I have setup SIP Sorcery local running on Windows Server 2003. I setup Google Voice through
Gizmo (SIP Sorcery registers to Gizmo) and the Google Voice call out works perfectly.
The problem that I have is that I would like to attach the FXO Port of my Grandstream HT503
ATA to SIP Sorcery local. That way PSTN calls as well as VOIP calls will be sent to the end IP
phone. I managed to get the HT503 to call out using SIP Sorcery. I however cannot get any
calls into SIP Sorcery.
One problem with the HT503 is that it doesn't seem to allow you to register with SIP Sorcery
on the FXO Port (neither the HT503 or SIP Sorcery "SIP Providers" seem to be able to act as
a registrar - SIP Sorcery does act as a registrar with IP Phones/clients). There is no
place on the HT503 to enter a registration string. There is however, an Unconditional
Forward to VOIP box.
This is how it is currently setup:
HT503:
Unconditional Call Forward to VOIP: 2000586@192.168.1.250:5060
(NOTE: the HT503 does use port 5062 on as it's incoming port for the FXO port - this is configured under SIP Providers to use port 5062 when SIP Sorcery dials the PSTN of the ATA)
SIP Sorcery "SIPDOMAINS.XML"
<sipdomains>
<sipdomain>
<domain>192.168.1.250</domain>
<owner></owner>
<sipdomainaliases>
<domainalias>192.168.1.250:5060</domainalias>
<domainalias>local</domainalias>
<domainalias>*</domainalias>
</sipdomainaliases>
</sipdomain>
</sipdomains>
I am not able to forward any call to any user on SIP Sorcery local. It just doesn't
connect. Does SIP Sorcery allow calls in this way? Shouldn't I be able to send a call to a
user by sending it to "username@192.168.1.250:5060"? I have even tried
"sip:username@192.168.1.250:5060" without any luck. Since the Gizmo call backs work
perfectly it seems that when SIP Sorcery is registered it accepts the calls in. Do I need
to enable SIP Sorcery local to allow anonymous calls in somehow?
Any suggestions would be greatly appreciated. And I thank you for any help you can provide
in advance.