[Freeswitch-users] Wrong port on response

[David Aldworth]

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 sip:atl.teliax.net SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.192:5060;branch=z9hG4bKc1362dfab.
Max-Forwards: 70.
Content-Length: 0.
To: Aastra Test <sip:pleasehelp(aastra)@atl.teliax.net>.
From: Aastra Test  
<sip:pleasehelp(aastra)@atl.teliax.net>;tag=cbf2963ddfab0d6.
Call-ID: ecae0e043ee65c5c5340749a9d28235f at 192.168.1.192.
CSeq: 288881481 REGISTER.
Contact: Aastra Test <sip:pleasehelp(aastra)@192.168.1.192:5060;transport=udp 
 >;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  
<sip:pleasehelp(aastra)@atl.teliax.net>;tag=cbf2963ddfab0d6.
To: Aastra Test  
<sip:pleasehelp(aastra)@atl.teliax.net>;tag=Sarvm1DjmU3Zg.
Call-ID: ecae0e043ee65c5c5340749a9d28235f at 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>
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