Re: [Freeswitch-users] Connect external phone

David,

I did double check firmware version for my phone, it’s the most up-to-date. I have been able to connect the phone inside the the local lan no problem, I’m attempting to test an external connection for office employee’s who are in different city locations.

I turned on trace using this command, sofia profile gexternal siptrace on. The only logging that I saw referencing M3 I provided below. I could turn on sofia loglevel for additional info if that helps.

I wonder, since my phone is behind a NAT if that’s causing an issue.

Thanks
Sean

On 5/2/10 1:00 AM, "David Ponzone" <david.ponzone@gmail.com> wrote:

Sean,

my suggestion would be not to reinvent the wheel.
If you need another profile to accept registrations, just copy internal.xml and modify what you need.

But I think there is another issue here.
FS sends back a 401, which is normal, but your SNOM should then send another REGISTER with Digest auth.
It doesn't OR your failed to include the whole trace.
If it doesn't, it means it's screwed.
I had this issue with a very bad ATA from Telco Systems.
But I think some people use the M3 on FS, so I guess there is way to make that work (perhaps a firmware update on the M3 or a parameter).

First thing, can you confirm ASAP you included the whole trace ?

David Ponzone Direction Technique
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gsm:   06 66 98 76 34

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Le 02/05/2010 à 03:30, Sean Holt a écrit :

Hello list,

I keep getting this authentication error when I attempt to connect a device from the outside. I followed the wiki that suggest creating a separate profile to handle external device connections.

Here’s my external xml file
<profile name="gexternal5090">
   
   <gateways>
     <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
   </gateways>
   <aliases>
     <alias name="outbound"/>
   </aliases>
   <domains>
     <domain name="$${domain}" parse="true"/>
   </domains>
   <settings>
     <param name="debug" value="0"/>
     <param name="sip-trace" value="no"/>
     <param name="rfc2833-pt" value="101"/>
     <param name="sip-port" value="5090"/>
     <param name="dialplan" value="XML"/>
     <param name="context" value="public"/>
     <param name="dtmf-duration" value="100"/>
     <param name="codec-prefs" value="$${outbound_codec_prefs}"/>
     <param name="hold-music" value="$${moh_uri}"/>
     <param name="use-rtp-timer" value="true"/>
     <param name="rtp-timer-name" value="soft"/>
     <param name="manage-presence" value="false"/>
     <param name="inbound-codec-negotiation" value="generous"/>
     <param name="nonce-ttl" value="60"/>
     <param name="auth-calls" value="false"/>
     <param name="rtp-timeout-sec" value="1800"/>
     <param name="rtp-ip" value="$${local_ip_v4}"/>
     <param name="sip-ip" value="$${local_ip_v4}"/>
     <param name="ext-rtp-ip" value="71.133.39.219"/>
     <param name="ext-sip-ip" value="71.133.39.219"/>
     <param name="rtp-timeout-sec" value="300"/>
     <param name="rtp-hold-timeout-sec" value="1800"/>
   </settings>
</profile>

I did a siptrace on the incoming connection attempt
   ------------------------------------------------------------------------
    REGISTER sip:71.133.39.219:5090 SIP/2.0
    Via: SIP/2.0/UDP x.x.x.x:5090;branch=z9hG4bKvvaun9r6.1u8ej
    Max-Forwards: 70
    From: <sip:1001@71.133.39.219>;tag=i9w3pt4bm4tpsl
    To: <sip:1001@71.133.39.219>
    Call-ID: fo33ia7sk6qy9vnk6h
    CSeq: 75203 REGISTER
    Contact: <sip:1001@x.x.x.x:5090;line=20880>
    Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK
    Expires: 600
    User-Agent: snom-m3-SIP/02.02 (MAC=0004132AD0BE; HW=255)
    Content-Length: 0

    ------------------------------------------------------------------------
send 591 bytes to udp/[x.x.x.x]:5090 at 01:26:47.109165:
    ------------------------------------------------------------------------
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 10.0.10.231:5090;branch=z9hG4bKvvaun9r6.1u8ej;received=207.47.31.108
    From: <sip:1001@71.133.39.219>;tag=i9w3pt4bm4tpsl
    To: <sip:1001@71.133.39.219>;tag=Zc7mBaBN723Bp
    Call-ID: fo33ia7sk6qy9vnk6h
    CSeq: 75203 REGISTER
    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
    Supported: timer, precondition, path, replaces
    WWW-Authenticate: Digest realm="71.133.39.219", nonce="bd1ecf94-5589-11df-b3f4-c1a375adf51a", algorithm=MD5, qop="auth"
    Content-Length: 0

    ------------------------------------------------------------------------

Thoughts?
Thanks
Sean
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