at the # symbol, and then default-lease-time was appended to the end.
Removing the # symbol got things working.
Craig Johnson
(former Novell partner / sysop)

On 30/08/2014 21:16, phxazcraig wrote:
> While working through this, I solved the issue, but decided to post this
> anyway as it may help others to find these sorts of errors.
>
> I'm working on migrating from NetWare 6.5sp8 to OES11sp2. Client has
> Shoretel VOIP phones. Existing NetWare-based DHCP has no problem.
> Option 156 has been configured to give out the required text information
> that Shoretel phones require.
>
> Problem is that I could not get the OES11 DHCP to run with that option.
> Nor could I migrate the existing option over - the Migration Tool (in
> OES11) says it successfully migrates DHCP, but I cannot start the dhcpd
> daemon. Error is that it failed, and in the rc.dhcpd.log file I see
> an error:
>
> LDAP Line 26: unknown option dhcp.Shoretel_Boot.
> LDAP Line 26: unexpected end of file
> LDAP: cannot parse dhcpService entry 'cn=newdhcpservice,o=LIBRARY'
> Configuration file errors encountered -- exiting
>
>
> If I look in the file (created when LDAP reads DHCP config from
> eDirectory apparently) dhcp-ldap-startup.log I can see the problem entry
> at line 26:
>
> option Shoretel_Boot
> "FTPSERVERS=172.30.43.8,COUNTRY=1,LANGUAGE=1,L AYER default-lease-time
> 259200 ;
>
>
> This option does NOT show up in the newdhcpservice option when I look at
> it in ConsoleOne, or DSBROWSE, or DNS/DHCP Management Console.
>
> This option DOES show up in the DNS/DHCPManagement Console if I look at
> the DHCP (NetWare) tab and look at Other DHCP Options for some of the
> configured subnets, but it actually has different text from the above,
> specifically:
>
> FTPSERVERS=172.30.43.8,COUNTRY=1,LANGUAGE=1,LAYER2 TAGGING=1,VLANID=9
>
> Note that it does not have a " character anywhere in the entry. This
> option is configured as a Global DHCP text option.
>
> Novell TID 7009464 mentions the issue, though not for Option 156. In
> that TID there is this:
> "Situation #2
> Migrate a working DHCP server with DHCP options that are of type
> "Text" to an OES server.
> Load the DHCP server service... it fails to load and gives similar
> errors to the ones listed above."
>
> Under resolution the TID says to delete and recreate the dhcp service
> object without the text option and it will load. That doesn't work for
> me as I still get an LDAP error pointing to the Shoretel_Boot unknown
> option. (I dare not try deleting it from the NetWare DHCP config and
> risk breaking the client's phone system).
>
> One of the options in the TID to fix this is to re-enter the data using
> the DNS/DHCP Management Console - but that didn't work.
>
> Here is the answer:
> First, the log files are misleading. The error message points to not
> being able to read the newdhcpservice object entry - but the problem was
> elsewhere. In fact the problem showed up in the logs even when there
> were no option 156 entries at all in any object inside the
> newdhcpservice or the newdhcpservice object itself. The problem
> existed in the NetWare configuration of the object for one of the dhcp
> subnets.
>
> Specifically, there was an illegal character in the text entry for
> option 156 - the # character was in there, like this:
>
> FTPSERVERS=172.30.43.8,COUNTRY=1,LANGUAGE=1,LAYER# 2TAGGING=1,VLANID=9
>
> If you look at the error log entry for syntax error you can see that the
> option 156 text stopped at the # symbol, and then default-lease-time was
> appended to the end.
>
> Removing the # symbol got things working.
>
> Craig Johnson
> (former Novell partner / sysop)
Thanks for taking the time to post the above as I'm sure it will help
someone else in the future.
Simon
Novell Knowledge Partner
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