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#435991 01/18/10 10:45 AM
Joined: Jan 2010
Posts: 3
OP Offline
Joined: Jan 2010
Posts: 3
Just been pre-setting this up before delivery to client, and come across a strange issue with SIP.

WAP IP is assigned as a DHCP reservation on the router, and is DMZ so port forwarding should be no issue.

The phone is set to use the correct proxy / domain, and the right SIPID and Secret, and it shows as 'Registered'... (the SIP Provider web UI shows that the device is online too)

However, that's where it ends.

I can originate a call from VoIP to PSTN, but the VoIP doesn't acknowledge connection... I can't call PSTN to VoIP, nor VoIP to VoIP.

I've run packet-sniffing software, and I'm not seeing anything like the 'usual' SIP traffic I normally do. In fact, I'm not even seeing the registration stuff. All I see is the usual ARP, DHCP broadcasts, and then that's it - the device seems to be totally silent.

Even when I try to make a VoIP call, it just times out, tries to fall back to PSTN, but WireShark shows no attempt at the call.

The TCP stack on it is fine, as I can use the Web UI of the phone. On the WAN side, port 80 is the only open port. On the LAN side, there's port 80, port 53 for DNS as well as port 5002 and 5012 - but I can't seem to connect any service on those ports.

Can anyone help me with some more technical information on the unit, so I can get it working as it should?

Thanks for reading

Atcom VoIP Demo
VoIP Demo
Joined: Dec 2004
Posts: 4,096
Joined: Dec 2004
Posts: 4,096
That is probably part of your problem. If port 80 is the only open port on the wan side. How does the sip talk to the system? Surely not port 80 probably more along the lines of port 5060 or something. That depends on your provider though.

Joined: Jan 2010
Posts: 3
OP Offline
Joined: Jan 2010
Posts: 3
When I say 'open', I mean it's open to a portscan. The strange thing I've noticed this morning is that there MUST be SIP messages getting passed, as if I dial a known-dead number, the call fails, whereas a known-good does not.

Seems more like the SIP traffic is flowing (somehow!) but it's not able to send/receive the RTP side.

For a 'SoHo' system, it's the most uncustomisable thing I've ever come across. I've never ever had a problem with a single VoIP device until now!!

Is there a CLI / Telnet / SuperUser hidden somewhere on this device?

Moderated by  nameless, pvj 

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