On Sep 19, 2008, at 6:08 PM, David Aldworth wrote:
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
This isn't a bug. If you notice the phone explicitly said in its
contact for us to contact them via 192.168.1.192:5060 so you'll need
to enable stun on the phone or rport. If you were to enable rport on
the aastra it would just work correctly. If Aastra doesn't support
RFC3581 or STUN then they are worthless phones just like Polycom. Its
not the registrars problem to fix your nat issues. The phone should
support RFC3581 (rport) or STUN and it would just work like the Snom's
do.
Try adding this param to your sofia profile. It will break cisco
phones or any other phone that follows the sip spec. This explicitly
breaks RFC to accommodate broken phones.
<param name="NDLB-force-rport" value="true"/> in your sofia profile.
/b
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: [EMAIL PROTECTED]
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: [EMAIL PROTECTED]
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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