Internal Calls Possible

[vps4hire]

Lets say i have 2 clients they both have sip devices and DID numbers allocated to them. Can customer 1 call customer 2 without using outbound trunk??

What will the setup be?? I deally i need to make extension to extension calls without charging the clients

[Samir Doshi]

Hi, 

Yeah, you can do that using DID numbers. 

Steps : 

1. Create DID for customer

2. Set call type = Local 

3. Set Destination = SIP Device username 

Best Regards
--
Samir Doshi
iNextrix Technologies Pvt. Ltd.
http://www.inextrix.com

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On Wed, Mar 4, 2015 at 10:19 PM, vps4hire <omp...@gmail.com> wrote:

Lets say i have 2 clients they both have sip devices and DID numbers allocated to them. Can customer 1 call customer 2 without using outbound trunk??

What will the setup be?? I deally i need to make extension to extension calls without charging the clients

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[vps4hire]

what will be the call flow from one user to the other??  caller 1 want to call caller 2 from sip to sip. Does the call go straight to the other caller or does it have to pass through to the trunk provider first???

[Samir Doshi]

User1 just have to dial DID number of user2. That will be on-net call. It will not use trunk. 

Best Regards
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Samir Doshi
iNextrix Technologies Pvt. Ltd.
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[vps4hire]

thanks a lot

[Jon Morby]

On 4 Mar 2015, at 19:07, Samir Doshi <samir...@inextrix.com> wrote:

User1 just have to dial DID number of user2. That will be on-net call. It will not use trunk. 

I’m currently having problems with this … calls from user1 to user2’s DDI are going out via the trunk provider and then back in 

Each user has a dial translation which translates the number into full iddd format (country+number) ie 448444844480 based on where they’re dialling from (so UK would prefix 44 if dialled as a local number, Spain 34 and so on) … currently these calls go out via the trunk provider and then back in rather than routing directly “on net"

If I insert the DDI into the database in “local” format (08444844480) then it routes on net, but then inbound calls fail as they’re presented in iddd format (448444844480) by our upstreams .. I’ve tried dial translations on the trunk but that doesn’t seem to have any effect (and switching the way calls are translated will break as we route calls for Spain, France, UK, USA, etc “local” numbers)

It seems the outbound dial logic needs to take into account the post translation numbers before attempting to dial on-net / off-net / etc??

www.fidonet.com - low cost domains, web hosting and virtual servers

www.fidotelecom.com - UK based ITSPA - holders of the ITSPA Quality Mark

[Samir Doshi]

Hi Jon, 

Number translation applies only on outbound calls. 

As of now to match with the DID numbers for on/off-net calls, you will need to add same DID twice. This is quite duplication of work. 

We will look into that to find some work around. 

Best Regards
--
Samir Doshi
iNextrix Technologies Pvt. Ltd.
http://www.inextrix.com

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[Jon Morby]

Thanks

With more than 250,000 DIDs in our database that is a lot of duplication :)

Jon Morby

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[Mojo]

Hi Samir,

I am trying to do this. I have done steps 1 2 3 in your message but on-net calls are not working.

EXECUTE sofia/default/xxx@<mysipserver> log(WARNING Accountcode yyy. Dialed number (xxx) origination rates not found!!)

2015-08-20 05:35:46.525835 [WARNING] mod_dptools.c:1670 Accountcode yyy. Dialed number (xxx) origination rates not found!!

What else do I need to do?

Mojo

[Mojo]

A further question: why is ASTPP looking for origination rates when I have defined the DID and SIP Device User Name?

Mojo

[Samir Doshi]

If you are dialing exactly same number as DID number than it must not look for origination rates. 

Would you please provide us call log from astpp.log? That will be helpful to find the issue. 

Best Regards
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iNextrix Technologies Pvt. Ltd.
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[David]

Hello,

Can someone help out with this inboud calls issue?

[Samir Doshi]

Configure 0/44 in Global DID translation. That should do your job. 

You will get that field from Settings -> Global. 

Sent with Mailtrack

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iNextrix Technologies Pvt. Ltd.
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[OBELLO]

Do anybody know how to pass login information for curl auto login?

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[Samir Doshi]

What are you trying to implement? 

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[OBELLO]

we have a mobile application and we want url to send user name and password to long in, get balance rechard....

[Hardik Patel]

Hi OBELLO

the current code is redirecting users to the dashboard after successful login so it will not useful for you if you wish to pass username and password in recharge API then you need to customize the method or you can implement new method for authentication.

Thanks and Regards,
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S - hardik.inextrix
P - +1 855 580 1802 | +91 98983 47245

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[OBELLO]

how do we implement autologin passing the variables using POST/GEt?

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[Hardik Patel]

That's depend on your choice. you can use any method.

Thanks and Regards,
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[OBELLO]

can you please give a sample url?

Bellow will not work

http://www.xtz.com:8089/username=abc&password=cxt 

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[Hardik Patel]

It will not work simply you need to customize it as per your need.

so you have to make some coding in existing method or you can develop a new method / function for that.

Thanks and Regards,
--
Hardik Patel

iNextrix Technologies Pvt.Ltd

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[OBELLO]

is there a documentation on this that will guide us?

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[Hardik Patel]

Nop, i don't think that you required documentation for that. its simple PHP code that you need to do.

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[OBELLO]

Please how do we setup call forward on busy ?

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[Hardik Patel]

ASTPP does not have this feature by default but if you are familiar with lua code then you can customize the script code and make it feasible.

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[OBELLO]

Can you help out with this ?

we need help setting up auto login, balance check, auto rechard and other features such as call forwading, call waiting, forward on busy...

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[Hardik Patel]

Sure, 

You can share your requirement on sa...@inextrix.com we will develop all the functionalities for you.

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[Rahul Mallick]

Hi Samir,

I did what you suggested, but the problem is the SIP Account is not receiving incoming call, its coming as user busy.

I am calling as pstn to DID to Local extension.

DID setup is done as local. Any suggestion, please

[Samir Doshi]

Did you check sip log? FS must be getting busy from UAC. 

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[Rahul Mallick]

​What is UAC?​

​​

Sent from Mail for Windows 10

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[OBELLO]

How do we setup priority routing?

we have 2 DIDs, we want the first DID ring first and then the second DID

On Mon, Sep 11, 2017 at 11:45 AM, David <dobca...@gmail.com> wrote:

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[OBELLO]

I am getting SUBSCRIBER_ABSENT even when the sip device is registered and was able to make outbound calls.

Please help

On Mon, Sep 11, 2017 at 11:45 AM, David <dobca...@gmail.com> wrote:

[David]

We recently migrate to this platform from a2billing and it has been a hell of headache...

We do have sip profiles that are able to make phone calls.

we do allocate the DID for the sip devices.

the sip devices can make phone call but not able to receive calls both internal(sip-sip) and from external(PSTN-sip DID)

We keep getting SUBSCRIBER_ABSENT, how could such be when users are able to make outbound calls to PSTN

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[Samir Doshi]

Which DID call type are you using? 

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[Rahul Mallick]

I am using did local to transfer to a sip account

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[Samir Doshi]

Try with SIP-DID and check. 

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[OBELLO]

pstn to our server -inbound call 

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[David]

How do we setup forward on busy on version 3.5

[Samir Doshi]

At the moment forwarding feature is not yet implemented in ASTPP. Either you will need to write custom code for that or you can email to sa...@inextrix.com. 

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Best Regards
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Samir Doshi
iNextrix Technologies Pvt. Ltd.
http://www.inextrix.com

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