Siemens S685IP NAT / STUN Configuration

[SnAzBaZ]

I am trying to setup a Siemens S685IP phone with our hosted 3CX machine. The phone is located behind a m0n0wall firewall, with a 1:1 NAT setup.

I am able to make calls from the phone, and I get full two way audio. However the phone will not receive a call.

Here is a partial log from when the call is made:

07:52:03.908 [CM503008]: Call(21): Call is terminated

07:51:39.068 [CM503025]: Call(21): Calling Ext:Ext.401@[Dev:sip:[email protected].*:3949]

07:51:39.068 [MS210002] C:21.2:Offer provided. Connection(transcoding mode): 64.34.*:9030(9031)

07:51:39.026 [CM503004]: Call(21): Route 1: Ext:Ext.401@[Dev:sip:[email protected].*:3949]

07:51:39.025 [MS210000] C:21.1:Offer received. RTP connection: 81.187.*:15082(15083)

07:51:39.025 [CM503010]: Making route(s) to <sip:[email protected].*:5060>

07:51:39.024 Remote SDP is set for legC:21.1

07:51:39.024 [CM505003]: Provider:[*] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [A&A] PBX contact: [sip:*@64.34.*:5060]

07:51:39.023 [CM503001]: Call(21): Incoming call from 07*@(Ln.10004@013*) to <sip:[email protected].*:5060>

07:51:39.018 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10004 forwards to DN:401

07:51:39.017 Looking for inbound target: called=013*; caller=07*

07:51:39.015 [CM500002]: Info on incoming INVITE:
INVITE sip:013*:*@dolphin.fer*.com SIP/2.0
Via: SIP/2.0/UDP 81.187.*:5060;branch=z9hG4bK5c288e3a;rport=5060
Max-Forwards: 70
Contact: <sip:078*@81.187.*>
To: <sip:013*:*@dolphin.fer*.com>
From: "07*"<sip:07*[email protected].*>;tag=as1e7eaa0a
Call-ID: [email protected].*
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Date: Fri, 01 Jan 2010 12:51:38 GMT
Supported: replaces
User-Agent: A&A
Content-Length: 0

I have replaced sensitive information with an *, the log shows that all of these IP addresses are correct.

So what is stopping the Siemens from answering it's invite request ?

My firewall is setup log all blocked packets, and I can't see that it's blocking anything. Has anyone else got a Siemens Gigaset phone working from behind NAT ?

Thanks for your help.

 

[leejor]

07:51:39.026 [CM503004]: Call(21): Route 1: Ext:Ext.401@[Dev:sip:[email protected].*:3949]

07:51:39.025 [CM503010]: Making route(s) to <sip:[email protected].*:5060>

Is Ext 401 on IP 81.187.XXX.XXX port 3949 or on IP 64.34.XXX.XXX port 5060, or are there two Ext 401's ?
What does the 3CX log look like when 401 first registers? Is there another SIP device at the "remote" location using the same port number?

 

[SnAzBaZ]

07:51:39.026 [CM503004]: Call(21): Route 1: Ext:Ext.401@[Dev:sip:[email protected].*:3949]

07:51:39.025 [CM503010]: Making route(s) to <sip:[email protected].*:5060>

Is Ext 401 on IP 81.187.XXX.XXX port 3949 or on IP 64.34.XXX.XXX port 5060, or are there two Ext 401's ?
What does the 3CX log look like when 401 first registers? Is there another SIP device at the "remote" location using the same port number?

Ext 401 is on IP 81.187.xx.xx. The port changes each time the phone registers, I think this is something to do with the STUN server. I have to have STUN turned on on the Gigaset config page, otherwise it's internal IP (192.168.8.3) shows up in the 3CX logs, using STUN was the only way I could get it's real IP to appear.

64.34.xxx is the public IP of our 3CX server, this is hosted in a data centre and STUN has been disabled. I have Polycom SoundPoint IP phones working fine in exactly the same setup, but they have specific NAT settings in their config file that I can enable to make it work.

There are no other SIP devices at the remote location.

Thanks for your help.

 

[N_G]

You can use the SIP proxy manager if there is a Windows machine available in the remote lan... Then use the 3Cx tunnel to bypass firewall or phone issues

 

[SnAzBaZ]

Thanks for the reply. I started to have another look at this last night, but for some reason it now "just works". I noticed that the Siemens phone is now using port 5060 and not 3949 which may be why, but I don't know why it was using that port before.

I guess it could be a firewall thing, it just took a while for it to update it's NAT rule set properly. I'll see how we get on with it, and report back if it does not work.

Thanks

 

[slarty01]

Hi

Please I have an S685 IP and a basic 2 line pabx setup with one voip line. I have followed the 475 ip setup and got ir registered but can't get the phones to ring please how did you get on in the end?

Many thanks