how to config an asterisk box to send calls to astpp fresswicth?

[Jean Chamblain]

I've tried with no luck. please help. I can see the call coming on fresswitch astpp and I can see the call rejected for authentication. Should configure the asterisk box as a device or gateway?

[Jeffkingsley Kigg]

creat acl in the file acl.xml 

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From: Jean Chamblain <jeanch...@gmail.com>
To: as...@googlegroups.com
Sent: Tuesday, August 6, 2013 10:03 PM
Subject: how to config an asterisk box to send calls to astpp fresswicth?

I've tried with no luck. please help. I can see the call coming on fresswitch astpp and I can see the call rejected for authentication. Should configure the asterisk box as a device or gateway? --
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[David Morris]

ASTPP should add the IP of the asterisk box to the ACL automatically.

Have you created a customer? If so go to the customer settings (Global Accounts > Customer List ) and click the edit icon for the customer who you want to validate as the asterisk box.

When you are looking at the customer details tab click on the "IP & SIP SETTINGS" tab.

Under "IP MAPPING" enter the IP address of the Asterisk box and assign it to a Price List and click MAP IP.

If you are looking in fs_cli as you do this you will see that the ACL is automatically added.

Now when a call comes from asterisk the ACL will allow it through and the call will be processed and assigned to that particular user.

Thanks,

David

skype: david-3gtelecommunications

[Jean Chamblain]

Thx a Lot! It works.  I still can not get the caller id to show at the other end. any idea?

[David Morris]

Is it showing any CallerID or just "unknown"?

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[Jean Chamblain]

Another number like I did not send any to the provider.

[David Morris]

this might be a setting on your outbound trunk in ASTPP.

Try checking in Routing -> Gateways

click Edit next to the gateway that the call is going out to

Try checning "Caller-id-in-from" to True and see if that helps.

[Jean Chamblain]

Hi David,

I did an upgrade with the script and I have deleted all data except default sip profile. I did the settings just like you said and it does not work anymore, I'm getting  NO_ROUTE_DESTINATION in Fs_cli.

any Idea?

When I create a new customer and try to map IP, I still have the choice to pick the old deleted rate groups and I cannot see them under rate group. 

[Jean Chamblain]

I cannot find acl.xml ??

[Lok]

Hi David,

I would like to know if I have 10 channels DID into asterisk, as calling card frontend, for different caller, but want to use ASTPP with Freeswitch+gateway as billing platform, is this possible ? By the way, current freeswitch and ASTPP works with SIP phone.

Actually, I want to add PSTN as local access ontop of SIP phones. Any advice ?

Thanks in advance.

Meison.

[ircactus]

To bring in PSTN (POTS) lines into the system, you'll need to choose a gateway that supports this. Possible options:

Cisco 26xx series with 2 or 4 FXO cards configured as a voice gateway

Audio Codes Make gateway devices

Patton makes gateway devices.

If you're talking about adding PRI's all of the above make t1 interface solutions to bring PSTN to voip.

[A.Lok]

Hi David,

We are planning to buy from DID provider, 10

channels as local access, and making this as SIP incoming.

We currently have Freeswitch and ASTPP configured DID == extensions/AsTPP-customer, and has a pstn gateway, and distributing sip softphone per extension/aSTPP.customer.

Our current setup is able to have sip user calling sip user and sip user can call PSTN ..

However, for those without smartphone, no sip softphone, we planned to to buy DID to let user call to the DID ( with local access ).

But I am not sure if these is possible with our current platform, and I am thinking to let DID called into asterisk with A2billing and the connect to the same PSTN gateway. ..

Do you have any comment or advice ?

Thanks in advance ...

Aik Lok

Pls excuse typo & brevity.

Sent from my iPhone

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