VoIp Billing

I noticed few people were looking for a cisco / sip compatible billing solution. Check out www.cyneric.com reputable company with a great billing solution. Any questions, contact [email protected]

You are on the wrong place, dude. This is a technical forum.
You have to be kicked out of this forum for good.
And, it is stupid to name your own company "reputable... with a greal billing solution".
Have you seen the really reputable companies to promote themselves on the forums? The answer is - NO!
Let the customers tell how reputable is your company and how great is your billing solution.

Similar Messages

Call info from VOIP Router for billing purposes

Hi Pros,
My client has a VOIP network where all routers have FXS Cards & Dial Peers configured
--- No CallManager ---
They want to bill calls using router outputs.
Now, I thought of collecting debugging data on syslog servers. Can anybody help identify a debug command which can generate call setup (calling party & called party) & time info?
Pratik

Hi,
Check out this link.
http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a0080094e72.shtml
You can use syslog. You would need some clever parsing to generate the billing etc i think though.
Cheers,
Tim.

VoIP Design and Billing

I am in the process of designing a VoIP network between a country in Europe and the USA. Any suggestions on equipment, billing, calling cards, routing traffic, etcc...

Hi Wong
The Cisco technical documentation do not really give you detail info about all the details needed to get into the VoIP termination business. I was looking for contacts or companies that would route and send traffic to my VoIp network. In addition to what would be the smartest way and least cost to strat up this venture.

VOIP calls..how do they appear on my phone bill

i was looking to use fring or tango app to place a call..how does that appear on my wireless bill..is it as an itemized call or does it just show up as data usage? thanks so much

You shouldn't be using the same account on phone iphones. Have him setup his own and make sure his iCloud is logged in with that one

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.

Moving to Zendesk - VOIP advice needed.

Hi all,We're currently going through a major business change and we're moving from Freshdesk to Zendesk. Currently we use X-On as our VOIP provider, they've got the features we need but would like something that has tighter integration with Zendesk.I need the ability to:Create and modify MuHG'sAn easily configurable IVR for when people call in so they can be routed quicklyCall recordingThe ability to transfer a call out to an agent on their mobile, and would be great to have it carry on recordingI also need to be able to run detailed reports, for customer billing purposes and to make sure calls are getting through ok, etc.Zen voice seems great, but I can't find these options in the version we're purchasing.This is for a UK based business.We're not completely averse to moving to desk phones and away from the softphone approach, but the...
This topic first appeared in the Spiceworks Community

Hi all,We're currently going through a major business change and we're moving from Freshdesk to Zendesk. Currently we use X-On as our VOIP provider, they've got the features we need but would like something that has tighter integration with Zendesk.I need the ability to:Create and modify MuHG'sAn easily configurable IVR for when people call in so they can be routed quicklyCall recordingThe ability to transfer a call out to an agent on their mobile, and would be great to have it carry on recordingI also need to be able to run detailed reports, for customer billing purposes and to make sure calls are getting through ok, etc.Zen voice seems great, but I can't find these options in the version we're purchasing.This is for a UK based business.We're not completely averse to moving to desk phones and away from the softphone approach, but the...
This topic first appeared in the Spiceworks Community

VoIP Gk or Proxy Accounting?

Hi all.
My Customer is a long distance VoIP Service Provider, currently collecting calls from and delivering them to the PSTN using several Cisco 5350's. In this environment, billing is a "simple" matter - they collect syslog info from the PSTN gateways.
Now they want to interconnect VoIP carriers, and they're going to implement a Cisco Gk infrastructure with Cisco H.323 Proxies and RADIUS Accounting. Now generating billing info seems to be more involved.
I have a few questions:
1. Is a Cisco Proxy able to generate CDR's as detailed as a Cisco Voice Gateway?
2. Does a Cisco Gk know *exactly* when a call is connected and disconnected and would it be possible to centralize CDR generation on the Cisco GK?
Thank you very much!
michele

Hi Taimoor,
and thank you very much for your prompt response.
You propose me a different solution than the one I thought, and please let me check if I understand it right, so I can present it to the Customer.
--- My proposal: Get rid of all CDR's from the GW's and Proxies and let the GK (a *separate* Cisco router from the Gk) generate the CDR - I know this might not be the best choise as I don't know if the GK is able to generate detailed CDR in this environment.
--- Your proposal (as far as I understand it): use the CDR's generated by the GW's and the by Gk/Proxy - good because we already know how these CDR's are made and how to elaborate them
My question is: does your proposal still work if I use separate Cisco routers for the GK and the Proxy (which is the Customer's target)?
That is, would it be possible to have the Cisco Proxy generate detailed CDR's and bill in the following environment?
|-------- Cisco Voice GW
| (generates CDR's for outbound/outbound PSTN calls)
|
|
Cisco MCM GK----|
|
|
|-------- Cisco MCM Proxy
(generates CDR's or outbound/inbound VoIP calls)
Thank you
michele

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!

E63 and VOIP problem, pls help

Hi Guys
I got Nokia e63 and have successfully configured local VOIP number but it works when I use the " mobile internet" or " my service provider internet " as my default access point not my home wireless connection.
When I set my home wireless connection as default access point inside sip settings then I can make calls but cannot receive , I thought my netgear router or firewall causing problem , i configured e71 ( my friend's fone) and it work both ways without any problem.
when i change the access point to other than home wireless , no issues on e63 , incoming and outgoing work like a charm.
any known issues ? please advise bcoz its consumin my mobile internet
many thanks

I have a similar problem, I have my E63 for five months now, For first 2 months i was able to make voip calls no problem and was very happy.
I was using Mo-Call and Skype. First of all I noticed a degrading of my Mo-Call where I would have to dial several times to get through. then I was unable to connect at all. If I dial a number now I can sometimes hear a dial tone, the phone at the other end rings and picks up. I cant hear them and they cant hear me. annoying thing is that I get billed for the call by my voip service providers.
I soldiered on with skype which has now downgraded to the point that I can only communicate with instant messaging although it still rings me and I can ring out.
I have checked and my nokia software is up to date. needless to say I checked with several known good wi-fi points and no luck. I have had friends check thier smart phones on my wi-fi and all had no problems.
Mo-Call worked hard to try to identify the problem but my account with them works if I try it on other phones.
I find Nokia Customer Care website difficult to work with. I have been directed to thier local agent here in Dublin (Fonemenders) whos phone leads to an answering machine. a message was left and they do not respond.
Any help greatly appreciated.

X6 voip problem. Need to redial if receive an emai...

This problem has happened on X6 firmware 31.0.001 and brand new 32.0.002. It has also happened on skype and voipwise voip software. When on a voip call if you receive an email or sms you will lose audio on your call. Needing to hang up and dial again. It is an annoying problem as i use voip alot.

Hi GemmaMcD,
Welcome to the forum and thanks for posting. I’m sorry you’re having problems with the bills etc. I can take a look into this for you if you wish. Drop me an email with the details. You’ll find the “contact us” form in the about me section of my profile.
Cheers
David
BTCare Community Mod
If we have asked you to email us with your details, please make sure you are logged in to the forum, otherwise you will not be able to see our ‘Contact Us’ link within our profiles.
We are sorry but we are unable to deal with service/account queries via the private message(PM) function so please don't PM your account info, we need to deal with this via our email account :-)

Voip dtmf

Hello,
I'm trying to understand how dtmf is different from analog world to the VOIP world. In the analog world from my understanding dtmf are tones frequency so when caller dials numbers then does numbers have different tones frequency and it send to the telephone company to decode those tone frequencies and know where to route that call to the other person receiving the call. In the VOIP world with call manager there is in band or out band dtmf. But how does the process work when phone A calls phone B how does it know where to route the call. Like does dtmf turn into packets to tell cucm where to route the call and the user hears tones when pressing the numbers just to simulate the dtfm in analog world. What about dtmf in band with rtp packets does dtfm go first in rtp to let call be routed to the other phone or those the voice gateway or cisco switch see rtp dtmf and route to the correct phone?
Thanks,

Horacio,
What you are describing is call setup or call signaling. In particular, you are asking about how the client (IP Phone) communicates its intended destination to the server (CUCM).
In a VoIP environment, the client sends the dialed digits to the server using IP packets. Each call signaling protocol has a method for communicating this information in what is called an application header. The information does not rely on DTMF tones. The client will simply identify the digit(s) dialed in the appropriate header.
There are two general ways that a client, such as an IP phone, can present the dialed digits. One way is called "en bloc" or "all together". What this means is that the phone will send a packet to the server and the call setup request will provide all digits as a single "string". An example is your mobile phone. You can dial all of the digits of your intended party and then click "send" or "call". When you do this, your phone is sending all of the dialed digits to your carrier. This is en bloc dialing.
In CUCM, phones use en bloc dialing in several scenarios. Including redial and dialing from the corporate directory (or missed calls, received calls, etc.).
The second way a client can communicate dialed digits is digit-by-digit. CUCM supports digit-by-digit dialing with Cisco IP Phones that support SCCP and SIP. All this means, is as you dial a digit on your phone, that phone is sending a packet to the CUCM that identifies you dialed that single digit. The CUCM digit analysis process is collecting those digits, one-by-one. As soon as it determines there is a unique match, it will route the call.
None of what I described above leverages DTMF. Meaning, there are no tones exchanged. There is no need. The client just says "this dude dialed a 5", or whatever you dialed.
Now, you mentioned voice gateways. Well, a voice gateway that communicates with the CUCM is leveraging an IP protocol. In today's networks the protocol is usually SIP, H.323, or MGCP. Of course, Cisco gateways can also use SCCP, which is proprietary to Cisco. Regardless of the protocol used, the voice gateway sends digits to the CUCM and receives digits from the CUCM in a manner that is similar to IP phone clients. Which is to say, that they stick the destination information in a header, stick that header in a packet, and send it to the appropriate peer. No DTMF.
Of course gateways, by there very nature, connect two disparate systems. Gateways relay call setup information from one entity to another. For instance, let's say you have a T1 PRI connected to a voice gateway. The gateway has an IP connection to CUCM and an ISDN connection to the carrier (or whatever is on the other end of the PRI). It just so happens that the ISDN protocol also exchanges call setup information in a manner similar to IP protocols. Which is to say that the digits dialed by the calling party are exchanged in protocol messages NOT DTMF tones.
Up to this point, I have only focused on call setup because that seemed to be the premise of your question. I am not suggesting that DTMF isn't used in VoIP. It is. DTMF is used by applications such as voicemail systems, call centers, and other IVR-based systems. For example, let's say you call a given number and you hear a greeting which prompts you to press 1 to connect to Horacio and press 2 to connect to bill. This is an IVR system and how that system does its job is by interpreting the key presses using DTMF recognition.
This gets back to another thread you posted concerning "in band" and "out of band" (OOB) DTMF. "In band" simply means that the DTMF tones are packetized and sent in the RTP stream. They are digital samples of the analog tone, literally. The other end is responsible for understanding how to deal with that. If you and I were on a phone call and I kept hitting the number 5, you would hear it and you'd probably get bent because your ear isn't equipped to understand the DTMF representation of the digit "5".
Out of band (OOB) means that the sender is relaying the digits dialed via the call signaling protocol. It is similar to the whole digit-by-digit thing I described earlier but it is presented in a different message. This message gets relayed through your network, and any intermediary devices, and lands on the receiver's end. As long as every device in the call flow is using the same method to do the OOB signaling, you are golden. This part of the conversation can get long. Longer than it already has. I recommend researching "DTMF Relay" to start getting a better understanding.
HTH
-Bill
(b) http://ucguerrilla.com
(t) @ucguerrilla
Please remember to rate helpful responses and identify helpful or correct answers.

Radius and Billing

Dear NetPros,
I have configured the Radius & Billing Servers on my Cisco AS5350 which is terminating VoIP Traffic as given below. The First two are Mind Billing Primary and Secondary Billing Servers. The Third one is a billing server from another vendor. I want to send CDR information to all the three billing servers simultaneously. Currently the gateway is only sending the Radius and Billing information to the first available server. Is there any way for the gateway to send radius and billing information to all these three servers simultaneously???? Would appreciate any help or suggestion in this area. Thanx
aaa group server radius mind
server AAA.BBB.CCC.DDD auth-port 1645 acct-port 1646
server EEE.FFF.GGG.HHH auth-port 1645 acct-port 1646
server III.JJJ.KKK.LLL auth-port 1812 acct-port 1813
radius-server host AAA.BBB.CCC.DDD auth-port 1645 acct-port 1646 key 7 XXXXXXXXXXXXXXXXXXXX
radius-server host EEE.FFF.GGG.HHH auth-port 1645 acct-port 1646 key 7 YYYYYYYYYYYYYYYYYYYY
radius-server host III.JJJ.KKK.LLL auth-port 1812 acct-port 1813 key 7 ZZZZZZZZZZZZZZZZZZZZ
Cheers
Rushabh
Senior Project Researcher
PP-Ontime Co., Ltd.
Cellular ~ 669-2047331
www.pp-ontime.co.th

The AAA "Broadcast Accounting" feature allows accounting information to be sent to multiple AAA servers at the same time; that is, accounting information can be broadcast to one or more AAA servers simultaneously. This feature allows broadcasting among "groups of servers". And each server group can define its backup servers for fail over independently of other groups.
However, the restriction is that Accounting information can be sent simultaneously to a maximum of four AAA servers.
For the scenario mentioned, in order to send billing info to all the 3 servers simultaneously, the aaa accounting command can be configured globally, as in:
aaa accounting network default start-stop broadcast group mind1 group mind2 group mind3
The individual servers in the server group 'mind' may be split across different server groups.
aaa group server radius mind1
server AAA.BBB.CCC.DDD auth-port 1645 acct-port 1646
aaa group server radius mind2
server EEE.FFF.GGG.HHH auth-port 1645 acct-port 1646
aaa group server radius mind3
server III.JJJ.KKK.LLL auth-port 1812 acct-port 1813
(Backup servers within each server-group may be defined)
Simultaneously accounting records are sent to the first server in each group. If the first server is unavailable, fail over occurs using the backup servers defined within that group.

Distinctive ring vs. new line vs. adding VOIP service like Vonage or Ooma

I'm currently a FIOS/phone (double play) customer. I need an additional phone number and am considering my options (getting a new line put in, signing up for a new VOIP service like VOnage, going cheap & just getting distinctive ring), trying to understand everything and weigh the costs/functionality/gotchas.
"Distinctive Ring" is a new option I just learned about, and it seems the easiest, and definitely the cheapest (around $5 extra a month). However it also seems the most limited and there are a few things I don't understand for 100%.
If anyone can answer any of these questions I'd appreciate it:
- Can you setup SEPERATE voicemail greetings for the existing landline vs. the new "distinctive" ring number? Or would everyone calling hear the same greeting?
- If only same outgoing message can be used, then I assume that each number uses the same voicemailbox, but can you setup different mailboxes for each line in a different way, like having your outgoing message say "Press 1 to leave message for Herman or press 2 to leave message for Granpa", thus create at least quasi-seperate voicemailboxes based on who the person was trying to call? And if so, how?
- As I understand it not all phones support "distinctive ring" - so how can I findout which of my existing phones do support this (other than signing up for the service then calling myself and seeing how the phones behave)?
- Is there any way for my callerID boxes to show which # it is that is being called instead of just who is calling?
- And is this true, that calling out I would only be placing calls from my original line, that for example on callerid of who I am calling, even if pressing callback on my phone and they called my new second #, they would see my original #, not the new one?
Thanks for any hints.

Hi, the only tip I can give you is as far as testing if your phone is compatible with distinctive ring you could try one of two things, neither of which are free unfortunately.
1) Call a number that is busy and then use Repeat Dial (*66) to let you know when the number is free. You will get a special ringback when the number is no longer busy.
2) Try dialing *61 on your Verizon landline phone. In many states you will reach a feature called either Priority Call or Selective Distinctive Alert. If you add numbers to the list, when those numbers call you there will be different kinds of special rings on your phone.
Again, there will be charges on your Verizon bill to do either of these. If you use *61, be sure to clear your list and turn off the feature when you are done so you will not incur any further charges.

VoIP on Nokia phones

What Nokia models support VoIP? Anyone can give me some advices or infos about VoIP on mobile phones? Anyone has some ideea about providers and costs? Please, my gf is half way around the world and my phone bills are killing me...

All N or E series smartphones with wi-fi support VOIP.
Fring is one of the best ways to use VOIP, check their site out for price details. www.fring.com

IVR on VoIP dial-peer

In older cisco IOS, IVR can be implemented on a VoIP dial-peer with the restriction that you are using G.711 codec.
In the latest release for AS5xx0 series, has this facility been extended to other codecs?
I am using G.729 and running a calling card business. But unlike conventional networks, my customers are on the IP side (using softphones or IP phones or other smaller SOHO Gateways). So I want to provide IVR and do RADIUS billing for calling cards for calls coming in on the IP side of the gateway.
Also, can an incoming VoIP dial-peer match an outgoing dial-peer?

I hope the other codec support is there. Not sure. I checked in the following document and couldnt find it.
VoIP with IVR
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_example09186a0080094305.shtml

VoIp Billing

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