I am trying to get maximum usability from Linksys PAP2T.
The phone connected to line 1 and it configured with Voipstunt for outgoing calls. Username here has to be __alphabetical__ (something like
abcdef@voipstunt.com).
I want to receive incoming calls on the same phone. Sipbroker seems to be a perfect candidate, but it uses __numerical__ username for the destination (something like
123456@yourprovider.com) – can NOT be combined with Voipstunt on line 1.
All right, I have second line in PAP2T – configure it to receive incoming calls without registration, username 123456, setup Sipbroker destination
123456@myhost.com – it works, I can call my Sipbroker alias from SIP phone (X-Lite) or through Sipbroker PSTN phone number. But I am receiving call to Line 2 – have to have second phone… Not convenient, I want to use only ONE phone!
Now I configure PAP2T Line 2 to forward incoming calls (after a few seconds) to Line 1. Calling my Sipbroker alias from X-Lite – it works. First it rings phone on line 2, than rings line 1, I can answer and talk… Calling the same Sipbroker alias through PSTN number – line 2 rings for a few seconds (I can answer and talk), but line 1 does NOT ring - as soon as ATA attempts to forward call to line 1 caller hears busy beeps...
What’s wrong? The same setup does work from soft-phone (X-Lite) and does not work through Sipbroker PSTN. Does Sipbroker prohibit forwarding? Why?
Is there a way to setup Sipbroker destination with alphabetical username (user@provider.com)? Any other ideas? Almost sure that Asterisk may solve the problem, but I would prefer to avoid unnecessary complexity.
Thank you for reading my long description.
Any suggestions would be appreciated.
--- oleg
P.S.
Just recognized – Sipbroker allows alphabetic username in destination, but does NOT allow port (user@provider.com:5061). Although the listing on
SIPBroker - Provider White Pages includes one record with port (proxy.netphonedirectory.org:5065). How to specify port?