3.1.3c does not support Polycom 335's and SipXEcs doesn't support differing firmware levels on different phones (the configuration templates are fixed, in 4.1 they are fixed at 3.2.2).
The messages w/o the authentication are coming from the sipxbridge process, not the phones, can you give me a pointer as to why you think the phone's are not sending the authentication? As far as I can tell from the generated configurations the ITSP authentication data isn't put on the phones at all, its all in the bridge. Thanks for any tips, -Eric On Jan 4, 2010, at 2:31 AM, Tony Graziano wrote: > There are known issues with polycom 3.2.x firmware at this time. I'd suggest > using 3.1.3RevC. > > On Sun, Jan 3, 2010 at 9:56 PM, Eric Varsanyi <[email protected]> wrote: > I've followed the setup as closely as possible from here: > http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration > > I am registering successfully with voip.ms and I can receive inbound calls > w/o issue. I have a linux firewall (iptables, using shorewall) set to forward > my external port 5080 to my internal port 5080 on the sipx test box. > > I can use 'Telephone' (the google SIP client) from behind my NAT (directly, > without involving sipxecs) w/o issue to make outbound calls. > > I'll attach the packets in question below, but the basic issue is all my > outbound calls end up getting challenged with a '407' response from voip.ms > and then I get a fast busy on the inside extension (a polycom 650). I don't > understand how to configure the 'local' part of sipxbridge and it seems > 'wrong' that its talking about port 5090 which I haven't configured anywhere. > Its not shown in the traces below but the un-natted packets arriving at the > firewall are coming from the machine running sipxbridge, not from the polycom > phone. > > I'm running a recent (1/3) svn checkout of sipxecs with polycom 3.2.2 > firmware. > > I need some hints or pointers as to where to start digging on this. It seems > like I need a way to make sipxbridge send the authentication info with the > INVITE even though I'm registering. I've configured voip.ms to expect > authentication rather than a static IP (since I don't have a static IP in > this test rig). > > Thanks for any clues, > -Eric Varsanyi > > Here's the sipxbridge.xml (I don't undertand the 5090 part at all, I can't > find it anywhere in the GUI). I have the 'NAT Traversal' stuff set up to use > STUN (and I see it contacting the external stun server and getting back a > reasonable looking BindingResponse packets coming back from it with my > external IP. > > <?xml version="1.0" ?> > <sipxbridge-config > xmlns="http://www.sipfoundry.org/sipX/schema/xml/sipxbridge-00-00";> > > <bridge-configuration> > <global-port>5080</global-port> > <external-address>192.172.252.163</external-address> > <external-port>5080</external-port> > <local-address>192.172.252.163</local-address> > <local-port>5090</local-port> > <sipx-proxy-domain>d3.foo21.com</sipx-proxy-domain> > <sipx-supervisor-host>d3.foo21.com</sipx-supervisor-host> > <sipx-supervisor-xml-rpc-port>8092</sipx-supervisor-xml-rpc-port> > <stun-server-address>stun01.sipphone.com</stun-server-address> > <sip-keepalive-seconds>20</sip-keepalive-seconds> > <media-keepalive-seconds>1</media-keepalive-seconds> > <xml-rpc-port>8088</xml-rpc-port> > <call-limit>-1</call-limit> > <music-on-hold-support-enabled>true</music-on-hold-support-enabled> > <music-on-hold-address>~~mh~</music-on-hold-address> > <music-on-hold-delay-miliseconds>500</music-on-hold-delay-miliseconds> > <music-on-hold-supported-codecs>PCMU,PCMA</music-on-hold-supported-codecs> > > <route-inbound-calls-to-extension>operator</route-inbound-calls-to-extension> > <log-level>INFO</log-level> > <log-directory>/etc/sipxpbx/sip1/INSTALL/var/log/sipxpbx/</log-directory> > <location-id>1</location-id> > </bridge-configuration> > > <itsp-account> > <itsp-proxy-domain>voip.ms</itsp-proxy-domain> > <user-name>NNNNNN_sipx</user-name> > <password>XXXXXX</password> > <register-on-initialization>true</register-on-initialization> > <itsp-proxy-address>74.54.54.178</itsp-proxy-address> > <use-global-addressing>true</use-global-addressing> > <strip-private-headers>false</strip-private-headers> > <default-asserted-identity>true</default-asserted-identity> > <is-user-phone>true</is-user-phone> > <loose-route-invite>true</loose-route-invite> > <registration-interval>600</registration-interval> > > <sip-session-timer-interval-seconds>1800</sip-session-timer-interval-seconds> > <sip-keepalive-method>CR-LF</sip-keepalive-method> > <rtp-keepalive-method>NONE</rtp-keepalive-method> > </itsp-account> > > </sipxbridge-config> > > > This is the outside (post firewall natting) version of the INVITE to voip.ms: > > No. Time Source Destination Protocol Info > 94 0.000111 24.118.160.78 74.54.54.178 SIP/SDP > Request: INVITE sip:[email protected];user=phone, with session > description > > Frame 94 (1129 bytes on wire, 1129 bytes captured) > [Protocols in frame: eth:ip:udp:sip:sdp] > > Internet Protocol, Src: 24.118.160.78 (24.118.160.78), Dst: 74.54.54.178 > (74.54.54.178) > Version: 4 > Header length: 20 bytes > Source: 24.118.160.78 (24.118.160.78) > Destination: 74.54.54.178 (74.54.54.178) > > User Datagram Protocol, Src Port: onscreen (5080), Dst Port: sip (5060) > Source port: onscreen (5080) > Destination port: sip (5060) > Length: 1095 > > Session Initiation Protocol > Request-Line: INVITE sip:[email protected];user=phone SIP/2.0 > Method: INVITE > Request-URI: sip:[email protected];user=phone > Request-URI User Part: 17635191497 > Request-URI Host Part: 74.54.54.178 > [Resent Packet: False] > Message Header > Call-ID: [email protected] > CSeq: 1 INVITE > Sequence Number: 1 > Method: INVITE > From: "sipxbridge" > <sip:[email protected]>;tag=4650615181087041906 > SIP Display info: "sipxbridge" > SIP from address: sip:[email protected] > SIP from address User Part: 7632102101 > SIP from address Host Part: 24.118.160.78 > SIP tag: 4650615181087041906 > To: <sip:[email protected];user=phone> > SIP to address: sip:[email protected] > SIP to address User Part: 17635191497 > SIP to address Host Part: 74.54.54.178 > Via: SIP/2.0/UDP > 24.118.160.78:5080;branch=z9hG4bK886c16e318f58e96df2a31bae2bce938383733 > Transport: UDP > Sent-by Address: 24.118.160.78 > Sent-by port: 5080 > Branch: z9hG4bK886c16e318f58e96df2a31bae2bce938383733 > Max-Forwards: 70 > User-Agent: sipXecs/4.1.0 sipXecs/sipxbridge (Linux) > Contact: <sip:[email protected]:5080;transport=udp> > Contact Binding: <sip:[email protected]:5080;transport=udp> > URI: <sip:[email protected]:5080;transport=udp> > SIP contact address: sip:[email protected]:5080 > Route: <sip:74.54.54.178:5060;transport=udp;lr> > Session-Expires: 1800;refresher=uac > References: [email protected];rel=chain > [Expert Info (Note/Undecoded): Unrecognised SIP header > (References)] > [Message: Unrecognised SIP header (References)] > [Severity level: Note] > [Group: Undecoded] > Allow: INVITE,BYE,ACK,CANCEL,OPTIONS > Content-Type: application/sdp > Content-Length: 376 > Message Body > Session Description Protocol > ... > > This is the response from voip.ms: > > No. Time Source Destination Protocol Info > 98 -0.000020 74.54.54.178 192.172.252.163 SIP > Status: 407 Proxy Authentication Required > > Frame 98 (630 bytes on wire, 630 bytes captured) > Internet Protocol, Src: 74.54.54.178 (74.54.54.178), Dst: 192.172.252.163 > (192.172.252.163) > Source: 74.54.54.178 (74.54.54.178) > Destination: 192.172.252.163 (192.172.252.163) > User Datagram Protocol, Src Port: sip (5060), Dst Port: onscreen (5080) > Source port: sip (5060) > Destination port: onscreen (5080) > Length: 596 > Session Initiation Protocol > Status-Line: SIP/2.0 407 Proxy Authentication Required > Status-Code: 407 > [Resent Packet: False] > [Request Frame: 93] > [Response Time (ms): 52] > Message Header > Via: SIP/2.0/UDP > 192.172.252.163:5080;branch=z9hG4bK886c16e318f58e96df2a31bae2bce938383733;received=192.172.252.163 > Transport: UDP > Sent-by Address: 192.172.252.163 > Sent-by port: 5080 > Branch: z9hG4bK886c16e318f58e96df2a31bae2bce938383733 > Received: 192.172.252.163 > From: "sipxbridge" > <sip:[email protected]>;tag=4650615181087041906 > SIP Display info: "sipxbridge" > SIP from address: sip:[email protected] > SIP from address User Part: 7632102101 > SIP from address Host Part: 24.118.160.78 > SIP tag: 4650615181087041906 > To: <sip:[email protected];user=phone>;tag=as382d7e55 > SIP to address: sip:[email protected] > SIP to address User Part: 17635191497 > SIP to address Host Part: 74.54.54.178 > SIP tag: as382d7e55 > Call-ID: [email protected] > CSeq: 1 INVITE > Sequence Number: 1 > Method: INVITE > User-Agent: VoIPMS SERAST > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Proxy-Authenticate: Digest algorithm=MD5, realm="sip.us2.voip.ms", > nonce="7121ad02" > Authentication Scheme: Digest > Algorithm: MD5 > Realm: "sip.us2.voip.ms" > Nonce Value: "7121ad02" > Content-Length: 0 > > ... and here is our ACK back to them. This is the last thing sent outbound. > > No. Time Source Destination Protocol Info > 100 0.000052 24.118.160.78 74.54.54.178 SIP > Request: ACK sip:[email protected];user=phone > > Frame 100 (530 bytes on wire, 530 bytes captured) > Internet Protocol, Src: 24.118.160.78 (24.118.160.78), Dst: 74.54.54.178 > (74.54.54.178) > Source: 24.118.160.78 (24.118.160.78) > Destination: 74.54.54.178 (74.54.54.178) > User Datagram Protocol, Src Port: onscreen (5080), Dst Port: sip (5060) > Source port: onscreen (5080) > Destination port: sip (5060) > Length: 496 > Checksum: 0x4ac0 [validation disabled] > [Good Checksum: False] > [Bad Checksum: False] > Session Initiation Protocol > Request-Line: ACK sip:[email protected];user=phone SIP/2.0 > Method: ACK > Request-URI: sip:[email protected];user=phone > Request-URI User Part: 17635191497 > Request-URI Host Part: 74.54.54.178 > [Resent Packet: False] > [Request Frame: 94] > [Response Time (ms): 57] > Message Header > Call-ID: [email protected] > Max-Forwards: 70 > From: "sipxbridge" > <sip:[email protected]>;tag=4650615181087041906 > SIP Display info: "sipxbridge" > SIP from address: sip:[email protected] > SIP from address User Part: 7632102101 > SIP from address Host Part: 24.118.160.78 > SIP tag: 4650615181087041906 > To: <sip:[email protected];user=phone>;tag=as382d7e55 > SIP to address: sip:[email protected] > SIP to address User Part: 17635191497 > SIP to address Host Part: 74.54.54.178 > SIP tag: as382d7e55 > Via: SIP/2.0/UDP > 24.118.160.78:5080;branch=z9hG4bK886c16e318f58e96df2a31bae2bce938383733 > Transport: UDP > Sent-by Address: 24.118.160.78 > Sent-by port: 5080 > Branch: z9hG4bK886c16e318f58e96df2a31bae2bce938383733 > CSeq: 1 ACK > Sequence Number: 1 > Method: ACK > Route: <sip:74.54.54.178:5060;transport=udp;lr> > User-Agent: sipXecs/4.1.0 sipXecs/sipxbridge (Linux) > Content-Length: 0 > > > > _______________________________________________ > sipx-users mailing list [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > Why do mathematicians always confuse Halloween and Christmas? > Because 31 Oct = 25 Dec. >
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