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 Post subject: Softphone DID and sip trunk
PostPosted: Thu Feb 11, 2010 7:18 am 
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Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
Hi,

I have installed Elastix and I am going to accept calls from Sofphones and forward them to a terminating company.

Simply this is my setup

Softphone --> Asterisk --sip trunk--> provider

For this work do I need a DID number?

Can I use a imaginary number (something like 500) as DID and forward all calls coming to that to sip trunks according to the prefix?

An explanation is appreciated. :)


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Thu Feb 11, 2010 8:52 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
VoIP endpoints register directly to A2Billing, and make calls outbound. No DID is required.

Joe


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Thu Feb 11, 2010 9:48 am 
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Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
jroper wrote:
VoIP endpoints register directly to A2Billing, and make calls outbound. No DID is required.

Joe


Thank you for the prompt reply.

Can you direct me a document which has directions for the setting I should do. So far I found explanations which involve DID.

In current setup I have, when dial a number, it says the balance and request enter the number again. When enter the number and #, it says the destination is not available.

This is the error from log:

a2billing.php|1: file:Class.A2Billing.php - line:719 - ERROR ::> RateEngine didnt succeed to match the dialed number over the ratecard (Please check : id the ratecard is well create ; if the removeInter_Prefix is set according to your prefix in the ratecard ; if you hooked the ratecard to the Call Plan)

Any hint?

Thanks


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Thu Feb 11, 2010 10:14 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
The clue is in the error message - you do not have a rate to match the destination you are dialling for that call plan belonging to the customer.

Joe


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Fri Feb 12, 2010 12:00 am 
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Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
Thanking you, There was a prefix problem and fixed it. Now it is working fine and I am very happy :) .

I need a help fore one more adjustment. In current setup I have, when dial a number, it says the balance and request to enter the number again and press #. In my case, dialing number twice is an extra burden to the user. Is there a technique to get following set of events?

1. dial the number by user
2. Tell the balance
3. Transfer the call to the destination (after a message such as "Please wait while transfer your call" or without any message)

My release is Asterisk2Billing - Version 1.3.0 (Yellowjacket).

Thanking you.


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Fri Feb 12, 2010 1:14 pm 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Read the agi-conf. The answer is in there - e.g. use_dnid=yes

and seriously consider upgrading to latest stable ( 1.5)


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Mon Feb 15, 2010 3:40 am 
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Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
Thank you for the reply. Now everything works fine. :)

We all should consider to do a donation to keep this excellent work up. Its more than I expected for my small business and its free of charge!!!

By the way how can I stop any chances of billing without really connecting to the call termination gateway. I felt that some calls are billed without connecting to the destination phone. May be the error with termination gateway. But still I like to know whether any configuration changes should be made.

This is my setup again

Softphone --> Asterisk --sip trunk--> termination gateway --> destination phone

Thanks


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Mon Feb 15, 2010 7:17 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
You have to ensure that the carrier is not falsely answering the call. If they are, raise a trouble ticket with them, or change carrier.

Joe


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Mon Feb 15, 2010 8:23 am 
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Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
jroper wrote:
You have to ensure that the carrier is not falsely answering the call. If they are, raise a trouble ticket with them, or change carrier.

Joe


I am a newbie and can you elaborate a little or send me a link to read "falsely answering the call". How can I know that my carrier is falsely answering
the call?


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Mon Feb 15, 2010 8:48 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
The carrier answers, and you start billing your customer before the person you are actually calling answers the call.

Joe


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Tue Feb 16, 2010 5:03 pm 
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Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
Thank you. It helped.

When we use x-lite as the sip phone, the timer of the display start from the time connected to the server. Can we do any adjustment so that it starts counting when it really connected to the destination.

Any ideas are appreciated.

Thanks :)


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Tue Feb 16, 2010 5:09 pm 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Don't answer the call at A2Billing.

Joe


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Thu Feb 18, 2010 11:14 am 
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Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
Thanks,

When I dial a number. A server simulated ringing tone is coming. To hear the original ringing tone, Asterisk says to remove option r from App Dial, is this replaced by a2billing?. Where should I change it?.

Thanks again


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Thu Feb 18, 2010 11:24 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Change the dial command parameters in the agi-conf

Joe


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 Post subject: Re: Softphone DID and sip trunk
PostPosted: Sat Feb 20, 2010 12:20 pm 
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Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
Thanks. It works fine.

Could you please clarify this too.

I have two providers for the same destination with two different buying rates.

I created two trunks for them and added second trunk as the FAILOVER TRUNK of first. First one has a Rate Card and Rate. Do I need a Rate Card for second trunk? or I should use the same Rate Card and add a Rate for the FAILOVER TRUNK?

Thanks :)


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