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@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@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>
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: ecae0e043ee65c5c5340749a9d28235f@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@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> _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.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:
User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/ v3.2.8.45.
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:
User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/ v3.2.8.45.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.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:
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
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> wrote:
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:
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
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That would def. solve the problem :P /b On Sep 21, 2008, at 2:03 PM, unknown wrote:
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
Brian, How do you go about configuring the 7960 to FreeSWITCH? I've got the same problem where the Contact field in the SIP header is set to port 5060 but the NAT is assigning an alternative port. There doesn't seem to be any stun or rport support on the phones. I can set the external IP, just not the port. So when FS responds it always goes to the wrong place. Thanks, David On Sep 21, 2008, at 1:03 PM, unknown wrote:
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> wrote: 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:
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
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participants (4)
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Brian West
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Darren Schreiber
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David Aldworth
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unknown