SPA-112 check SIP registration with SNMP

Which OID do I need to use to check the SIP registration with SNMP?

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Linksys SPA9000 and SIP registration with an E61

HI, I have my own IP telephony but I want to register my phone to be used with my Linksys SPA9000, anybody has the setup to do this with an E61?

Unfortunately my Nokia, an N95, can't register with the SPA9000 either. I managed to 'partially' register the phone. The phone registered itself an i saw it in the pbx status page.
(http:// - ip-adres of SPA9000 - /admin/status). The phone itself keeps notifying that it couldn't establish a connection and won't connect to the PBX.
What did the trick is realising that the PBX proxy listens to port 6060. So i added that port in the phone for the proxy and registrar. I know it's a part of the solution and i am working on it.
My plan
- setting up a syslog service on my linux server and search the logs for error mesages during the registration proces
- upgrading my firmware. The SPA9000 is up-to-date, but my N95 doesn't have the latest version yet. So i want to check the release notes from Nokia for information about solved SIP errors.
- checking the forums for tips
- ask Nokia for help

How to check CPU % Utilization with SNMP

We are using Ipswitch's What's Up Professional 2006 to monitor devices on our network. Their default snmp graphing utilities use the HOST RESOURCES MIB to collect and graph system performance statistics over time. I have them running perfectly on all my Windows servers, but for my Solaris systems I am only able to retrieve Memory, Disk, and Network Interface statistics. The CPU monitor is unable to retrieve data from our Solaris systems. A typical system is running Solaris 10 with the default agent, NET-SNMP version 5.0.x. Does anyone know if the default agent supports checking CPU usage in the HOST RESOURCES MIB? If not can anyone point me to a different MIB & instance that will return % CPU utilization of a Solaris 10 server? Any input would be greatly appreciated. I've searched everywhere for the solution but I am unable to find any straight-forward answers, hopefully someone can help.

Hi
You can do it via T Code ST02.I am sending you a link hope it help you
<a href="http://help.sap.com/saphelp_erp2004/helpdata/en/02/96263e538111d1891b0000e8322f96/content.htm">Check this</a>
Rewards point if helpful
Thanks
Pankaj Kumar

Multiple registration with sip-ua

Hi,
someone know a way to do multiple registration with a single 2811 using sip-ua configuration with multiple accounts??
thnx
s.

Hi, I have got to authenticate more than one account in the SIP provider with the hidden command "credentials" the problem that I have now is how to route all the calls done to the second account to the extension 101.
I want that incoming calls from 964812530 goes to extension 100 and incoming calls from 965072519 goes to extension 101
How can I do it?
I have tried this but it's not running:
sip-ua
credentials username 965072510 password 115849534F43415C557B7967 realm beta.awa
voz.com
authentication username 964812539 password 13544744535D4E7A7A757A70
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 5
timers register 250
registrar ipv4:213.162.201.146 expires 60
sip-server ipv4:213.162.201.146
voice service voip
sip
voice translation-rule 1
rule 1 /1../ /964812539/
voice translation-rule 3
rule 1 /964812539/ /100/
voice translation-rule 4
rule 1 /965072510/ /101/
voice translation-profile SIPout
translate calling 1
voice translation-profile incoming
translate called 3
voice translation-profile incoming2
translate called 4
voip-incoming translation-profile incoming
All the incoming calls are going to extension 100
regards

SPA 112 PROBLEM WITH SIMULTANEOUS CALLS

HELLO,I HAVE A PROBLEM IN THE ATA SPA 112 ABOUT SIMLUTANEOUS CALLS. The ata has 2 lines,when a call falls in line 1, the call on line 2 falls at the same time...and when I'm on line 1 and I recevive a call on line 2 the call in line 1 becomes mute and then drops.I didn' have these problems with the old pap2t .
Can you please help me to solve this problem ? Do I have to change something in the configuration ? Thanks,Mario

Hi,
thanks for responding.
I had found out about the timing problem in the meantime, but did not find a way to mark this thread as solved.
There are two timing values: one is the PSTN answer delay, and the other one the PSTN ring timeout.
It seems that ring timeout should be longer than ring time + ring pause. As long as this conndition is not met. the dial plan is not even considered

SPA 112 - Faxing with ATA gateway

I had a regular phone line but the only purpose for having that line was for sending faxes, which only happens a few times per month. So, naturally, I felt it was a waste to be paying so much money for it.
I purchased the SPA 112 and configured it to use with VoiceNetwork.ca. I was successfully able to send faxes.
I later cancelled my phone number and switched my internet to cable (since I was on DSL before).
I can no longer successfully complete a fax ever since I switched to cable internet. I don't know why because the speed is 3 times faster upload and 5 times faster download.
I'm guessing there is a setting that needs to be changed in the ATA admin panel to adjust for cable internet.
My configuration settings are attached as screenshots.
Firmware: 1.3.2-XU (014) Jul 2 2013

I had a regular phone line but the only purpose for having that line was for sending faxes, which only happens a few times per month. So, naturally, I felt it was a waste to be paying so much money for it.
I purchased the SPA 112 and configured it to use with VoiceNetwork.ca. I was successfully able to send faxes.
I later cancelled my phone number and switched my internet to cable (since I was on DSL before).
I can no longer successfully complete a fax ever since I switched to cable internet. I don't know why because the speed is 3 times faster upload and 5 times faster download.
I'm guessing there is a setting that needs to be changed in the ATA admin panel to adjust for cable internet.
My configuration settings are attached as screenshots.
Firmware: 1.3.2-XU (014) Jul 2 2013

Updating spa 112 with firmware 1.0.2 (006)

I recently purchased a spa 112 and configured for the VoIP offered by my ISP. It work more or less fine but I would like to update to the latest firmware to have access to call display and other options.
The update process mentioned specifying a path where the new firmware is saved in the Phone Adapter Configuration Utility under the administration "tab". In my case, all that shows up on the left hand side is the quick setup option. This is true regardless of the actual tab selected.
It looks like I have an older hardware (1.0.0) and firmware 1.0.2 (006) version. Could someone direct me to the procedure to update to a newer firmware?
Michel

Hey,
I had exactly the same problem. My original firmware was as well 1.0.2 and I wanted to upgrade to 1.3.3. However changing the browser didn't work (IE 8, IE 11, Firefox, Chromium, Safari).
Instead I used the following direct upgrade link:
http:///Upgrade_run.asp;session_id=
where has to be replaced by the generated ID after login (see address bar).
Example:
http://192.168.1.132/Upgrade_run.asp;session_id=40c5188219bf4ea4beb4952003c13365
Marc

SIP Registration Request not reaching provider

I have a home lab setup and recently decided to add PSTN access via a SIP trunk and termination. I got the applicable information from the provider, set up the IOS configuration, but I'm not registering. I checked the CCSIP messages logs, no response. I talked to the provider, they are not getting my messages at all. Other traffic to and from the internet passes just fine (including SSL VPN).
I've talked to my ISP many times, and they continually insist they do not block any traffic except port 25.
Here is the applicable configuration:
sip
bind control source-interface Vlan1
bind media source-interface Vlan1
session transport tcp
header-passing
no call service stop
sip-ua
credentials username XXXXXXXXX password 7 XXXXXXXXXXX realm XXXX
authentication username XXXXXXXXXXX password 7 XXXXXXXXXXXX
registrar ipv4:XX.XX.XX.XXX expires 3600
sip-server ipv4:XX.XX.XX.XXX
(the source button puts the whole post in source tags... annoying)
And here's the debug output from ccsip all:
Apr 10 21:17:33.091: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone CDT to SIP default timezone = GMT
Apr 10 21:17:33.091: //594/000000000000/SIP/Info/sipSPISendRegister: Associated container=0x4F7E3670 to Register
Apr 10 21:17:33.091: //594/000000000000/SIP/Transport/sipSPISendRegister: Sending REGISTER to the transport layer
Apr 10 21:17:33.091: //594/000000000000/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
Apr 10 21:17:33.091: //594/000000000000/SIP/Transport/sipSPITransportSendMessage: msg=0x4F3BB008, addr=XX.XX.XX.XX, port=5060, sentBy_port=0, local_addr=10.XX.XX.XX, is_req=1, transport=1, switch=0, callBack=0x0
Apr 10 21:17:33.091: //594/000000000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
Apr 10 21:17:33.091: //594/000000000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
Apr 10 21:17:33.091: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:Xx.XX.XX.XX, rport:5060 with laddr:Xx.XX.XX.Xx
Apr 10 21:17:33.091: //594/000000000000/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x4F3BB008
Apr 10 21:17:33.091: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x4F3BB008, addr=XX.XX.XX.XX, port=5060, local_addr=10.Xx.XX.XX, connId=3 for UDP
I took out the message because I don't think it's relevant, and has too many IPs to XX out :)
Apr 10 21:17:37.091: //594/000000000000/SIP/Error/act_sent_register_wait_100: act_sent_register_wait_100: Out of retries
Apr 10 21:17:37.091: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDecrementOverloadCount: Count:Local 0 Global 1
Apr 10 21:17:37.091: //594/000000000000/SIP/Error/ccsip_api_register_result_ind: Message Code Class 4xx Method Code 100 received for REGISTER
Apr 10 21:17:37.091: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_register_reset_dns_cache: CCSIP_REGISTER:: registrar 0 DNS resolved addr reset
Apr 10 21:17:37.091: //594/000000000000/SIP/Error/ccsip_api_register_result_ind: SIP Registration Retries Exhausted
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/sipSPIRegPthruProcessResponse: Processing response w/ resp code == 408
Apr 10 21:17:37.091: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetRPCBFromRCB: Retreiving RCB [0x4FA37D28] from RPCB [0x0]
Apr 10 21:17:37.091: //594/000000000000/SIP/Error/sipSPIRegPthruProcessResponse: Error NO RPCB
Apr 10 21:17:37.091: //-1/xxxxxxxxxxxx/SIP/Info/ccsipRegisterStartRCBTimer: Starting timer for pattern XXXXXXXX for 180 seconds
Apr 10 21:17:37.091: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDeleteContextFromTable: Context for key=[576] removed.
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/sipSPIUdeleteCcbFromUACTable: ****Deleting from UAC table.
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x4B6CC660 key=AD7E2293-C01A11E3-80AEE452-E7A949D0
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/ccsip_offer_ans_delete:
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/ccsip_iwf_delete:
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 4B6CC660

Nevermind; I downloaded a SIP phone app for my smart phone, got it to connect in the same network. After that, I put the gateway in a DMZ and it started working. Not entirely sure why it was required, but hey, it works!
Cheers

Cisco SPA 112 outbound call issue.

Firstly, i applogize if i havent posted this in the correct section but this was my best educated guess.
Recently upgraded to a Cisco SPA 112 which works fine.. but only for a few days,
then If i try to call a number i get the dial tone but the number fails to
connect & all i hear is silence. If i hang up the phone to retry it fails to
disconnect correctly and i get no dial tone for several minutes. After leaving the phone
hungup for a few mins the dail tone returns but the same problem is still present.
A hard/soft reset of the 112 fixes the issue temporarily, but reoccures again withinn a few days.
Running Sipgate on lines 1 and 2 on Virgin media UK ISP.
Model:
SPA112, 2 FXS
Hardware Version:
1.0.0
Boot Version:
1.0.1 (Oct 6 2011 - 20:04:00)
Firmware Version:
1.3.1 (003) Dec 17 2012
Recovery Firmware:
1.0.2 (001)
Im no expert and now at a loss and could do with some expert help tbh.

Dan, i will see how it goes over the next week or so with the updated F/W & shall report back, and thanks for the link as i wasnt sure how to save a syslog.
Should i set the verb as high as 6, or is that an over kill.
Gabriel, I have an FTP running on my sat receiver which i will attempt to use to catch the log, but as for the the "sipgate registration", when i loose conection my ata reports its still registered & the sipgate URL, isnt in real time as far as i can tell.
My ata would have normally shown its fault since my first post 6 days ago, but unfortunately a recent power cut has reset my testing period back to 1 day.

Cisco SPA 112 adapter Issue

Hi All,
We are using Cisco SPA 112 adapter and it is having 2 lines,I configured both of the lines.
There is no issue in line 1 but line 2 is automatically went to disable mode.After that i have to manually enabled that option.
Please help me to fix this issue

Dan, i will see how it goes over the next week or so with the updated F/W & shall report back, and thanks for the link as i wasnt sure how to save a syslog.
Should i set the verb as high as 6, or is that an over kill.
Gabriel, I have an FTP running on my sat receiver which i will attempt to use to catch the log, but as for the the "sipgate registration", when i loose conection my ata reports its still registered & the sipgate URL, isnt in real time as far as i can tell.
My ata would have normally shown its fault since my first post 6 days ago, but unfortunately a recent power cut has reset my testing period back to 1 day.

Force/lock rtp ports in SPA 112/122.

Hi,
I hope someone can help me. I am looking for a setting inside the Cisco/Linksys spa 112/122 which can force the rtp source port to be the same on the sip provider - is that possible? When the spa 112/122 connects to the sip provider is the port number from the box is 16384(standard range 16384-16482 inside the spa) but on the sip provider will respons with eg. port 36741.
/Tom

Hi Bro
Before you proceed to add the line shown below, I'm guessing you're unable to access and PING 172.20.16.8 once you've successfully VPN in, am I right? If yes, which groupname and username did you use? Lastly, did you use IPSEC VPN Client or WebVPN?
access-list inside_nat0_outbound extended permit ip 10.20.60.0 255.255.255.0 172.20.16.0 255.255.255.0
Regards,
Ram

How to get the number of routes in MLS L3SW table with SNMP

Hello experts,
Is there any way to get the number of routes in the MLS-hardware Layer 3-switching table with SNMP, like with CLI command "show mls cef summary"?
If possible, please let me know the OID and which MIB do I use.
[e.g.]
#show mls cef summary
Total routes: 400000
<omit>
====
Device : C6509E
IOS : s72033-advipservicesk9_wan-mz.122-33.SXH3a.bin
====
Thank you,

function buttonClick()
var table = profileTable;
var lnRow = table.rows.length;
var insertedRow = table.insertRow(parseFloat(lnRow));
var cell1 = insertedRow.insertCell();
cell1.innerHTML ="<tr><td><Input type=\"hidden\" >>>name=\"rowNum\" value="+cnt"+></td></tr>";
document.profileform.submit;
on submit it goes to the second page, but the value i got using >>>System.out.println("row number from text >>>box"+request.getParameter("rowNum")); is null. What is wrong with >>>my coding. Can anyone solve this.HI carry
Check the value of bold data
function buttonClick()
var table = profileTable;
var lnRow = table.rows.length;
var insertedRow = table.insertRow(parseFloat(lnRow));var cnt=inRow
var cell1 = insertedRow.insertCell();
cell1.innerHTML ="<tr><td><Input type=\"hidden\" >>>name=\"rowNum\" value="+cnt+"></td></tr>";
document.profileform.submit;
}try with it

LMS 3.2 with SNMP v3 not working

Hi ,
My network is currently running with SNMP v2 configured in easch devices. With snmp v2 our LMS 3.2 server is working fine. However we have planned to migrate our network to snmp v3 . I have configured my few devices for SNMP v3 and added them to my LMS server.
Except DFM module these new SNMP v3 devices are working fine in all other modules. In DFM these devices are reflecting under "snmp timeout" group.
I checked with device center -> management station to device; where the SNMP v3 connections are showing "okey"
following are tyhe configuration i have done in my devices.
snmp-server group v3g v3 priv read testr write testw
snmp-server user v3u v3g v3 auth md5 test123
snmp-server view testr iso in
snmp-server view testw iso in
snmp-server host 10.X.X.38 version 3 priv v3u
snmp-server user v3u v3g v3 auth md5 test1234 priv des56 test4321
snmp-server group v3g v3 priv read testr write testw
snmp-server user v3u v3g v3 auth md5 test123
snmp-server view testr iso in
snmp-server view testw iso in
snmp-server host 10.X.X.38 version 3 priv v3u
snmp-server user v3u v3g v3 auth md5 test1234 priv des56 test4321
followinfg are my module details.
LMS      : 3.2
CM        : 5.2
CV          :6.1.9
CS          :3.3.0
DFM     : 3.2.0
IPM     : 4.2.0
RME     : 4.3.0

DFM behaves different than the other modules.
DES56 is not a supported privacy algorithm for DFM. You can use DES or AES128.
Supported Algorithms in DFM
The details of the algorithms supported in DFM are:
•AuthNoPriv Mode — Supported Auth Algorithm: MD5 and SHA
•AuthPriv Mode
–Supported Auth Algorithm: MD5 and SHA
–Supported Privacy Algorithm: DES and AES128
–Unsupported Privacy Algorithm: 3DES, AES192, and AES256
For more details check :
http://www.cisco.com/en/US/docs/net_mgmt/ciscoworks_device_fault_manager/3.2/user/guide/useDevMg.html#wp1483766
-Thanks
Vinod

LMS 4.0.1 and User tracking with SNMP v3

Hi! (again )
I've another problem with our new LMS 4.0.1.
We manage our devices with SNMP v3 but the user tracking don't want to work flawlessly.
I've attached an example from our SNMP configuration. Basicly it's the same in our devices.
1st the problem was that no matter what I did the User tracking didn't want to find any host. I left it and worked on something else. After 2 weeks suddenly appeard couple of thousand end host.
As earlier (LMS 2.6 or 3.2 with snmp v2) it is the same that LMS cannot differentiate normal end host and IP Phones although we have several thousand from both. But this is only one problem.
The other is that there are switches with the same IOS and SNMP configuration and from one I get the UT data and from another one I didn't get anything. Only from some 4506 (aprox. 12-15) and 6506 (2) works and we have 20+ 4506 and 10+ 6506. Not to mention the other switches (couple of houndred 2960 and 3750).
I'll be grateful if somebody could advice what to do.
Thanks
Gabor

Understanding Debugger Utility
The utility displays a report on the reasons why User Tracking failed to discover end hosts on specific ports.
In many cases, User Tracking may not perform as expected. This may be because of problems in other LMS applications. For instance LMS Server may have devices that are not discovered or inadequate VLAN discovery in Topology Services.
You can run the utility to troubleshoot problems, or provide the report and log generated by the utility when you contact TAC for help in diagnosing problems.
The debugger utility uses the data collected by LMS Server and reports the reasons for the missing ports in User Tracking.
This tool also has an SNMP component embedded which runs an SNMP query for the table as a part of verification for SNMP failure. For example, SNMP bugs in Catalyst operating system because of which User Tracking may fail to discover devices.
This generates an Action Report that you can use to analyze the data.
The Debugger Utility:
1. Checks the switch ports in a sequential order.
2. Reports violation of basic rules for each of the missing ports such as link ports and trunk ports.
3. Checks for SNMP retrieval of data, if the ports pass the validity check.
4. Generates an Action Report suggesting possible remedial actions to retrieve the valid missing ports.
Using Debugger Utility
The Debugger Utility is available at $NMSROOT/campus/bin/ (where $NMSROOT is the directory where you have installed CiscoWorks).
To run the Debugger Utility, run the command:
utdebug -switch switch-ip -port port1[,port2 ...] [-export filename]
where,
switch is the switch to which the end hosts are connected.
ports are the ports on the switch which have missing end hosts User Tracking.
-export filename specifies that the debug messages be stored in the file specified. If this option is not used, the messages are displayed on the console.
For example,
utdebug -switch 10.29.6.12 -port 5/12
utdebug -switch 10.29.100.10 -port Fa0/10
utdebug -switch 10.29.6.14 -port Gi6
Pretty sure you will find this and perhaps more in the build in help of LMS
Cheers,
Michel

2921 SIP connection with providers

Hii,
I have cucm10.5 and 2X 2921 routers and I want to now which is the best way to make a SIP Trunk with my providers in order to not use E1 lines anymore. It's more easy to make the SIP from Cucm/providers or directly from router/providers ???
Any ideas or proper step, thanks a lot for help:)

Hey Jaime,
Can you check if I already have the licence for the CUBE?
License Type Supported
permanent Non-expiring node locked license
extension Expiring node locked license
evaluation Expiring non node locked license
evalRightToUse Right to use evaluation non node locked license
rightToUse Right to use non node locked license
paid subscription Expiring node locked subscription license
with valid end date
extension subscription Expiring node locked subscription license
evaluation subscription Expiring node locked subscription license
License Operation Supported
install Install license
clear Clear license
annotate Comment license
save Save license
revoke Revoke license
call-home License call-home
Call-home Operation Supported
show pak Display license pak via call-home
install Install license via call-home
revoke Revoke license via call-home
resend Fetch license via call-home
Device status
Device Credential type: DEVICE
Device Credential Verification: PASS
Rehost Type: HARDWARE

SPA-112 check SIP registration with SNMP

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