Voip accounting

Dear all
I am sending my cdr cisco voice gateway to my ACS.
but information are .csv based on i can not read them or have graph.
any idea or hint please?

Hi
extraxi aaa-reports! has some canned reports that are specifically for voip - call quality, billing etc.
This requires tallying up the various CDRs for each call leg and so is quite laborious to do manually!

Similar Messages

Need help in setting up PSTN to VOIP gateway with only one VOIP account.

Hi!
I am having hard time configuring SPA 3102 for # 3 and # 4 in following requirement list. I have only one VOIP account and my provider does not support simultaneous registrations even on different port. So I want to setup gateway without PSTN line being registered, so only Line 1 is registered with a VSP.
1. POT terminal can dial out using PSTN or VOIP based on dialplan. Done!
2. POT terminal can receive call from VOIP. Done!
3. If a call is received at PSTN line then it should be dealt as gateway call if from particular number, otherwise it is sent to POT Terminal.
4. If the call is not attended thru POT Terminal then it should be redirected to a particular number using VOIP after certain number of rings.
Regards,
Babar.

#3
i think that it is not possible,
- the default bahaviour of the SPA3102 is to recieve the call from the PSTN line, forward it to the phone connected to the FXS port.
- if there is no answer on the FXS it is then forwarded to a certain number via voip or given a second dial tone.
#4
This is functioning ans is the default scenario.,
register the PSTN line the same as the Line 1.
set the register to " NO"
make call without register to "NO"
ans call without regsiter to "NO"
- if you need to forward that to a voip number of your choice
set the PSTN To VOIP dial plan to 2 and change the dial plan on the PSTN line to S0 < : target phone number >

Sending an INVITE to my VoIP account by a SIP Servlet

I tried to make an outbound call to an ordinary landline number through my VoIP account by a SIP Servlet, but never been successful.
I tried something similar to the following:
javax.servlet.sip.SipFactory factory=(javax.servlet.sip.SipFactory) this.getServletContext().getAttribute(javax.servlet.sip.SipServlet.SIP_FACTORY);
javax.servlet.sip.SipApplicationSession appSession =factory.createApplicationSession();
to = “sip:land_line_number@my_voip_proxy_server.com”
from=”sip:my_voip_did_number:my_voip_password@ my_voip_proxy_server.com”
javax.servlet.sip.SipServletRequest invite = factory.createRequest(appSession, "INVITE", from, to);
invite.pushRoute(URI of my_voip_outbound_proxy_server.com);
invite.send();
I use Weblogic SIP server. I configured sip.xml, and everything else is correct to my knowledge.
I always get the error message something similar to “Cannot resolve path to my_voip_proxy_server.com”
(Note: When I use SJPhone softphone with my VoIP account details, it simply rings to a landline)
Is the way I incorporating my_voip_outbound_proxy_server correct?
Any help from anyone is greatly appreciated.
Thank you very much.

hi there,
I am at the same point too..
I am using a softphone called mirrial , Using Bea WEB Logic SIP Server...Deployed sar arhieve , did not define any outbound proxy in client side , did not specify default servlet name for sip server..
Is there anybody who succedded such kind of proxy calls ?
And can clarify the details if have?
Thanks

Asha 503 voip accounts not registering on 3G & not...

Hi all.
I had some trouble to find the proper Forum area for this issue. I hope this is the right place.
Here is the description of the issue:
    [1]Before inserting any SIM into Asha 503:
    [1.1]Through Wi-Fi, mobile registers and makes voip calls, both using encryption and not using encryption;
    [2]After inserting Operator SIM Card (Vodafone)
    [2.1]Accounts (both using and not using encryption) do not register via 3G (even though the sim card allows that - I use same card in other devices);
    [2.2]WiFi connection doesn't register encrypted accounts any more. Tried removing SIM card and restart phone but looks like the "feature" got permanently damaged
I did also try to reinstall firmware but issues remain (meaning that something is being left there from Operator SIM). Anyhow, I should be able to insert a SIM in the phone, otherwise it is not a phone right?
Further info: Previously I have been testing an Asha 311:
    [1]Registers and calls through 3G and WiFi (both using or not using encryption);
    [2] Has an issue with Encrypted sip accounts- when looses WiFi and comes back to same WiFi network, is not able to re-register (this problem doesn't affect sip accounts which are not encrypted).
    [2.1] This problem is not related to SIM - After reinstaling firmware, the symptoms remain.
NOTE: I quit from trying to solve this problem - that is why I bought the 503 .
If anybody as a clue how to help solving Asha's 503 issue, I would be truly grateful and happy.
Dinis
PS In the past I came accross a problem in Asha 311 that was returning error when some apps were trying to access to internet and the solution was this:
http://developer.nokia.com/Community...ries_40_phones
I am not sure if this would be the same kind of problem occurring now in the Asha 503 (however, Asha503 doesn't seem to have same kind of settings as in Ash 311)

Out of the blue after a few restarts, Asha 503 started to work with no problems when connection is WiFi (both encrypted and not encrypted accounts)*.
When connection is 3G it doesn't work at all (it is the same if account was created via UI or though provisioning).
So, the updated Status is:
Asha 503 - voip over 3G not working
Accounts (both using and not using encryption) do not register via 3G (even though the sim card allows that - I use same card in other devices);
Experienced only once - after testing several times turning on and off WiFi to force 3G, did not register via WiFi any of the accounts - had to restart the device. But this happened only once. Normally there are no issues registering through WiFi.
Asha 311j - issues reregistering encrypted accounts when mobile leaves and comes back to WiFi
Registers and calls through 3G and WiFi (both using or not using encryption);
Has an issue with Encrypted sip accounts- when looses WiFi and comes back to same WiFi network, it is not able to re-register (this problem doesn't affect sip accounts which are not encrypted).
This problem is not related to SIM - After reinstaling firmware, and not inserting SIM card, the symptoms remain.
*the unique difference from this morning was that I had removed and inserted again the SIM card - after I had changed some mobile data settings via provisioning. but in the morning was not working.
If anyone has any idea on how to properly solve this, would help me loads.
Cheers all

Asha 501: How to setup voip account for nimbuzz..

Hi,
Is nimbuzz voip activated services for asha 501.
If so, please let me know how to activate it. Also, do I need to pay it.

Hello, vsai.
Have you gone through this?: http://discussions.nokia.com/t5/Pool-of-Knowledge/Series40-phones-and-VoIP/td-p/1146267.
Let us know the outcome.

Implement Direct Inward System Access (DISA) in VoIP Environment

Hi,
May i know, is it possible to implement DISA Call in VoIP environment. If yes, how we can make it? Is it some configuration in CE Router at SRST Sites or CE Router at Main Sites? Also can you give me the information how to implement it?
As I understand DISA (Direct Inward System Access) allows someone calling in from outside the telephone switch (PBX) to obtain an "internal" system dialtone and dial calls as if from one of the extensions attached to the telephone switch. Frequently the user calls a number DISA number with invokes the DISA application. The DISA application in turn requires the user to enter his passcode, followed by the pound sign (#). If the passcode is correct, the user will hear dialtone on which a call may be placed.
Please advise me as soonest.
Thanks in advanced
Rgds,
Izazi Zainy

Giving users access to system dial tone via DISA is a security hole on PBX's and VOIP system so be careful how you use it. The following note describes how to use a TCL script and audio prompts to allow a user to call in and authenticate via an account number and PIN before they can dial an internal number. This will allow basic DISA type functions on a H323 gateway. Obviously you would also want to log the details of who made the call and when they made it, so syslog VOIP accounting is enabled to send a CDR to a syslog server.
We use an inbuilt TCL script that is inbuilt in IOS called 'clid_authen_collect'. This script authenticates the call with the ANI (Calling number) and DNIS (Called number) of the incoming call, or if this fails, it then prompts the user to enter an account number and then a PIN number. Since the call is coming in on an FXO (or FXS) port, there is no associated ANI and DNIS, so the script immediately prompts the user for the account number and PIN. We do the authentication by a local 'username XXX password YYY' command in the router config. The user keys in the account code and PIN (can use the # as a string terminator to speed the process up and if the values entered match a local username and password, it then prompts for the user to enter the actual destination telephone number.
I have also enabled syslog accounting for call detail records, so when the call completes you get a basic record of the called number and durations. If they wanted to use a full blown AAA server, they could run the authentication from this and this way keep a full log of all users calling in, and it would also log the CDR's for billing etc ...
The router needs the following audio .AU files on the flash memory :
Test#sh flash
System flash directory:
File Length Name/status
1 14097360 c2600-is-mz.122-11.T.bin
2 14150 enter_account.au
3 14869 auth_fail_retry.au
4 11510 enter_pin.au
5 52644 enter_destination.au
[14190860 bytes used, 2062068 available, 16252928 total]
16384K bytes of processor board System flash (Read/Write)
Test#
(obviously needs the IOS image but the important files are the audio prompts)
The .au files are the audio prompts that the IVR plays. These are in Sun/Next audio 64Kbps G711ulaw audio format. Use an audio editor to create the files and save them in this format.
When a call comes in on FXO port 1/0/0, you will hear a prompt to enter the account code. Key in the account number, followed by a #, then key in the PIN , followed by #. The caller will be prompted to enter the destination phone number, and this is matched on any subsequent voip or pots dial peers.
Configured user account numbers/passwords are 1000/1000 and 1001/1001
Refer to the attachment for the full router configs. Hope this helps.

Lumia 920 voip setting

can i use a voip account from braintel on my nokia 920

can i use a voip account from braintel on my nokia 920
i have a pte #************ ( sip:083.138.185.146 )
pls send me sip sitting ( *******@yahoo.com )
thanks 2 U
MODERATOR'S NOTE: This post has been edited. We would like to inform you that we have removed your personal details from your post as it is unwise to publish it publicly on this forum.

Ovi Chat - VoIP Where?

Hey everyone,
I couldn't find anything on this so I thought I should ask. So I have "AIM" as a VoIP account and all of my buddies import into my contacts. Now lets say I signed up for a @ovi.com account at www.ovi.com (which I did) and I used that same login to - log into the VoIp account in my n900.... it logs me in successfully and says I am available...
NOW WHAT? Where do I add buddies to it? I read somewhere ovi chat is really ovi contacts.. So at www.ovi.com I added them there under contacts but the phone does not "pull" the contacts into my phone contacts. Am I doing something wrong? Ovi Chat is like non existent. Is there a PC app? is there anything that I "log onto" to add contacts so they can show up in my phone contacts like my AIM buddies do?
Any answers would be most appreciative. Thanks!
Solved!
Go to Solution.

If you open "contacts" on your phone, edit a contact (or create a new one) you can add an Ovi address.. that's it.. Now you can send them IM messages..
contacts is phone AND IM contacts rolled into one

Missing aaa accounting commands

Hi,
I might be being REALLY STUPID, but I am trying to config a 12.3 IOS router to send command accounting records to an ACS 3.3 server via RADIUS.
When a input the 'aaa accounting commands 15 default group radius' command, it is accepted by the router, but show the config, and its not there. This is the same for all command levels. This router is logging VoIP accounting records too, to the same RADIUS box, without problems.
Have I missed somthing about setting up AAA ?
Grateful for any help!
Thanks
Pete Moore

Even if IOS did support it, the format of any RADIUS cmd accounting will be inferior for a couple of reasons
1) The ACS TACACS+ reports are totally geared up for this with pre-defined columns for each T+ attrbute.
2) ACS has a dedicated cmd accounting report which splits out cmds from sessions
3) To package in RADIUS, IOS would have to create many cisco-av-pair VSA instances. In the RADIUS accounting logs these will all be compressed into a single column of the format
"attr1=value1;attr2=value2;..."
Depending on what you want to do with the data this format is quite restrictive.
My advice is to enable TACACS+
Darran

How can I use the N8 with more then one VOIP-Accou...

Hello,
on my older Nokia-phones (N82 and E52) there wasn't a problem to use more then 2 different VOIP-Accounts. Now I have successfully create one VOIP-Account on my N8. The target should be to create an another (different) VOIP-Account and switch to this, if it's necessary. The second VOIP-Account should be a backup-solution ;-)
Thanks

OK, so did you try creating a second VOIP account on the N8? What steps did you try, and what were the results? If you post specific details, you will get better answers.
Lumia 920, Lumia 800
Nokia N8-00 (NAM, Product Code: 059C8T6), Symbian Belle, Type RM-596, 111.030.0609
Nokia 5800 XpressMusic (NAM, Product Code: 0577454) Software v51.2.007, Type RM-428

Linksys WRP400 - VoIP Sample configuration guide for VoiceMeUp

Thanks to Linksys for that fine device. Few installations so far with this device and it has prooven his stability and reliability.
Note: Good T38 support and excellent QoS.
Here is a sample of configuration to VoiceMeUp
Linksys WRP400 - Sample configuration guide
[english] - [français]
Enjoy!
For more informations about VoiceMeUp VoIP services, please visit us at voicemeup.com
Message Edited by NothinElse on 08-25-2008 01:09 PM

Clearly this is too late to help you, but since this is showing up on Google and there are about a thousand views, this answer might be helpful to another.
Once you access the Voice Admin menu and provision your device for your VoIP provider, access to the Voice menu is subsequently "Locked".
Your VoiP provider's system has set a new unique password for the Voice menu once the provisioned device has synced with your providers system.
You should be able to get that password from the control panel on you VoIP account profile under "equipment" or a call to support or an e-mail request should garner the password.
The Login will be "admin" and the password provided will get you back into Voice menu system.
You will also need this password if you want to reset the Voice features back to factory defaults. Because of this system it appears that a WRP400 cannot be unlocked by merely using the reset button.

GK supports Accounting ?

Hi All,
Our company is the ITSP. Currently we are planning provide long distance calls using IP Phones like ATA 186. We have a 3640 router with GK feature. But we don't have the voice gateways. My questions are follows.
1.) The gatekeeper supports Accounting or Billing using radius ?
2.) Can we provide prepaid solution for ATA 186 ?
Thanks in advance.

About the first question is important to configure accounting comand under gatekeeper configuration
aaa new-model
aaa authentication login default group radius local
aaa authentication login console none
aaa authorization exec default group radius local
aaa accounting connection default start-stop group radius
aaa accounting connection h323 start-stop group radius
aaa session-id common
radius-server host 10.255.20.18 auth-port 1645 acct-port 1646
radius-server retransmit 3
radius-server key CEdiS
radius-server authorization permit missing Service-Type
gatekeeper
accounting
It's all
Obviously you must configure an ACS software to receive a voip accounting.
Bye

C3-01 WiFi / WLAN & VoIP / SIP Question

Hello,
I am very interested in purchasing a C3-01, as I don't want a smart phone, but I like the advanced features this phone offers. Specifically, I want to make use of the VoIP SIP client that comes built in.
My question is this: I will be using VoIP / SIP on the WLAN (Wi-Fi) connection. In another post on this forum, I saw a user saying that Wi-Fi gets turned off automatically when the phone is locked.
So, does this mean that if I've got a VoIP account set up, and the C3-01 locks (either manually or automatically), that the WLAN will drop out and I won't be able to get incoming VoIP calls?
If this is how it works, this is a pretty major oversight, and I will look for something else.
Thanks
Solved!
Go to Solution.

Yeah, Sadly Some Country still have the v05.68 as the latest firware, But the reason might be that the Firmware released in your Country may contain the fixes included in the Latest Firware for Our Country.
So thats ok, If Nokia Finds any fixes to be done then, they will release a new Firware Update for your Country. Be Rest Assured.
And Buy C3-01.
If I've helped in any way, a click upon the White star to the left would always be appreciated.
If however my answer also solves your problem clicking below " ACCEPT AS SOLUTION " it will benefit other users!

SPA3102 pstn-voip dial tone without entering pin

Hello,
Once spa3102 is authenticated by pin, the adapter gives dial tone without entering pin (just the # sign after the beep) even when a different phone is used to call the spa3102 pstn number.
Isn't this a security compromise? Any one can access my voip account if the pin does not work.
Any help will be greatly appreciated.

that's weird...if PIN Authentication is enabled, it prompts the caller to enter the PIN number followed by a # key
how long is your PIN? i have the same set-up in my house and authentication works fine and i tried pressing # after the beeps but it didn't give me a dial tone....
| isolate! isolate! isolate! |

Broadband Talk and VoIP (again)

Hi - I have several VoIP accounts which I access from my iPhone including my BBT number. For years the BBT always worked (more reliably than the HH2 which regularly decides not ring for incoming calls until I do a reset) until many months ago (12 - 15?). I can receive incoming calls but when I try to set this up as the outgoing VoIP it fails when registering. All other accounts work perfectly.
I was out and about this a.m. and on coming home I needed to use VoIP. I noticed that BBT had now registered successfully and all the other accounts were failing. The reason for this turned out to be that the iPhone was connected to my HH2 BTWIFI-with-FON wifi (as it does when out and about) rather than my standard wifi network.
Sooooo can anyone give me a really good techie reason for this, why all but BBT works for normal wifi and only BBT works with FON wifi? Is there going to be any way of having them all work together again? I think this may not be for India.
Ta, Andy.

BBT relies on a number of servers, for example, here is part of the list from the home hub configuration file.
add name=BBT_servers type=ip addr=81.144.106.40
add name=BBT_servers type=ip addr=81.144.106.228
add name=BBT_servers type=ip addr=81.144.106.8
add name=BBT_servers type=ip addr=81.144.106.36
add name=BBT_servers type=ip addr=81.144.106.4
add name=BBT_servers type=ip addr=62.239.169.132
add name=BBT_servers type=ip addr=62.239.169.136
add name=BBT_servers type=ip addr=62.239.169.140
add name=BBT_servers type=ip addr=62.239.169.144
add name=BBT_servers type=ip addr=62.239.169.148
These are used when you are connected via the normal path, using your own SSID.
The number of servers are slowly being reduced, and its not unusual to find one which does not function correctly.
I have BBT, and sometimes I have to restart the home hub, as I cannot get dial-tone, and sometimes even incoming calls fail.
When you use BT Wifi/FON, your connection is passed from the home hub, to the BT Wifi network for authentication, over a VLAN. Its quite possible that the actual BBT servers that are used at the remote end, are in a different range, so its quite possible for a BBT call to fail on your normal SSID, but work fine on the BT Wifi connection.
Most questions on this user to user forum, are non-technical, and can be answered quicky. Bear in mind that this community forum has been going for quite a while now, so my posting count will be quite high, as are some of the other founder members.
I spent 42 years with GPO, Post Office, BT, and worked on many systems including Telecom Gold. Prestel, Packet Switch (X25), dial-up Internet, SMDS, SDH, ATM, and Business data systems.
The reply I have given you is the best you are likely to get, unless you can contact the people who are still actually maintaining the BBT network. I suspect there are not many now, withe the pending closure of the service.
There are some useful help pages here, for BT Broadband customers only, on my personal website.
BT Broadband customers - help with broadband, WiFi, networking, e-mail and phones.

Voip accounting

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