[Freeswitch-users] Connect external phone
Sean Holt
sean at obscuradigital.com
Sun May 2 09:56:53 PDT 2010
David,
I did double check firmware version for my phone, it¹s the most up-to-date.
I have been able to connect the phone inside the the local lan no problem,
I¹m attempting to test an external connection for office employee¹s 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 that¹s causing an issue.
Thanks
Sean
On 5/2/10 1:00 AM, "David Ponzone" <david.ponzone at 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
> email: david.ponzone at ipeva.fr
> tel: 01 74 03 18 97
> 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.
>>
>> Here¹s my external xml file
>> <profile name="gexternal5090">
>> <!-- This profile is only for outbound registrations to providers -->
>> <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 at 71.133.39.219>;tag=i9w3pt4bm4tpsl
>> To: <sip:1001 at 71.133.39.219>
>> Call-ID: fo33ia7sk6qy9vnk6h
>> CSeq: 75203 REGISTER
>> Contact: <sip:1001 at 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 at 71.133.39.219>;tag=i9w3pt4bm4tpsl
>> To: <sip:1001 at 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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