[touche]

Hi guys,

I have a sipura 3000 and i use Inphonex to make my calls.
Today i bought other Voip system – Vono – only to receive calls from
other state of my country. My question is: Is it possible to put these Inphonex/Vono to work together? Eg: Inphonex makes the calls and Vono only receives them?

If it's possible, please tell me how to do it, step by step.

Thanks

[hwittenb]

You may be able to do it, depending...

You need to put the credentials for your incoming did provider, Vono, on line 1 and set it to register. You need to be able to get it to work without using an outbound proxy, i.e. just your username, password, displayname, and the inphonex proxy.

A STUN server may help with your router's network address translation transversal. On the Sip tab you would set
Substitute VIA Addr:YES
Send Resp To Src Port:YES
STUN Enable:YES
STUN Server:stun.softjoys.com (or any other STUN server)
on the Line 1 tab set NAT Mapping Enable: YES
on the Line 1 tab set NAT Keep Alive Enable: YES

Assuming that Inphonix will make a call without registration, just by sending your password when you place the call, in the Gateway1 field you put userid@inphonex.com where userid is your userid. In the GW1 Auth ID field you put your userid, in the GW1 Password field you put your password, and set GW1 NAT Mapping YES.

To dial all your outgoing calls on Inphonex, which is set for gateway 1 or @gw1, you setup a dial plan something like this (xx.<:@gw1>).

[touche]

Hi hwittenb,

thank you, for your explanation, i'll try it!
But if nothing happens, could you please try to set up it for me through internet?
I'll give you my address and port if you don't mind.

Bellow i sent the configuration of Vono, maybe it could help you.

(domain): vono.net.br
(username): “my login”
(password): “my pass”
Codecs suported: (preferred): iLBC, GSM, g.729, g.711a e g.711u

SIP server: 200.146.79.165
SIP port: 5060

or 200.146.79.165:5060

Thanks a lot!

Joaquim Saraiva

[airsky98]

i'm planning to do the same setup also

just found out where the problem lies with such setup..

ur primary account will have a userid like (123456) , so when u dial out . it will be using 123456@example.com

when u setup a gw1, even with userid(654321)/password.. when u dial out via gw1, the phone will still send authentication as 123456@gw1.com instead of 654321@gw1.com

the only way to solve it , is to set a translation rule for the calling id to be 654321 when called id is for gw1..... (those can be done on cisco router as i face such problem before when i was playing with it last time)

i wonder can this sipura 3000 do that? if so, 2 diffierent voip systems will work........

[hwittenb]

Originally Posted by airsky98

when u setup a gw1, even with userid(654321)/password.. when u dial out via gw1, the phone will still send authentication as 123456@gw1.com instead of 654321@gw1.com

If you use ethereal to look at the packets sent from a Sipura adapter I think you will find that your analysis is not correct. A sip Invite is sent to the gateway proxy with authentication showing the userid listed in the GW Auth ID field.

It is true that the Display Name field of line 1 will be used in the sip Invite as the contact name.

The biggest configuration problem with using providers configured in the gateway fields is that a user often has an outbound proxy configured under Line 1 and in this case the sip Invite is sent to that proxy instead of the gateway proxy. In this case the sip Invite idoes not work. The other problem is that the adapter does not send a Register request to the gateway provider. Some providers require registration before they will process a sip Invite.

[airsky98]

opps.. that's what i trying to say, maybe i shouldn't use authenticate that word...

it's irritating that it keep sending the 'wrong' id to my gateway 1 (another sip provider)

anyway, do u have any idea what the SIP proxy,etc for PSTN line for ???

[hwittenb]

Originally Posted by airsky98

it's irritating that it keep sending the 'wrong' id to my gateway 1 (another sip provider)

For the proxy in a gateway you put userid@proxy. For example if the proxy was sip.voipstunt.com and your userid was airsky98 you would put airsky98@sip.voipstunt.com.

anyway, do u have any idea what the SIP proxy,etc for PSTN line for ???

The entry for the sip proxy etc on the PSTN tab is for the voip account that you wish to use to bridge a call with the pstn line. This is a separate voip account from the one that you have on line 1. You can forward a call from line 1 to the pstn line, but otherwise you must use the voip account that you have listed on the pstn tab to bridge a call with the pstn line.