Hi everyone,
let me start off telling you where I am stuck right now and then describe how I got to this point.
Basically, I have a brand new Cisco 7970 IP Phone and am trying to register it with an Ubuntu Server 7.10 running the newest stable Asterisk(1.4) and FreePBX(2.3.1).
I have been basically following the instructions to set up the Ubuntu machine on these two FreePBX walkthrough web sites :
blog.thegoldfish.net/asterisk-wi … -tutorial/
aussievoip.com/wiki/freePBX-Ubuntu
Then, I went through the video tutorial at AsteriskTutorials on how to register a 7970 with FreePBX at :
asterisktutorials.com/showpr … oductID=10
I did exactly as shown, and used the extension 555 in FreePBX for this new SIP device.
So I was able to get hold of the most up-to-date SIP firmware for the 7970 from the Cisco web site, namely version SIP70.8-3-3SR2S .
I have also built the three necessary XML files needed to configure this particular 7970 phone, and they are here :
SEP001A6B69ABA4.cnf.xml : cid-ff3ef0764138e401.skydrive.li … A4.cnf.xml
dialplan.xml : cid-ff3ef0764138e401.skydrive.li … alplan.xml
XMLDefault.cnf.xml : cid-ff3ef0764138e401.skydrive.li … lt.cnf.xml
I have gone on to set up a TFTP server daemon on the Ubuntu server and have dumped the above three XML files as well as all the firmware files into the TFTP folder. To test if the TFTP server works, I tried to grab files using the Windows DOS tftp client command and it works.
So, now, it is time to register the 7970 phone. Since I am not sure if everything will work right away, I decided to the Cisco 7970 softphone emulator(newest Cisco IP Communicator software) on my computer to try it out first before actually using a real hard phone. (Note, that SEP001A6B69ABA4.cnf.xml contains the MAC address for the IP Communicator software) I go ahead and set the Alternate TFTP server IP addresses to that of the Asterisk server, and rebooted the phone.
This is where I am stuck. The phone shows what you see in this screenshot :
cid-ff3ef0764138e401.skydrive.li … enshot.JPG
It stays in Registration forever.
I go to the Asterisk command line and typed “show sip peers” and I see :
Name/username Host Dyn Nat ACL Port Status
555 (Unspecified) D N 0 UNKNOWN
1 sip peers [Monitored: 0 online, 1 offline Unmonitored: 0 online, 0 offline]
and “show sip users” gives me :
Username Secret Accountcode Def.Context ACL NAT
555 555 from-internal No Always
So on the phone (IP Communicator), I got to Settings --> Status --> Status Message and this is what I see :
cid-ff3ef0764138e401.skydrive.li … enshot.JPG
I am concerned why there is the line “Error Verifying Config Info” and also that long “TFTP Error : SK5…” line, which I don’t understand.
So, I try to check the Error Trace File of IP Communicator , which I put up here :
cid-ff3ef0764138e401.skydrive.li … ceFile.txt (save to local and then open for easier read)
From the file, it is unclear what “Error Verifying Config Info” means, but the tell-tale section is here :
Tue Nov 27 17:53:19.109 : SPCL : ( 8608) tftpRead : read tftp://172.19.125.41/SEP001A6B69ABA4.cnf.xml, C:\DOCUME~1\iayap\APPLIC~1\Cisco\COMMUN~1\cache\SEP001A6B69ABA4.cnf.xml
Tue Nov 27 17:53:19.140 : SPCL : ( 8608) tftpRead : read returned 0, Ok
Tue Nov 27 17:53:19.140 : DET : ( 8044) -VM| Thread-4|cip.cfg.g:? - SEP001A6B69ABA4.cnf.xml
Tue Nov 27 17:53:19.140 : EE : ( 8044) Thread-4|cip.cfg.g:? - SEP001A6B69ABA4.cnf.xml
Tue Nov 27 17:53:19.156 : ERROR : ( 9244) XMLParseXmlFile return error
Tue Nov 27 17:53:19.156 : DET : ( 9244) trying to set sec mode to NONE (currently NONE)
Tue Nov 27 17:53:19.156 : DET : ( 9244) sec mode set to NONE (was NONE)
Tue Nov 27 17:53:19.156 : DET : ( 8044) -VM| Thread-4|cip.cfg.g:? - Error Verifying Config Info
Tue Nov 27 17:53:19.156 : EE : ( 8044) Thread-4|cip.cfg.g:? - Error Verifying Config Info
So it looks like the XML Parser somehow did not like what it saw in the SEP***.xml file, but doesn’t say what it is.
FINALLY, I go to the CLI command prompt of Asterisk to do “sip set debug” and I see these messages when the phone tries to register :
<— SIP read from 171.71.227.77:2388 —>
REGISTER sip:172.19.125.41 SIP/2.0
Via: SIP/2.0/UDP 171.71.227.77:5060;branch=z9hG4bK00002fed
From: sip:555@172.19.125.41;tag=001a6b69aba4000200003d35-00003373
To: sip:555@172.19.125.41
Call-ID: 001a6b69-aba40002-00004d20-0000046f@171.71.227.77
Max-Forwards: 70
Date: Sun, 00 Jan 1900 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-SIPIPCommunicator/8.0
Contact: sip:7b452e87-4496-4762-e11f-b26751a1884b@171.71.227.77:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-001a6b69aba4”;+u.sip!model.ccm.cisco.com=“30016”;+u.sip!devicename.ccm.features.cisco.com="SEP001A6B69ABA4"
Content-Length: 0
Expires: 3600
<------------->
— (12 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 171.71.227.77 : 5060 (no NAT)
<— Transmitting (NAT) to 171.71.227.77:2388 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 171.71.227.77:5060;branch=z9hG4bK00002fed;received=171.71.227.77
From: sip:555@172.19.125.41;tag=001a6b69aba4000200003d35-00003373
To: sip:555@172.19.125.41
Call-ID: 001a6b69-aba40002-00004d20-0000046f@171.71.227.77
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:555@172.19.125.41
Content-Length: 0
<------------>
<— Transmitting (NAT) to 171.71.227.77:2388 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 171.71.227.77:5060;branch=z9hG4bK00002fed;received=171.71.227.77
From: sip:555@172.19.125.41;tag=001a6b69aba4000200003d35-00003373
To: sip:555@172.19.125.41;tag=as4282d10c
Call-ID: 001a6b69-aba40002-00004d20-0000046f@171.71.227.77
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="2a52efa4"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘001a6b69-aba40002-00004d20-0000046f@171.71.227.77’ in 32000 ms (Method: REGISTER)
Really destroying SIP dialog ‘001a6b69-aba40002-00004d20-0000046f@171.71.227.77’ Method: REGISTER
Can someone please help? I can provide more info to this issue if necessary.
Thanks
John
