[Freeswitch-users] Connect external phone

David Ponzone david.ponzone at gmail.com
Sun May 2 01:00:19 PDT 2010


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

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pièces jointes sont confidentiels et établis  
à l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autorisée est interdite. Tout message électronique est  
susceptible d'altération. IPeva décline toute responsabilité au titre  
de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes  
pas destinataire de ce message, merci de le détruire immédiatement et  
d'avertir l'expéditeur.




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">
>   <!-- 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
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100502/60704af5/attachment-0001.html 


More information about the FreeSWITCH-users mailing list