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[daldworth at teliax.com]

Hello -


Got an issue with Freeswitch not responding on the port that the initial request was made on. I'm not beyond believing that it is a NAT or router issue except that I can register a Cisco phone from another location or a softphone from the same location without any problem. This Aastra just won't work for some reason.


We have connectile-dysfuntion turned on. Otherwise we are using the default profile settings. Auth is on (as you can see from the below). Basically, the Reg request comes from port 41450, but freeswitch responds on port 5060. Again, other UA's work fine, just one Cisco and one Aastra from this site do not. Meanwhile a soft phone from this site, and the same model cisco from another site do not work.


SIP dump and external profile are below. Thank you for any help. David


U 2008/09/19 16:51:45.145303 63.211.239.34:41450 -> 70.42.223.23:5060
REGISTER [url=sip:atl.teliax.net]sip:atl.teliax.net[/url] SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab.
Max-Forwards: 70.
Content-Length: 0.
To: Aastra Test <[url=sip:pleasehelp(aastra)@atl.teliax.net]sip:pleasehelp(aastra)@atl.teliax.net[/url]>.
From: Aastra Test <[url=sip:pleasehelp(aastra)@atl.teliax.net]sip:pleasehelp(aastra)@atl.teliax.net[/url]>;tag=cbf2963ddfab0d6.
Call-ID: ecae0e043ee65c5c5340749a9d28235f@192.168.1.192 (ecae0e043ee65c5c5340749a9d28235f@192.168.1.192).
CSeq: 288881481 REGISTER.
Contact: Aastra Test <[url=sip:pleasehelp(aastra)@192.168.1.192:5060;transport=udp]sip:pleasehelp(aastra)@192.168.1.192:5060;transport=udp[/url]>;expires=300.
Allow-Events: talk,hold,conference.
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO.
Expires: 300.
User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45.
.
U 2008/09/19 16:51:45.145493 70.42.223.23:5060 -> 63.211.239.34:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab;received=63.211.239.34.
From: Aastra Test <[url=sip:pleasehelp(aastra)@atl.teliax.net]sip:pleasehelp(aastra)@atl.teliax.net[/url]>;tag=cbf2963ddfab0d6.
To: Aastra Test <[url=sip:pleasehelp(aastra)@atl.teliax.net]sip:pleasehelp(aastra)@atl.teliax.net[/url]>;tag=Sarvm1DjmU3Zg.
Call-ID: ecae0e043ee65c5c5340749a9d28235f@192.168.1.192 (ecae0e043ee65c5c5340749a9d28235f@192.168.1.192).
CSeq: 288881481 REGISTER.
User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: 100rel, timer, precondition, path, replaces.
WWW-Authenticate: Digest realm="atl.teliax.net", nonce="d3538e86-9d86-dd11-82bf-001143e64915", algorithm=MD5, qop="auth".
Content-Length: 0.




<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="external">
<!-- This profile is only for outbound registrations to providers -->
<gateways>
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
</gateways>

<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="5060"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="manage-presence" value="false"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="true"/>
<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="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
</settings>
</profile>

[brian at freeswitch.org]

On Sep 19, 2008, at 6:08 PM, David Aldworth wrote:

[brian at freeswitch.org]

Yep just checked in my 55i that I have and no STUN nor RPORT. Great
for the LAN silly for the WAN. the force-rport is your only option
and its a global per profile setting.

/b

On Sep 19, 2008, at 6:08 PM, David Aldworth wrote:

[d at d-man.org]

The Aastra 55i should have STUN support, but not on the firmwares released
with the phone initially. Check your firmware version here:

http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-FE4A1E69/03/hs.xsl/21
669.htm

For the 480i, though, I think you're SOL.

- Darren

-----Original Message-----
From: Brian West [mailto:brian@freeswitch.org]
Sent: Friday, September 19, 2008 4:35 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Wrong port on response

Yep just checked in my 55i that I have and no STUN nor RPORT. Great for the
LAN silly for the WAN. the force-rport is your only option and its a global
per profile setting.

/b

On Sep 19, 2008, at 6:08 PM, David Aldworth wrote:

[brian at freeswitch.org]

I have looked all over the web interface and can't find it. I have the latest firmware too... weird. Still no rport support..?!?

/b

On Sep 21, 2008, at 11:45 AM, Darren Schreiber wrote:

[chris.chen2004 at gmai...]

Hi Brian, I have Aastra 55i, it does have STUN and Rport support, and I like its rport support. On web interface, under network settings->Advanced Network Settings->Rport (RFC 3581)Enabled for Rport
or STUN Server STUN Port
for STUN support, however I like rport better, which I can have call forwarding working seamlessly for freeswitch in public ip address, even my Aastra is behind my doule-NATed home network.

Best regards,
Chris


On Sun, Sep 21, 2008 at 12:51 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[brian at freeswitch.org]

That would def. solve the problem

/b

On Sep 21, 2008, at 2:03 PM, unknown wrote: