[chetannain]

Hi Gang,

I require urgent advice on dial plan configuration for the following setup:

1. We have a VoIP gateway from Innovaphone that does not feature Analog PSTN interfaces, but only digital PSTN interfaces. It was decided to use the Sipura to connect to the public analog network, and create a SIP trunk from the Innovaphone gateway to the Sipura. So, in effect, the user on the Innovaphone gateway would dial a '0' from his phone followed by the rest of the number (a cellphone number like 0420xxxxxx), the Innovaphone would then route this user to the Sipura via the SIP trunk, and dial out the dialed number from the analog FXO line on the Sipura.

Innovaphone ---SIP Trunk---->Sipura----FXO to PSTN network---->

2. After extensive testing, I have found that when the user on the Innovaphone dials the number, he directly gets the analog PSTN dial tone from the Sipura adapter. He then has to dial the entire cellphone number again to dial out. In other words, user dials 0420xxxxxx, gets the PSTN dial tone, and then has to dial 0420xxxxxx again to make the call to the cellphone (i am in Australia).

So, the Innovaphone gateway sends the entire number to the sipura, and the sipura processes only the 0 from the number and then after a delay, provides the analog dial tone and discards the rest of the number that was sent to it. Hence, the user has to dial the entire number again.

My question is:

Is is possible to configure such a dial-plan on the sipura FXO interface so that the sipura would receive the entire number from the Innovaphone, after a while then present the dial tone, and after presenting the dial tone, dial the rest of the number that was received (rather than discarding it)??

Guys, any and all help is GREATLY appreciated. I have spent a HUGE amount of time trying to figure out the dial plan string, and i do not know if this is even possible.

Thank very much.

regards,
Chetan

[hwittenb]

I am not knowledgeable about the capabilities of your Innovaphone PBX, however you must be sending a Sip Invite to the SPA3102 in order for the voip-to-pstn gateway to return the dialtone. The key to what you wish to do is in programming the PBX..

The SPA3102 is designed to either return a dialtone as you describe or it can receive a Sip Invite and dial a number directly. I don't believe you can do what you want to do with a dial plan after the SPA accepts the call unless you wish to restrict the number to a set "hot-line" number. My impression is you wish to be able to dial any pstn number.

I am aware of two ways to receive a sip invite and dial a number directly on the SPA without returning a dial tone.

One technique is called http authentication on the SPA. If, on the PBX, you can setup the SPA as a proxy with a user id / password you can dial the number you wish on the PBX. Using a sip invite, the PBX sends the number to the SPA with the user id / password, the SPA will dial the number on the FXO port. This is a common way the SPA3102 FXO port is interfaced to an Asterisk PBX.

Another technique is to send a sip uri to the SPA that includes the number you wish the SPA to dial. The format of the sip uri would be 1234567@192.168.1.100:5061 where 1234567 is the number you wish to dial, 192.168.1.100 is the ip address of the SPA3102 and 5061 is the port number of the PSTN tab on the SPA3102.

The default dial plan on the voip-to-pstn tab must allow the number being dialed.

[chetannain]

Thanks hwittenb, I appreciate your detailed advice.

At the moment, the setup is such that the Sipura registers to our Innovaphone gateway. The Innovaphone gateway is our SIP server.

As you said:
#####
One technique is called http authentication on the SPA. If, on the PBX, you can setup the SPA as a proxy with a user id / password you can dial the number you wish on the PBX. Using a sip invite, the PBX sends the number to the SPA with the user id / password, the SPA will dial the number on the FXO port. This is a common way the SPA3102 FXO port is interfaced to an Asterisk PBX.
#####

does this mean that instead of registering the sipura with our innovaphone, i would make the sipura the SIP server and the innovaphone the client? If so, how would i configure this on the sipura?

the config screens on the sipura for me are:





At the moment, the sipura registration on the SIP server is successful. Could you please assist with configuration options?

also , for the second option you suggested:

####
Another technique is to send a sip uri to the SPA that includes the number you wish the SPA to dial. The format of the sip uri would be 1234567@192.168.1.100:5061 where 1234567 is the number you wish to dial, 192.168.1.100 is the ip address of the SPA3102 and 5061 is the port number of the PSTN tab on the SPA3102.
####

how would i configure this?

many thanks !!
chetan

[chetannain]

Also, 192.168.2.100 is our Innovaphone SIP Registrar/Server..

[humba3]

does this mean that instead of registering the sipura with our innovaphone, i would make the sipura the SIP server and the innovaphone the client? If so, how would i configure this on the sipura?

Yes, and it's explained in my spa9k spa3k interop guide (SPA3k & SPA9k interop explained) - look for the proxy method which does exactly that. You just need to be able to configure a proxy without registration (but that still routes calls without registration) on the Innovaphone PBX.

[hwittenb]

Originally Posted by chetannain

does this mean that instead of registering the sipura with our innovaphone, i would make the sipura the SIP server and the innovaphone the client? If so, how would i configure this on the sipura?

You will need to set
Ans Call Without Reg: YES
VoIP Caller Auth Method: http digest

Under VoIP Users and Passwords (HTTP Authentication)
VoIP User 1 Auth ID: userid (same as you setup on your PBX)
VoIP User 1 DP: 3 (just a plain dial plan like (xx.))
VoIP User 1 Password: password (same as you setup on your PBX)

On your PBX you will need to setup the Sipura as a Proxy that doesn't register (because the Sipura cannot return an inbound register request).

You can do what you wish with the PSTN tab Proxy settings depending on whether or not you have inbound calls coming on the PSTN line and wish to make an inbound call to the PBX. If you do you need to have the pstn-to-VoIP gateway point to a simple dial plan like (xx.)

also , for the second option you suggested:
how would i configure this?

You should use the first method, http authentication option, if possible.

For the second option you need to set the VoIP Authentication to None, and probably Ans Call Without Reg. You would need to figure out how to send the call as a sip uri.