Siemens S685IP NAT / STUN Configuration

Discussion in '3CX Phone System - General' started by SnAzBaZ, Jan 1, 2010.

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  1. SnAzBaZ

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    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:401@81.187.*: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:401@81.187.*: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:401@64.34.*: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:401@64.34.*: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*0@81.187.*>;tag=as1e7eaa0a
    Call-ID: 719a2da746ba79405bbf9c6511a69418@81.187.*
    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.
     
  2. leejor

    leejor Well-Known Member

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    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?
     
  3. SnAzBaZ

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    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.
     
  4. Nick Galea

    Nick Galea Site Admin

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    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
     
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  5. SnAzBaZ

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    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
     
  6. slarty01

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    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
     
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