WRT54GL & Cisco 7960G VoIP Phone
Hello fellow Cisco users,
I use a Cisco VoIP phone, the ubiquitous 7960G IP phone, and have no problem using it with non-Cisco routers. The phone does not seem to work with the Cisco WRT54GL router under the setup I have as explained below.
I usually connect my Cisco IP phone directly to my laptop's ethernet port giving the phone a shared internet connection. This arrangement works fine with two non-Cisco routers I've used. With the Cisco WRT54GL router however, the phone cannot get an IP configuration, even after opening the needed ports, leading me to believe the problem is likely to be something with the router. I have also tried setting up a DMZ for the router without success. The biggest problem is that the phone uses DHCP (which I also prefer) and the DMZ and port forwarding features require I set the phone's IP address.
I would greatly appreciate any thoughts on how to resolve this.
Many thanks for suggestions.
PS: I forgot to add that the phone works (i.e. receive and make calls) when plugged via ethernet cable into the router. Apart from the problem of having an ethernet cable running across a room, this setup also has some features missing like date and time so calls aren't stamped.
I think your phone is not getting IP address when it is connected to the computer. But when you connect it directly to the modem, it works.
You can try to upgrade/re-flash the firmware on your router.
Connect the computer with the Ethernet cable to the router.
Download the latest firmware from Linksys website and save it on your computer. Open the setup page of the router. Click on Administration tab and go to Firmware upgrade sub tab. Browse the firmware file that you have already downloaded and upgrade it on your router.
After upgrading the firmware on the router, it is recommended that you should reset the router and reconfigure it. Press and hold the reset button on the router for 30 seconds. Release the reset button and wait for 30 seconds. Power cycle the router and reconfigure it.
Also try to change some advanced wireless settings on the router. Click on Wireless tab and go to Advanced wireless settings. Change the Beacon interval to 75, RTS threshold to 2307 and fragmentation threshold to 2306. Save the settings.
Similar Messages
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Issue defining DN 7965 because using 7965 SD Phone button template on Cisco 7965 VoIP phones
I am having issue with assigning DN7965 to a phone device. Error message indicates that 7965 is already associated and in use. We have associated phone template 7965SD because most our phones are CISCO 7965. Is there a way to overcome this issue ?
Hello, All!
I have continued testing this in my lab and have come up with the following results.
When setting up a new device, TAPS will generate an error if the DN is already in existence across multiple devices.
If replacing a device, you must use the same model device or else TAPS will generate an error.
Unfortunately, based on the above rules, I was unable to utilize TAPS for bulk upgrade of user devices from 7960 model phones to 7965 model phones. However, I have come up with an alternate solution.
Note: This solution would only need to be used in a scenario where you do not want to assign a MAC address before physically deploying the equipment. The below solution provides the flexibility of creating the phone configurations prior to deploying the equipment without having to have all hardware specifically assigned to a user/DN.
Bulk Update Cisco Phones without MAC Addresses
Bulk Admin > Phone > Export Phones > Specific Details (Pull relevant information, DN(s), Description, Owner User ID, CSS, so on…)
Download the file, change to csv, open in Excel
Clear contents of “Device Name” field. Parse through and update other fields if needed.
Save a copy as CSV, upload to CUCM as Insert Phones > Specific Details
In CUCM, create a Phone Template for import
Bulk Admin > Phone > Insert Phones > Choose File > Choose Phone Template > **Important** Select Create Dummy MAC
Run Job
Once job is completed, all 7965 devices will be created with a BAT prefix and dummy MAC address, check Device > Phone to confirm
…When ready to activate
Device > Phone > Search by DN for BAT device
Open device, change MAC address to new 7965 hardware, check “Is Active”
Click Save.
Plug-in 7965 device to network. When the phone comes online and gets past firmware it will pull newly generated SEP config with user ext.
I hope this helps. I appreciate the communities continued assistance. -
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 youDLUs = 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 -
Hello,
So I am wondering if the issue with VoIP phones is universal with the LRT224 Phone? Has this problem been reproduced in the Linksys labs? Will this issue be addressed by Linksys Engineers?
Would have settuping up VLAN's worked?Hey,
Sorry for the delayed response. We have one LRT224 with two WAN Connections and one LAPN600 for the access point. Our network is completely wireless with around 30 devices total (7 IP Phones). We use Cisco SPA525G Phones.
Our VoIP provider is Jive Communications.
I am using the EA4500 and everything works fine on it. The main issue I am having with these phones going through the LRT224 is that phone calls were not coming, and you could not dial other extension throughout the office.
If you have any other questions, please let me know. -
We have been tasked with deploying VoIP phones in our patient rooms at our hospital. So far we have only deployed them internally. We do use voice vlans for each floor that gets ip phones. We have a Session Border Controller in our DMZ for users that have our IP phones at their homes. A few thoughts have come up about using port security on switch ports to lock down the MAC address on the port. But windows now allows users to change their MAC address without the need for spoofing software. Not to mention the calls we would get at 2:00 am when the staff replaces an ip phone. We could use some kind of ACL on the port but it would always be changing as we deploy more and more ip phones. So we are kind of stuck. Has anyone went through this type of deployment and how did you do it? Maybe an internal SBC? Thoughts???
Sent from Cisco Technical Support iPhone AppA few thoughts have come up about using port security on switch ports to lock down the MAC address on the port.
No need. Just make sure the PC ports at the back of the phones are disabled or get the phones without the second port.
But this will not stop someone from unplugging the phone. So you'll need to employ a cable lock, like the following:
http://www.panduit.com/wcs/Satellite?c=Page&childpagename=Panduit_Global%2FPG_Layout&cid=1345565612156&packedargs=classification_id%3D651%26locale%3Den_us&pagename=PG_Wrapper
http://www.panduit.com/wcs/Satellite?c=Page&childpagename=Panduit_Global%2FPG_Layout&cid=1345565612156&packedargs=classification_id%3D652%26locale%3Den_us&pagename=PG_Wrapper
Another method you could try to use is make sure that the port where the phones are in public do not have any data VLANs configured.
Trying to hardcode an IP address to a MAC address is not foolproof. Anyone can spoof a MAC address and coding the values ain't easy. -
Problem transferring calls between Voip phones when the call is originated
Hi,
I have configured a gateway h323 (3640- 12.3) to communicate with Cisco Callmanager 3.3(4).
On the H323 gateway I have installed one BRI interface and one FXS interface.
Inside my network I am using VoIP telephones 7910.
Originating calls in the Voip telephones which destination is the PSTN is working fine;
Receiving calls from PSTN that destination is VoIP telephones is OK also;
Redirecting or doing Hold on calls between VoiP telephones works but ..
When redirecting or doing Hold on calls between a VoiP telephone and a call from PSTN I loose the voice stream.
Apparently, all signalling is working fine because when redirecting the call originated in the PSTN from one VoIP telephone to another VoIP telephone, I can see the Calling number ID. Even if I use the telephone keyboard, I can listen the tones, but I have no voice stream.
Please any help will be welcome.
Thanks,
David CostaI have tried different configurations and now it is working. The problem is that I couldnt identify were the miss configuration was.
Another question is: Can I use a codec with compression between voip interface of h323 gateway and the Callmanager or Voip phones to pass music on hold?
Now I only pass music on hold if I configure G.711 on the voip interface of h323 gateway.
Thanks,
David Costa -
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? -
External Ringer for VOIP Phone
I have external ringers connected to VG224's for VOIP phones in our shops. I have them configured and they work as they should for a while, but after a month or two it appears to blow up the port on the VG it is attached to. I am thinking it might have to do with resistance of the bells, but since I am not an electrician I am not sure. The ringer is a Premier PT-102 bell and per the description is 1000/2650 Ohm. Has anybody experienced problem with external ringers blowing up VG ports. Or is there any alternatives to the analog ringers that is designed to work with the Cisco VOIP phones. Thanks for any info anybody can supply.
I did get one of the Algo 8180 SIP Audio Alerters and it works great as an external ringer. Also since it has the capability of paging, we are trying to get it configured to use with our IPCelerate paging solution. Does anybody have any thoughts or ideas how to accomplish this. I can get it to page using a direct extension to the Algo device, but can not get it to receive pages from IPCelerate. Any help would be appreciated.
-
German menu language on 7960G SIP Phone
Hi All
I have flashed a 7960G SCCP Phone to SIP Firmware.
Is there any possibility to give this phone another menu language, like german?
Or is this running only on SCCP Firmware?
Thank youHi,
thank you for the help.
Now I have these files in my TFTP Server.
These are my files:
OS79XX.TXT
P0S3-08-12-00
SEP0014A8924D6D.CNF.XML
<device>
<devicePool>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
</ports>
<processNodeName> </processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<versionStamp>{Jan 01 2005 00:00:00}</versionStamp>
<loadInformation>P0S3-08-12-00</loadInformation>
<userLocale>
<name>German_Germany</name>
<langCode>de</langCode>
</userLocale>
<networkLocale>Germany</networkLocale>
<idleTimeout>0</idleTimeout>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL></servicesURL>
</device>
SIP0014A8924D6D.cnf
# SIP Configuration Generic File
# Line 1 appearance
line1_name: 01010101001
# Line 1 Registration Authentication
line1_authname: "UNPROVISIONED"
# Line 1 Registration Password
line1_password: "UNPROVISIONED"
# Line 2 appearance
line2_name: football
# Line 2 Registration Authentication
line2_authname: "UNPROVISIONED"
# Line 2 Registration Password
line2_password: "UNPROVISIONED"
####### New Parameters added in Release 2.0 #######
# All user_parameters have been removed
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "" ; Has no effect on SIP messaging
# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "User ID"
# Line 2 Display Name (Display name to use for SIP messaging)
line2_displayname: ""
####### New Parameters added in Release 3.0 ######
# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt: "SIP Phone" ; Limited to 15 characters (Default - SIP Phone)
# Phone Password (Password to be used for console or telnet login)
phone_password: "cisco" ; Limited to 31 characters (Default - cisco)
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none
SIPDefault.cnf
# SIP Default Generic Configuration File
# Image Version
image_version: P0S381200
language: german
# Proxy Server
proxy1_address: "" ; Can be dotted IP or FQDN
proxy2_address: "" ; Can be dotted IP or FQDN
proxy3_address: "" ; Can be dotted IP or FQDN
proxy4_address: "" ; Can be dotted IP or FQDN
proxy5_address: "" ; Can be dotted IP or FQDN
proxy6_address: "" ; Can be dotted IP or FQDN
# Proxy Server Port (default - 5060)
proxy1_port: 5060
proxy2_port: 5060
proxy3_port: 5060
proxy4_port: 5060
proxy5_port: 5060
proxy6_port: 5060
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 0
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 3600
# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: g711ulaw
# TOS bits in media stream [0-5] (Default - 5)
tos_media: 5
# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: 1
# Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt )
dtmf_outofband: avt
# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)
dtmf_db_level: 3
# SIP Timers
timer_t1: 500 ; Default 500 msec
timer_t2: 4000 ; Default 4 sec
sip_retx: 10 ; Default 10
sip_invite_retx: 6 ; Default 6
timer_invite_expires: 180 ; Default 180 sec
####### New Parameters added in Release 2.0 #######
# Dialplan template (.xml format file relative to the TFTP root directory)
dial_template: dialplan
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "" ; Example: ./sip_phone/
# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: "" ; SNTP Server IP Address
sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: EST ; Time Zone Phone is in
dst_offset: 1 ; Offset from Phone's time when DST is in effect
dst_start_month: April ; Month in which DST starts
dst_start_day: "" ; Day of month in which DST starts
dst_start_day_of_week: Sun ; Day of week in which DST starts
dst_start_week_of_month: 1 ; Week of month in which DST starts
dst_start_time: 02 ; Time of day in which DST starts
dst_stop_month: Oct ; Month in which DST stops
dst_stop_day: "" ; Day of month in which DST stops
dst_stop_day_of_week: Sunday ; Day of week in which DST stops
dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month
dst_stop_time: 2 ; Time of day in which DST stops
dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: 0 ; Default 0 (Do Not Disturb feature is off)
# Caller ID Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous)
# Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls)
# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: 101 ; Default 101
# Sync value of the phone used for remote reset
sync: 1 ; Default 1
####### New Parameters added in Release 2.1 #######
# Backup Proxy Support
proxy_backup: "" ; Dotted IP of Backup Proxy
proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)
# Emergency Proxy Support
proxy_emergency: "" ; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060)
# Configurable VAD option
enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable
####### New Parameters added in Release 2.2 ######
# NAT/Firewall Traversal
nat_enable: 0 ; 0-Disabled (default), 1-Enabled
nat_address: "" ; WAN IP address of NAT box (dotted IP or DNS A record only)
voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060)
start_media_port: 16384 ; Start RTP range for media (default - 16384)
end_media_port: 32766 ; End RTP range for media (default - 32766)
nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled
# Outbound Proxy Support
outbound_proxy: "" ; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060 ; default is 5060
####### New Parameter added in Release 3.0 #######
# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default)
####### New Parameters added in Release 3.1 #######
# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: 1 ; 0-Disabled, 1-Enabled (default)
# Telnet Level (enable or disable the ability to telnet into the phone)
telnet_level: 1 ; 0-Disabled (default), 1-Enabled, 2-Privileged
####### New Parameters added in Release 4.0 #######
# XML URLs
services_url: "" ; URL for external Phone Services
directory_url: "" ; URL for external Directory location
logo_url: "" ; URL for branding logo to be used on phone display
# HTTP Proxy Support
http_proxy_addr: "" ; Address of HTTP Proxy server
http_proxy_port: 80 ; Port of HTTP Proxy Server (80-default)
# Dynamic DNS/TFTP Support
dyn_dns_addr_1: "" ; restricted to dotted IP
dyn_dns_addr_2: "" ; restricted to dotted IP
dyn_tftp_addr: "" ; restricted to dotted IP
# Remote Party ID
remote_party_id: 0 ; 0-Disabled (default), 1-Enabled
####### New Parameters added in Release 4.4 #######
# Call Hold Ringback (0-off, 1-on, 2-off with no user control, 3-on with no user control)
call_hold_ringback: 0 ; Default 0 (Call Hold Ringback feature is off)
####### New Parameters added in Release 6.0 #######
# Dialtone Stutter for MWI
stutter_msg_waiting: 0 ; 0-Disabled (default), 1-Enabled
# RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 0 ; 0-Disabled (default), 1-Enabled
xmlDefault.CNF.XML
<?xml version="1.0"?>
-<Default>
-<callManagerGroup>
-<members>
-<member priority="0">
-<callManager>
-<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
</ports>
<processNodeName/>
</callManager>
</member>
</members>
<loadInformation6 model="IP Phone 7910"/>
<loadInformation124 model="Addon 7914"/>
<loadInformation9 model="IP Phone 7935"/>
<loadInformation8 model="IP Phone 7940"/>
<loadInformation7 model="IP Phone 7960">P0S3-8-12-00</loadInformation7>
<loadInformation20000 model="IP Phone 7905"/>
<loadInformation30008 model="IP Phone 7902"/>
<loadInformation30002 model="IP Phone 7920"/>
<loadInformation30019 model="IP Phone 7936"/>
<loadInformation30006 model="IP Phone 7970"/>
<loadInformation30018 model="IP Phone 7961"/>
<loadInformation30007 model="IP Phone 7912"/>
</callManagerGroup>
</Default>
The folders "German_Germany"; "germany" and the file German_Germany.aar are in TFTP folder,too. But my phone is doing nothing with this files.
This is the log of TFTP Server:
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:21:54.559]
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:21:58.562]
Rcvd DHCP Rqst Msg for IP 0.0.0.0, Mac 00:14:A8:92:4D:6D [04/03 13:22:45.892]
Previously allocated address 192.168.0.6 acked [04/03 13:22:45.893]
Connection received from 192.168.0.6 on port 50798 [04/03 13:22:46.037]
Read request for file <CTLSEP0014A8924D6D.tlv>. Mode octet [04/03 13:22:46.038]
File <CTLSEP0014A8924D6D.tlv> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:22:46.038]
Connection received from 192.168.0.6 on port 50798 [04/03 13:22:47.036]
Read request for file <CTLSEP0014A8924D6D.tlv>. Mode octet [04/03 13:22:47.036]
File <CTLSEP0014A8924D6D.tlv> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:22:47.036]
Connection received from 192.168.0.6 on port 50798 [04/03 13:22:51.035]
Read request for file <CTLSEP0014A8924D6D.tlv>. Mode octet [04/03 13:22:51.036]
File <CTLSEP0014A8924D6D.tlv> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:22:51.036]
Connection received from 192.168.0.6 on port 50799 [04/03 13:22:51.056]
Read request for file <SEP0014A8924D6D.cnf.xml>. Mode octet [04/03 13:22:51.057]
File <SEP0014A8924D6D.cnf.xml> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:22:51.057]
Connection received from 192.168.0.6 on port 50800 [04/03 13:22:51.085]
Read request for file <SIP0014A8924D6D.cnf>. Mode octet [04/03 13:22:51.086]
File <SIP0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:22:51.086]
Connection received from 192.168.0.6 on port 50801 [04/03 13:22:51.105]
Read request for file <MGC0014A8924D6D.cnf>. Mode octet [04/03 13:22:51.105]
File <MGC0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:22:51.105]
Connection received from 192.168.0.6 on port 50802 [04/03 13:22:51.124]
Read request for file <XMLDefault.cnf.xml>. Mode octet [04/03 13:22:51.127]
Using local port 65426 [04/03 13:22:51.127]
<XMLDefault.cnf.xml>: sent 3 blks, 1077 bytes in 0 s. 0 blk resent [04/03 13:22:51.135]
Connection received from 192.168.0.6 on port 50803 [04/03 13:22:51.180]
Read request for file <P0S3-8-12-00.loads>. Mode octet [04/03 13:22:51.180]
Using local port 65427 [04/03 13:22:51.180]
<P0S3-8-12-00.loads>: sent 1 blk, 458 bytes in 0 s. 0 blk resent [04/03 13:22:51.183]
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:24:17.151]
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:24:20.150]
Rcvd DHCP Rqst Msg for IP 0.0.0.0, Mac 00:14:A8:92:4D:6D [04/03 13:24:34.219]
Previously allocated address 192.168.0.6 acked [04/03 13:24:34.220]
Connection received from 192.168.0.6 on port 50823 [04/03 13:24:34.258]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:24:34.259]
Using local port 64133 [04/03 13:24:34.259]
Connection received from 192.168.0.6 on port 50823 [04/03 13:24:35.250]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:24:35.250]
Using local port 64134 [04/03 13:24:35.250]
Connection received from 192.168.0.6 on port 50823 [04/03 13:24:39.250]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:24:39.250]
Using local port 64135 [04/03 13:24:39.250]
<SIPDefault.cnf>: sent 13 blks, 6203 bytes in 0 s. 0 blk resent [04/03 13:24:39.284]
Connection received from 192.168.0.6 on port 50824 [04/03 13:24:39.485]
Read request for file <SIP0014A8924D6D.cnf>. Mode octet [04/03 13:24:39.485]
File <SIP0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:24:39.485]
TIMEOUT waiting for Ack block #1 [04/03 13:24:49.260]
TIMEOUT waiting for Ack block #1 [04/03 13:24:50.251]
Rcvd DHCP Rqst Msg for IP 0.0.0.0, Mac 00:14:A8:92:4D:6D [04/03 13:26:50.610]
Previously allocated address 192.168.0.6 acked [04/03 13:26:50.611]
Connection received from 192.168.0.6 on port 50857 [04/03 13:26:50.649]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:26:50.649]
Using local port 64137 [04/03 13:26:50.649]
Connection received from 192.168.0.6 on port 50857 [04/03 13:26:51.641]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:26:51.642]
Using local port 64138 [04/03 13:26:51.642]
Connection received from 192.168.0.6 on port 50857 [04/03 13:26:55.641]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:26:55.641]
Using local port 64139 [04/03 13:26:55.642]
<SIPDefault.cnf>: sent 13 blks, 6203 bytes in 0 s. 0 blk resent [04/03 13:26:55.676]
Connection received from 192.168.0.6 on port 50858 [04/03 13:26:55.874]
Read request for file <SIP0014A8924D6D.cnf>. Mode octet [04/03 13:26:55.875]
File <SIP0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:26:55.875]
TIMEOUT waiting for Ack block #1 [04/03 13:27:05.651]
TIMEOUT waiting for Ack block #1 [04/03 13:27:06.645]
Rcvd DHCP Rqst Msg for IP 0.0.0.0, Mac 00:14:A8:92:4D:6D [04/03 13:28:40.670]
Previously allocated address 192.168.0.6 acked [04/03 13:28:40.671]
Connection received from 192.168.0.6 on port 50797 [04/03 13:28:40.813]
Read request for file <CTLSEP0014A8924D6D.tlv>. Mode octet [04/03 13:28:40.813]
File <CTLSEP0014A8924D6D.tlv> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:28:40.814]
Connection received from 192.168.0.6 on port 50797 [04/03 13:28:41.803]
Read request for file <CTLSEP0014A8924D6D.tlv>. Mode octet [04/03 13:28:41.804]
File <CTLSEP0014A8924D6D.tlv> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:28:41.804]
Connection received from 192.168.0.6 on port 50797 [04/03 13:28:45.803]
Read request for file <CTLSEP0014A8924D6D.tlv>. Mode octet [04/03 13:28:45.803]
File <CTLSEP0014A8924D6D.tlv> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:28:45.804]
Connection received from 192.168.0.6 on port 50798 [04/03 13:28:45.824]
Read request for file <SEP0014A8924D6D.cnf.xml>. Mode octet [04/03 13:28:45.824]
File <SEP0014A8924D6D.cnf.xml> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:28:45.824]
Connection received from 192.168.0.6 on port 50799 [04/03 13:28:45.853]
Read request for file <SIP0014A8924D6D.cnf>. Mode octet [04/03 13:28:45.853]
File <SIP0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:28:45.853]
Connection received from 192.168.0.6 on port 50800 [04/03 13:28:45.874]
Read request for file <MGC0014A8924D6D.cnf>. Mode octet [04/03 13:28:45.876]
File <MGC0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:28:45.876]
Connection received from 192.168.0.6 on port 50801 [04/03 13:28:45.895]
Read request for file <XMLDefault.cnf.xml>. Mode octet [04/03 13:28:45.895]
Using local port 64147 [04/03 13:28:45.895]
<XMLDefault.cnf.xml>: sent 3 blks, 1077 bytes in 0 s. 0 blk resent [04/03 13:28:45.898]
Connection received from 192.168.0.6 on port 50802 [04/03 13:28:45.935]
Read request for file <P0S3-8-12-00.loads>. Mode octet [04/03 13:28:45.936]
Using local port 64148 [04/03 13:28:45.936]
<P0S3-8-12-00.loads>: sent 1 blk, 458 bytes in 0 s. 0 blk resent [04/03 13:28:45.937]
Rcvd DHCP Rqst Msg for IP 0.0.0.0, Mac 00:14:A8:92:4D:6D [04/03 13:30:28.597]
Previously allocated address 192.168.0.6 acked [04/03 13:30:28.597]
Connection received from 192.168.0.6 on port 50787 [04/03 13:30:28.636]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:30:28.636]
Using local port 64149 [04/03 13:30:28.636]
Connection received from 192.168.0.6 on port 50787 [04/03 13:30:29.627]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:30:29.628]
Using local port 64150 [04/03 13:30:29.628]
Connection received from 192.168.0.6 on port 50787 [04/03 13:30:33.627]
Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:30:33.628]
Using local port 64151 [04/03 13:30:33.628]
<SIPDefault.cnf>: sent 13 blks, 6203 bytes in 0 s. 0 blk resent [04/03 13:30:33.658]
Connection received from 192.168.0.6 on port 50788 [04/03 13:30:33.856]
Read request for file <SIP0014A8924D6D.cnf>. Mode octet [04/03 13:30:33.856]
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Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:30:41.756]
TIMEOUT waiting for Ack block #1 [04/03 13:30:43.637]
TIMEOUT waiting for Ack block #1 [04/03 13:30:44.629]
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:30:44.759]
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:32:11.253]
Rcvd DHCP inform Msg for IP 192.168.0.9, Mac 02:70:7E:7F:09:01 [04/03 13:32:14.253]
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Previously allocated address 192.168.0.6 acked [04/03 13:33:50.656]
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Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:33:50.696]
Using local port 62124 [04/03 13:33:50.696]
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Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:33:51.686]
Using local port 62125 [04/03 13:33:51.686]
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Read request for file <SIPDefault.cnf>. Mode octet [04/03 13:33:55.685]
Using local port 62126 [04/03 13:33:55.685]
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Connection received from 192.168.0.6 on port 50796 [04/03 13:33:55.915]
Read request for file <SIP0014A8924D6D.cnf>. Mode octet [04/03 13:33:55.915]
File <SIP0014A8924D6D.cnf> : error 2 in system call CreateFile Das System kann die angegebene Datei nicht finden. [04/03 13:33:55.915]
TIMEOUT waiting for Ack block #1 [04/03 13:34:05.698]
TIMEOUT waiting for Ack block #1 [04/03 13:34:06.691]
Is there anybody, who can help me with this problem. -
BEFSR41 fed from SPA2102 VoIP Phone Adapter with Router
urrently, a broadband cable modem is connected to the input of a new BEFSR41 4-port router. The router feeds 4 computers (a newer "primary" PC, and 3 secondary "old goats" that I use to print online grocery coupons). The 4 PC's are networked to share a common printer, a single directory on the "primary" PC, and using a 4-port KVM switch - I switch a single keyboard and mouse to whichever PC that I am using at the time.
I want to purchase a Linksys SPA2102 VoIP Phone Adapter with Router and integrate it into my existing system (noted above). This is probably a no-brainier for some, but I am new to routers and VoIP and that is why I am posting here, before buying the SPA2102.
According to the SPA2102 instruction manual that I downloaded from the Cisco website, there are two RJ45 connectors, one YELLOW and the other BLUE with the following typical connection configurations:
YELLOW . . . Connect one end of an Ethernet network cable (included) to the ETHERNET port of the Phone Adapter. Connect the other end to the Ethernet port of your PC.
BLUE . . . Connect one end of a different Ethernet network cable to the INTERNET port of the Phone Adapter. Connect the other end to your cable/DSL modem.
What I would like to do is, hopefully, simple and effective. I want to connect my cable modem to the SPA2102's BLUE connector, and from its YELLOW connector connect to the BEFSR41's input. Will this work, ensuring both the BEFSR41 and SPA2102 function properly?
Thank you.
CurlySueif you have access to the device and it is not remotely provisioned by the VOIP company. try to change the registration expires to a 24 hour time period as well as well as the Resync Periodic: in the provisioning tab to 24 hours as well.
-
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,
DanielTry 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.
-
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.
-
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
If you find this post helpful and are logged into the web interface,
please show your appreciation and click on the star below. Thanks. -
Using Cisco wireless IP phones in Cisco Call Manager 5.1
Hello,
Currently we are using Cisco CallManager ver.5.1.2.3000-2, Berbee InformaCast ver.6.1.1 and Cisco Unity 1.1 Build 10. We like to replace few Cisco IP Phone 7961G-GE with Cisco Unified Wireless IP Phone 7920 ver.3.0. Does CCM ver. 5.1.2 supports Wireless phones? If it does, how do I configure it? Also, do I have to purchase any new license or other hardware?
Any suggestion / recommendations greatly appreciate.
AlexYes, it does:
Cisco Unified IP Phone Series 7900
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/compat/ccmcompmatr.html#wp51474
Config on CUCM itself is no different from any other phones neither other requirements like DLUs.
Of course you should already have APs for connectivity.
HTH
java
if this helps, please rate -
Cisco 7821 IP Phone not registered in CUCM 9.1.2
Dear all,
We have CUCM 9.1.2.11900-12 and we have uploaded this device package: cmterm-devicepack9.1.2.12028-1.cop.sgn and reboot cluster
But the Cisco 7821 IP phone not registered in CUCM
can some one advise me?
Thanks and best regards,
TanHi
Can you please share the output for show version active from CUCM.
Please check the following thread as well:
https://supportforums.cisco.com/discussion/12030206/adding-7800-phones-cucm-8691
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We are experiencing issues when editing users & computers on a newly installed Server 2012 R2. The MMC randomly crashes when creating a new user. Sometimes we can enter 10-20 users flawlessly and sometimes it crashes every 2-3 users. In the logs we c
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Can we change budget status through project setup system status window?
Hi Experts, Can we change budget status through project statuses window? Thanks
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Hi gurur's how we give the validations to the table
hi gurur's how we give the validations to the table. plz help me