I just checked the SIP debug when my 7960 registers and it looks like NAT is enabled and working properly.
Does anyone have a 7961 on Asterisk that is going through NAT successfully? <-- SIP read from <HOME IP ADDRESS>:5061: REGISTER sip:<TB IP ADDRESS> SIP/2.0 Via: SIP/2.0/UDP <HOME IP ADDRESS>:5061;branch=z9hG4bK740d9e78 From: <sip:860001@<TB IP ADDRESS>>;tag=001a6dd2f84c00195c3209da-0ece5aea To: <sip:860001@<TB IP ADDRESS>> Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 Date: Fri, 25 Jan 2008 20:20:26 GMT CSeq: 116 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: <sip:860001@<HOME IP ADDRESS>:5061;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-000 0-001a6dd2f84c>";+u.sip!model.ccm.cisco.com="7" Authorization: Digest username="860001",realm="asterisk",uri="sip:<TB IP ADDRESS>",response="d2b6c69bf9ba5ee5ff808dea90963b64",nonce="5be57786",algor ithm=MD5 Content-Length: 0 Expires: 60 --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to <HOME IP ADDRESS> : 5061 (NAT) Transmitting (NAT) to <HOME IP ADDRESS>:5061: SIP/2.0 100 Trying Via: SIP/2.0/UDP <HOME IP ADDRESS>:5061;branch=z9hG4bK740d9e78;received=<HOME IP ADDRESS> From: <sip:860001@<TB IP ADDRESS>>;tag=001a6dd2f84c00195c3209da-0ece5aea To: <sip:860001@<TB IP ADDRESS>> Call-ID: [EMAIL PROTECTED] CSeq: 116 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:860001@<TB IP ADDRESS>> Content-Length: 0 --- Transmitting (NAT) to <HOME IP ADDRESS>:5061: SIP/2.0 200 OK Via: SIP/2.0/UDP <HOME IP ADDRESS>:5061;branch=z9hG4bK740d9e78;received=<HOME IP ADDRESS> From: <sip:860001@<TB IP ADDRESS>>;tag=001a6dd2f84c00195c3209da-0ece5aea To: <sip:860001@<TB IP ADDRESS>>;tag=as77362809 Call-ID: [EMAIL PROTECTED] CSeq: 116 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 60 Contact: <sip:860001@<HOME IP ADDRESS>:5061;transport=udp>;expires=60 Date: Fri, 25 Jan 2008 20:20:26 GMT Content-Length: 0 On 1/25/08 2:54 PM, "Gregory Wong" <[EMAIL PROTECTED]> wrote: > Thanks Chad. This config seemed to have worked a bit. I don't get the > "Unprovisioned" or "Error Verifying Config Info" messages anymore. However, > the phone sits at "Registering" and will never register. > > I took a look at the sip debug and I see the below messages. Do I need to > enable NAT in the SEP.cnf.xml file since I am behind NAT? I know my 7960 > config file has natEnabled = 1. > > Scheduling destruction of call > '[EMAIL PROTECTED]' in 15000 ms > > <-- SIP read from <MY HOME IP ADDRESS>:49157: > REGISTER sip:<TB IP ADDRESS> SIP/2.0 > Via: SIP/2.0/UDP <MY HOME IP ADDRESS>:1140;branch=z9hG4bK48e89c16 > From: <sip:86003@<TB IP ADDRESS>>;tag=0018195aa6770003efaf5095-54a486b0 > To: <sip:86003@<TB IP ADDRESS>> > Call-ID: [EMAIL PROTECTED] > Max-Forwards: 70 > Date: Mon, 08 Oct 2007 23:42:08 GMT > CSeq: 101 REGISTER > User-Agent: Cisco-CP7961G/8.3.0 > Contact: <sip:86003@<HOME IP > ADDRESS>:1140;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-000 > 0-0018195aa677>";+u.sip!model.ccm.cisco.com="30018" > Supported: (null),X-cisco-xsi-6.0.2 > Content-Length: 0 > Reason: SIP;cause=200;text="cisco-alarm:25 Name=SEP0018195AA677 > Load=SIP41.8-3-3SR2S Last=initialized" > Expires: 3600 > > > --- (14 headers 0 lines) --- > Using latest REGISTER request as basis request > Sending to <HOME IP ADDRESS> : 1140 (non-NAT) > Transmitting (no NAT) to <HOME IP ADDRESS>:1140: > SIP/2.0 404 Not found > Via: SIP/2.0/UDP <HOME IP > ADDRESS>:1140;branch=z9hG4bK48e89c16;received=<HOME IP ADDRESS> > From: <sip:86003@<TB IP ADDRESS>>;tag=0018195aa6770003efaf5095-54a486b0 > To: <sip:86003@<TB IP ADDRESS>>;tag=as1886ecd1 > Call-ID: [EMAIL PROTECTED] > CSeq: 101 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > > On 1/25/08 10:29 AM, "Chad Osmond" <[EMAIL PROTECTED]> wrote: > >> Try this configuration file... >> >> http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configu >> ration+files+for+SIP#Cisco7961with833SR2ConfigurationExamples >> >> >> Chad >> ________________________________ >> >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of Gregory >> Wong >> Sent: Friday, January 25, 2008 6:36 AM >> To: [email protected] >> Subject: [asterisk-users] Unprovisioned 7961 >> >> >> Hi Everyone, >> >> I am having some issues getting my 7961 working with Trixbox. I have >> loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes >> into an unprovisioned state. A status message shows up and says "Error >> Verifying Config Info". >> >> I have read quite a bit on this topic (getting 7961's to work with >> Asterisk and TB) and only came across a few postings where other people >> encountered this issue but no solution was given. I have checked the >> SEP.cnf.xml file for the phone and everything seems to be right. I even >> tried to remove some parts of the code as people suggested but no luck. >> I already have a 7960 on TB so I know that TFTP is working correctly. >> >> Any ideas on how I can get this to work would be much appreciated. >> >> Thank. >> ______________________________________________________________________ >> This email has been scanned by the MessageLabs Email Security System. >> For more information please visit http://www.messagelabs.com/email >> ______________________________________________________________________ >> >> >> ______________________________________________________________________ >> This email has been scanned by the MessageLabs Email Security System. >> For more information please visit http://www.messagelabs.com/email >> ______________________________________________________________________ >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
