provided by Star2Billing S.L.

Well i want to open a discussion about passing Caller ID's to a trunk.
Now whenever SIP friend calls SIP friend we are passing user alias not caller id number ( callerid number is used seems like to identify SIP friend with server ).

Outgoing calls

When calling regular number, we are passing useralias number as CallerID, but due to the trunk we are using - most of trunks, terminating providers don't support caller id passing. So how does Vonage do it???
For the calling cards this is not a problem, but if we want to look into offerring broadband phone service - that becomes an issue.

Incomming calls

Incomming calls are easier - as long as DID numbers pass caller ID we ( asterisk, a2billing ) will recieve it and can pass it to SIP friend.

So i think i have issue with a2billing passing caller id which can be defined in SIP callerid section versus sending useralias.
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Also looks like we have issues with unknown caller id's calling a2billing and calls being dropped. But i think someone came with a workarround and it's listed somewhere in this forum.

CallerID has two parts "name"<number> calleridname and calleridnummber

a2b discards the "name" but passes along the <number>

in true asterisk speak there is a parameter in the iax.conf and sip.conf that is called "callerid" this is where the official callerdid is defined. How it maps into a2b is another story. so don't think of callerid as user alias.

passing callerid is terminator specific. some will use the full id you present to them. for example Voipjet does send both the callerid name and the callerid number. Some won't . For example sipdiscount and their other offerings don't or they use whatever they choose. some will just send the number ala a2b. So if you wanted to offer full voip service then you would have to find a terminator that would terminate with callerid as per your needs or you would want to have control over your PSTN lines.

The thing to ask of a2b is to have it include the "callerid name" sent in the callerid.

Hi , i get what you saying. You can find providers that can offer callerid passing, but it's hard to find ones who offer callerid name passing. I wouln't worry about that much, because even if they would do that it would be expensive. I think even vonage doesn't pass Cname. But for SIP -to -SIP it would be nice for a2b to send callerid name.

So the issue it for a2b to map callerid section to actuall callerid that it passes on to the trunk. Now i have customer's who uses names as useralias - for example "joe". So whenever he calls another sip friend he sees "joe" as Callerid. Here comes another problem - if i want to call joe customer from regular phone or voip adapter - i need to dial "9joe". So if you have numeric key pad on your phone you cannot do that, unless they have a DID number and etc.

I guess i got distracted, my main point or wish is to have a2b map caller id section in the setup to pass caller id number not useralias to the trunk

In today's A2B i think you have to take a clear stance on what you wnat to do with "user alias".

1. use it a username to login to customer interface
2. use it as user's extension number to be dialed as SIP friends call

you clearly identify the alphanumeric issues that makes doing both frought with limitations.

I for one only use it for case 1. This means that I don't utilize SIP friends calling via A2B.

In my case I define all my SIP friends with the Freepbx interface sip-additional.conf and not the A2B interface. This might be a problem for you if you have lots of users and if you trying to do this via automated signup

With this approach Sip friends calls can be done "outside" of a2b and rely on the the regular asterisk dialplan to display the callerid name and number. This way you don't have to deal with the prefix either. yes you can do it this way and still route call through a2b so that it does not have to screw up your DID billing if you do use it.

It does get tricky but this way you can get the "user alias" to access the account and have a real ID that you can use for sip friend dialing

Hello everybody,

Wish to join on this topic too!


Areski's Asterisk2billing v1.0 series was using the cardnumber to send as callerID, not the alias.

but problem happened - other person can get the cardnumber and many using same box for calling card too. so they can dial the access number and can make use out of your callingcard.

At the beginning, we didn't have forum, so we was discussing through mail.

Just writting for your information.

Hi rabon yes using cardnumber was a bad call - because anybody can see it and use it - your are right. Well now i think a2billing is perfect for calling card use, but looking in the future - DID voip - we will need some kind of caller id passing technique. It is nice to have user alias for login only, so i guess temporary thing would be to have DID number as user alias.

In this case if we would offer voip package with DID and had a terminating provider who passes on caller id that would kind of work.
User alias = DID ------> passing on ....

You right gue it gets complicated if we want to have SIP friends calling other SIP friends, being able to dial out through a2billing to pstn and have DID incomming numbers.

If we purelly have SIP to SIP setup - we don't even need a2billing, but if they want to call out how do you set up a2billing then? SIP friend would be registered with asterisk as an extension - i guess we can add *88, DeadAGI( a2billing.php|3) some kind of line in extensions.conf. So they can dial *88 and enter pin number and etc. Can we register SIP friend with asterisk as an extension and at the same time as a2billing SIP friend - that would be my question????

In this case they can call other sip friends for free without bothering a2billing ( also using asterisk to pass caller id and name ) and automatically identified by a2billing and being able to call out to pstn numbers. This thing is getting complicated.....

So anyways - Areski - what do think??? can you implement an option of in the customer setup - to be able to pick custom caller id and pass it!!!!!!

Although the callerid in the additional_a2billing_sip.conf file is the same as the account, it wouldn't really have to be I don't think. I have to check, but I swear i changed that before. Anyhow, I'd think you should be able to just change callerid in additional_a2billing_sip and then have that be passed along. Or am I oversimplifying something here ??

The callerID in sip_additional_a2billing.conf is used ( at least by me ) to identify and register sip friend with server and connect it to a calling card account. I think $user alias needs to be replace with sip's callerid somewhere in the code. I need to look through the code and find that spot - but basically after everything is almost done -- authentification - we need to replace $useralias = $callerid and pass it to the trunk.

1. set the callerid parameter in the sip friend in sip_additional_a2billing.conf to "name"<number>. There is no code in a2b that is reading the callerid from a database. Asterisk passes the callerid with the call and a2b extracts the number from $callerIdNUM. Unfortunately you have to do it manually because there is no entry on the setup or customer forms for callerid

2.


without being pedantic, you cannot register a SIP friend in asterisk and as an extension in a2b. By this i mean the sip friend cannot be defined multiple times in sip.conf and sip_additional_a2billing.conf Or if you you are using freepbx in sip_additional.conf. that does not mean that you can't accomplish goal of accessing asterisk applications without going through a2b or replicating them in a2b.

you have to have a better than average understanding of asterisk to do it but the benefits are worthwhile. in my opinion the struggle to make it work is worth the effort.

HI

In the case of voipjet, what does the extension.conf file look like in order to pass the callerid?

I have been using voipjet but I cant see the callerid being displayed when I call my home number..

Thanks
Marcias

try this fix... It is for A2B with FreePBX. You can use it as a guideline to fix your end.
http://forum.asterisk2billing.org/viewtopic.php?t=2127