Cisco Voip Phone (7940G)

VOIP setup and troubleshooting
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swherdman
Posts: 7
Joined: Sat May 09, 2009 10:46 pm
Location: Sydney

Cisco Voip Phone (7940G)

Post by swherdman » Thu May 28, 2009 12:00 am

Anyone had any luck getting these working, its running the latest SIP firmware

SIP000BBEE396F1.cnf

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# SIP Configuration Generic File (start)

image_version:P0S3-08-11-00;

# Line 1 Settings
line1_name: "Exetel"                     ; Line 1 Extension\User ID
line1_displayname: "028006xxx"           ; Line 1 Display Name
line1_shortname: "028006xxx" 
line1_authname: "028006xxx"         ; Line 1 Registration Authentication
line1_password: "xxxxxx"         ; Line 1 Registration Password
line1_description "Scott Herdman"	;
proxy1_port: 5060	;
proxy1_address: "58.96.1.2";
proxy_register: 1;

# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Config0067"            ; Has no effect on SIP messaging

# Time Zone phone will reside in
time_zone: PST; 

# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "2"      ; 0-Disabled (default), 1-Enabled, 2-Privileged

# Phone prompt/password for telnet/console session
phone_prompt: "cisco_phone"                              ; Telnet/Console Prompt
phone_password: "123"                          ; Telnet/Console Password

# Enable_VAD (1-enabled, 0-disabled)
enable_vad: "0"

# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
user_info: phone

# URL for external Directory location
#logo_url: "http://10.0.1.3/10-20logo.bmp"                    ; URL for branding logo to be used on phone display
logo_url: "http://www.loligo.com/asterisk/Cisco/79xx/current/asterisk-tux.bmp" ;


sntp_mode: unicast
sntp_server: "192.168.1.200"


# SIP Configuration Generic File (stop)

SIP000BBEE396F1.cnf.xml

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<device>
<loadInformation model="IP Phone 7940">P0S3-08-11-00</loadInformation>
</device>
SIPDefault.cnf

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image_version: P0S3-08-11-00
proxy1_address: "58.96.1.2"            ; Can be dotted IP or FQDN
proxy2_address: ""              ; Can be dotted IP or FQDN
proxy3_address: ""              ; Can be dotted IP or FQDN
proxy4_address: ""              ; Can be dotted IP or FQDN
proxy5_address: ""              ; Can be dotted IP or FQDN
proxy6_address: ""              ; Can be dotted IP or FQDN
proxy_register: 1
messages_uri:   "1"
phone_password: "1234" ; Limited to 31 characters (Default - cisco)
#sntp_mode: unicast
#sntp_server: "192.168.254.254"
time_zone: "GMT" ; assuming you're in GMT
time_format_24hr: 1 ; to show the time in 24hour format
date_format: "Y/M/D"  ; format you would like the date in
dial_template: dialplan
OS79XX.TXT

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P003-08-11-00 
XMLDefault.cnf.xml (with a Hard link to xmlDefault.CNF.XML

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<Default>
  <callManagerGroup>
     <members>
        <member priority="0">
           <callManager>
              <ports>
                 <ethernetPhonePort>2000</ethernetPhonePort>
                 <mgcpPorts>
                    <listen>2427</listen>
                    <keepAlive>2428</keepAlive>
                 </mgcpPorts>
              </ports>
              <processNodeName></processNodeName>
           </callManager>
        </member>
     </members>
  </callManagerGroup>
  <loadInformation7  model="Cisco 7940">P0S3-08-11-00</loadInformation7>
<authenticationURL></authenticationURL>
 <directoryURL></directoryURL>
 <idleURL></idleURL>
 <informationURL></informationURL>
 <messagesURL></messagesURL>
 <servicesURL></servicesURL>
</Default>


JasonM

Re: Cisco Voip Phone (7940G)

Post by JasonM » Thu May 28, 2009 9:25 am

What happens when you make a call?
Does the device indicate it has registered with Exetel successfully?

swherdman
Posts: 7
Joined: Sat May 09, 2009 10:46 pm
Location: Sydney

Re: Cisco Voip Phone (7940G)

Post by swherdman » Thu May 28, 2009 9:34 am

no just sits there for a min or so with "Calling (out INV)" on the screen then it changes to "proxy unavailable, switching t" then beeps an enaged like tone with "Reorder" on the screen

has network connectivity as it manged to pull down the wallpaper/logo for the phone

Code: Select all

logo_url: "http://www.loligo.com/asterisk/Cisco/79xx/current/asterisk-tux.bmp" ;

JasonM

Re: Cisco Voip Phone (7940G)

Post by JasonM » Thu May 28, 2009 9:47 am

Change proxy to sip1.exetel.com.au - does that work?
PM the VoIP DID to me, I'll check if it's registering.

Munka
Posts: 289
Joined: Sat Oct 22, 2005 8:22 pm
Location: Rural NSW

Re: Cisco Voip Phone (7940G)

Post by Munka » Thu May 28, 2009 10:40 am

That is a great sip enabled phone, though mine seems to have been bricked somehow, something I notice though are the config sheets ending in .xml are meant to be renamed to xxxxxxx.cnf dropping the .xml
And if the TFTP server is running on a windows machine the zip file should be extracted directly to the TFTP root folder and renamed there, overcoming the windows issue when changing to files to .cnf

Hope thats clear, I maybe telling you things you are already aware of , but I notice in your config examples they still end with .xml :)
Munka

swherdman
Posts: 7
Joined: Sat May 09, 2009 10:46 pm
Location: Sydney

Re: Cisco Voip Phone (7940G)

Post by swherdman » Thu May 28, 2009 8:21 pm

Changed

Code: Select all

cat SIPDefault.cnf SIP000BBEE396F1.cnf | grep proxy
proxy1_address: "sip1.exetel.com.au"            ; Can be dotted IP or FQDN
proxy2_address: ""              ; Can be dotted IP or FQDN
proxy3_address: ""              ; Can be dotted IP or FQDN
proxy4_address: ""              ; Can be dotted IP or FQDN
proxy5_address: ""              ; Can be dotted IP or FQDN
proxy6_address: ""              ; Can be dotted IP or FQDN
proxy_register: 1
proxy1_port: 5060	;
proxy1_address: "sip1.exetel.com.au;
proxy_register: 1;
Yes indeed a good phone little complex to setup, so if anyone is reading this and thinking of one, to do it properly you need everything listed below
  • You need a Cisco account that allows restricted software downloads to get the latest firmware (or find it online)
  • A TFTP server running somewhere, Mac, windows, linux, (windows users i recomend the solar winds one)
  • DHCP server that allows you to specify the option of passing the client the TFTP servers IP during the lease
  • A spare hour or 3 and not being afraid to dig though a few config files
done it on with both windows servers and linux servers, no idea about Mac's sure is possable just dont know the apps off the top of my head

the additional.xml file seems to be optional, most places dont mention it but i built it and thew it in just in case,

JamesR
Posts: 424
Joined: Sun May 06, 2007 10:20 am

Re: Cisco Voip Phone (7940G)

Post by JamesR » Fri May 29, 2009 5:12 pm

In both of the examples of configuration you use above are either using GMT or PST as the time zone... you might want to change this ;)
Regards,

JamesR
Customer since 2005

dengdeng
Posts: 6
Joined: Fri Feb 15, 2008 3:52 pm
Location: NSW

Re: Cisco Voip Phone (7940G)

Post by dengdeng » Wed Jul 01, 2009 2:16 pm

Hi swherdman,

I'm having exactly the same problem as you were. Very frustrating, it's been few days couldn't figure out what went wrong.

Sorry, as a newbie and I don't quite understand the post below.
Could you or anyone please give me some more info, which file did you change to fix the issue?

Thanks in advance.
swherdman wrote:Changed

Code: Select all

cat SIPDefault.cnf SIP000BBEE396F1.cnf | grep proxy
proxy1_address: "sip1.exetel.com.au"            ; Can be dotted IP or FQDN
proxy2_address: ""              ; Can be dotted IP or FQDN
proxy3_address: ""              ; Can be dotted IP or FQDN
proxy4_address: ""              ; Can be dotted IP or FQDN
proxy5_address: ""              ; Can be dotted IP or FQDN
proxy6_address: ""              ; Can be dotted IP or FQDN
proxy_register: 1
proxy1_port: 5060	;
proxy1_address: "sip1.exetel.com.au;
proxy_register: 1;

swherdman
Posts: 7
Joined: Sat May 09, 2009 10:46 pm
Location: Sydney

Re: Cisco Voip Phone (7940G)

Post by swherdman » Wed Jan 06, 2010 3:16 am

Never did get this working, just broken it out again, found some more doco this time, plus i worked out how to get debugging on, full debug of all SIP related "stuff" when attempting to make a test call

Code: Select all

cisco_phone> [03:51:47:15506] SIPTaskProcessListEvent: cmd = 0x161700
[03:51:47:15506] sip_cc_event LINE 0/1: --0x0004ea09--                     : SIP_STATE_IDLE <- E_CC_SETUP
[03:51:47:15507] idle_ev_cc_setup: All digits collected.  Placing the call
[03:51:47:15508] SIPSM 0/2/2: idle_ev_cc_setup                   : Setup
[03:51:47:15508] SIPSPISendInvite: Sending INVITE...
[03:51:47:15515] sipTransportSendMessage: ccb <0>: config <58.96.1.2>:<5060> - remote <58.96.1.2>:<5060>
[03:51:47:15515] sipTransportSendMessage: Got handle 3
[03:51:47:15516] sipTransportSendMessage: Opened a one-time UDP send channel to server <58.96.1.2>:<5060>, handle = 8 local port= 5060
[03:51:47:15517] sipTransportSendMessage:Sent SIP message to <58.96.1.2>:<5060>, handle=<8>, length=<1060>, message=
[03:51:47:15517] INVITE sip:0423712xxx@sip1.exetel.com.au SIP/2.0
Via: SIP/2.0/UDP 58.96.1.2:5060;branch=z9hG4bK28de94d4
From: "new_exetel" <sip:asterix@sip1.exetel.com.au>;tag=000bbee396f100065c8c03cc-3a685e4c
To: <sip:0423712xxx@sip1.exetel.com.au>
Call-ID: 000bbee3-96f10006-7a12ce38-6c787dbd@58.96.1.2
Max-Forwards: 70
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:asterix@58.96.1.2:5060;user=phone;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "new_exetel" <sip:asterix@sip1.exetel.com.au>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 270
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 10346 0 IN IP4 58.96.1.2
s=SIP Call
t=0 0
m=audio 31806 RTP/AVP 0 8 18 101
c=IN IP4 58.96.1.2
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
[03:51:47:15520] sipTransportSendMessage: Closed a one-time UDP send channel handle = 8
[03:51:47:15520] LINE 0/2: sipTransportSendMessage            : Stopping reTx timer
[03:51:47:15521] LINE 0/2: sipTransportSendMessage            : Starting reTx timer (500 msec)
[03:51:47:15522] CHANGE STATE: LINE 0/2:                                    : State change: SIP_STATE_IDLE -> SIP_STATE_SENT_INVITE
[03:51:47:15522] SIPTaskProcessListEvent: cmd = 0x160200
[03:51:47:15523] SIPProcessUDPMessage: recv UDP message from <58.96.1.2>:<50195>, length=<529>, message=
[03:51:47:15524] SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK28de94d4;rport=5060
From: "new_exetel" <sip:asterix@sip1.exetel.com.au>;tag=000bbee396f100065c8c03cc-3a685e4c
To: <sip:0423712xxx@sip1.exetel.com.au>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8556
Call-ID: 000bbee3-96f10006-7a12ce38-6c787dbd@58.96.1.2
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip1.exetel.com.au", nonce="4b4365770e17106b6f5cc38e3f39500963d33676"
Server: OpenSer (1.1.0-tls (i386/linux))
Content-Length: 0

[03:51:47:15526] SIPTaskProcessSIPMessage: Line filter: Determining destination line...
[03:51:47:15527] sip_sm_ccb_match_branch_cseq: Matched branch_id & CSeq
[03:51:47:15528] SIPTaskProcessSIPMessage: Line filter: Call ID match:  Destination line = <0/2>.
[03:51:47:15528] SIPTaskProcessSIPMessage: Received SIP response.
[03:51:47:15530] sipSPICheckResponse: Response match: callid=000bbee3-96f10006-7a12ce38-6c787dbd@58.96.1.2, cseq=101, cseq_method=INVITE
[03:51:47:15530] SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers...
[03:51:47:15532] LINE 0/2: sip_sm_check_retx_timers           : Stopping reTx timer.
(callid=000bbee3-96f10006-7a12ce38-6c787dbd@58.96.1.2, cseq=101, cseq_method=INVITE)
[03:51:47:15534] SIPTaskProcessSIPMessage: Recv 4xx/5xx/6xx message.
[03:51:47:15535] sip_sm_process_event LINE 0/2: --0x000508c1--                     : SIP_STATE_SENT_INVITE <- E_SIP_FAILURE_RESPONSE
[03:51:47:15536] LINE 0/2: SIP 407 Proxy Authentication required
[03:51:47:15538] SIPSPIAddRouteHeaders: Route info not available; will not add Route header.
[03:51:47:15539] sipRelDevCoupledMessageStore: Storing for reTx (cseq=101, method=INVITE, to_tag=<329cfeaa6ded039da25ff8cbb8668bd2.8556>)
[03:51:47:15541] sipTransportSendMessage: Opened a one-time UDP send channel to server <58.96.1.2>:<5060>, handle = 8 local port= 5060
[03:51:47:15542] sipTransportSendMessage:Sent SIP message to <58.96.1.2>:<5060>, handle=<8>, length=<369>, message=
[03:51:47:15542] ACK sip:0423712xxx@sip1.exetel.com.au SIP/2.0
Via: SIP/2.0/UDP 58.96.1.2:5060;branch=z9hG4bK28de94d4
From: "new_exetel" <sip:asterix@sip1.exetel.com.au>;tag=000bbee396f100065c8c03cc-3a685e4c
To: <sip:0423712xxx@sip1.exetel.com.au>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8556
Call-ID: 000bbee3-96f10006-7a12ce38-6c787dbd@58.96.1.2
CSeq: 101 ACK
Content-Length: 0

[03:51:47:15543] sipTransportSendMessage: Closed a one-time UDP send channel handle = 8
[03:51:47:15544] Proxy-Authenticate= Digest realm="sip1.exetel.com.au", nonce="4b4365770e17106b6f5cc38e3f39500963d33676"
[03:51:47:15546] sipSPISendInviteMidCall: Sending INVITE...
[03:51:47:15547] sipSPIGenRequestURI: Forming Req-URI (Caller): using original Req-URI
[03:51:47:15549] SIPSPIAddRouteHeaders: Route info not available; will not add Route header.
[03:51:47:15553] sipTransportSendMessage: ccb <0>: config <58.96.1.2>:<5060> - remote <58.96.1.2>:<5060>
[03:51:47:15554] sipTransportSendMessage: Got handle 3
[03:51:47:15555] sipTransportSendMessage: Opened a one-time UDP send channel to server <58.96.1.2>:<5060>, handle = 8 local port= 5060
[03:51:47:15555] sipTransportSendMessage:Sent SIP message to <58.96.1.2>:<5060>, handle=<8>, length=<1285>, message=
[03:51:47:15556] INVITE sip:0423712xxx@sip1.exetel.com.au SIP/2.0
Via: SIP/2.0/UDP 58.96.1.2:5060;branch=z9hG4bK1b3c7ae0
From: "new_exetel" <sip:asterix@sip1.exetel.com.au>;tag=000bbee396f100065c8c03cc-3a685e4c
To: <sip:0423712xxx@sip1.exetel.com.au>
Call-ID: 000bbee3-96f10006-7a12ce38-6c787dbd@58.96.1.2
Max-Forwards: 70
CSeq: 102 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:asterix@58.96.1.2:5060;user=phone;transport=udp>
Proxy-Authorization: Digest username="0280072xxx",realm="sip1.exetel.com.au",uri="sip:0423712xxx@sip1.exetel.com.au",response="0f88efbe40aff99b62e06a8c42f36c21",nonce="4b4365770e17106b6f5cc38e3f39500963d33676",algorithm=md5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "new_exetel" <sip:asterix@sip1.exetel.com.au>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 270
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 10346 0 IN IP4 58.96.1.2
s=SIP Call
t=0 0
m=audio 31806 RTP/AVP 0 8 18 101
c=IN IP4 58.96.1.2
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
[03:51:47:15559] sipTransportSendMessage: Closed a one-time UDP send channel handle = 8
[03:51:47:15559] LINE 0/2: sipTransportSendMessage            : Stopping reTx timer
[03:51:47:15560] LINE 0/2: sipTransportSendMessage            : Starting reTx timer (500 msec)
[03:51:47:15562] SIPTaskProcessListEvent: cmd = 0x160200
[03:51:47:15562] SIPProcessUDPMessage: recv UDP message from <58.96.1.2>:<50195>, length=<418>, message=
[03:51:47:15563] SIP/2.0 403 Spoofed To-URI detected
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK1b3c7ae0;rport=5060
From: "new_exetel" <sip:asterix@sip1.exetel.com.au>;tag=000bbee396f100065c8c03cc-3a685e4c
To: <sip:0423712xxx@sip1.exetel.com.au>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8d85
Call-ID: 000bbee3-96f10006-7a12ce38-6c787dbd@58.96.1.2
CSeq: 102 INVITE
Server: OpenSer (1.1.0-tls (i386/linux))
Content-Length: 0

[03:51:47:15565] SIPTaskProcessSIPMessage: Line filter: Determining destination line...
[03:51:47:15566] sip_sm_ccb_match_branch_cseq: Matched branch_id & CSeq
[03:51:47:15567] SIPTaskProcessSIPMessage: Line filter: Call ID match:  Destination line = <0/2>.
[03:51:47:15567] SIPTaskProcessSIPMessage: Received SIP response.
[03:51:47:15569] sipSPICheckResponse: Response match: callid=000bbee3-96f10006-7a12ce38-6c787dbd@58.96.1.2, cseq=102, cseq_method=INVITE
[03:51:47:15569] SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers...
[03:51:47:15572] LINE 0/2: sip_sm_check_retx_timers           : Stopping reTx timer.
(callid=000bbee3-96f10006-7a12ce38-6c787dbd@58.96.1.2, cseq=102, cseq_method=INVITE)
[03:51:47:15574] SIPTaskProcessSIPMessage: Recv 4xx/5xx/6xx message.
[03:51:47:15574] sip_sm_process_event LINE 0/2: --0x000508c1--                     : SIP_STATE_SENT_INVITE <- E_SIP_FAILURE_RESPONSE
[03:51:47:15577] SIPSPIAddRouteHeaders: Route info not available; will not add Route header.
[03:51:47:15578] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=INVITE, to_tag=<329cfeaa6ded039da25ff8cbb8668bd2.8d85>)
[03:51:47:15580] sipTransportSendMessage: Opened a one-time UDP send channel to server <58.96.1.2>:<5060>, handle = 8 local port= 5060
[03:51:47:15581] sipTransportSendMessage:Sent SIP message to <58.96.1.2>:<5060>, handle=<8>, length=<369>, message=
[03:51:47:15581] ACK sip:0423712xxx@sip1.exetel.com.au SIP/2.0
Via: SIP/2.0/UDP 58.96.1.2:5060;branch=z9hG4bK1b3c7ae0
From: "new_exetel" <sip:asterix@sip1.exetel.com.au>;tag=000bbee396f100065c8c03cc-3a685e4c
To: <sip:0423712xxx@sip1.exetel.com.au>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8d85
Call-ID: 000bbee3-96f10006-7a12ce38-6c787dbd@58.96.1.2
CSeq: 102 ACK
Content-Length: 0

[03:51:47:15582] sipTransportSendMessage: Closed a one-time UDP send channel handle = 8
[03:51:47:15585] CHANGE STATE: LINE 0/2:                                    : State change: SIP_STATE_SENT_INVITE -> SIP_STATE_RELEASE
[03:51:47:15587] SIPTaskProcessListEvent: cmd = 0x161700
[03:51:47:15587] sip_cc_event LINE 0/2: --0x0004f73d--                     : SIP_STATE_RELEASE <- E_CC_RELEASE_COMPLETE
[03:51:47:15588] LINE 0/2: sip_sm_call_cleanup                : Cleaning up the call
[03:51:47:15589] CHANGE STATE: LINE 0/2:                                    : State change: SIP_STATE_RELEASE -> SIP_STATE_IDLE
[03:51:53:16186] SIPTaskProcessListEvent: cmd = 0x161700
[03:51:53:16186] sip_sm_process_cc_event: No ccb with matching gsm_id = <2>


Munka
Posts: 289
Joined: Sat Oct 22, 2005 8:22 pm
Location: Rural NSW

Re: Cisco Voip Phone (7940G)

Post by Munka » Wed Jan 06, 2010 9:27 am

[quote="swherdman"]Never did get this working, just broken it out again, found some more doco this time, plus i worked out how to get debugging on, full debug of all SIP related "stuff" when attempting to make a test call

Actually, sorry I have to withdraw the offer below as telnet access seems to be turned off and I don't have time ATM to set up a tftp server to change the config to allow it.
Sorry and best of luck.

Funny after you started this thread I "de-bricked" mine and it's been working faultlessly since, if you want to remind me how to cli into it using putty I will run a debug on mine and post the output if you think the comparison would be helpful, though I should point out my 7940G is running stand alone, no asterix server. Let me know.
Munka

swherdman
Posts: 7
Joined: Sat May 09, 2009 10:46 pm
Location: Sydney

Re: Cisco Voip Phone (7940G)

Post by swherdman » Thu Jan 07, 2010 12:16 am

ITS WORKING

well i can make calls but not recieve them, well as yet anyway, if anyone's intrested in details ill just dump you a copy of my tftp root

Munka
Posts: 289
Joined: Sat Oct 22, 2005 8:22 pm
Location: Rural NSW

Re: Cisco Voip Phone (7940G)

Post by Munka » Thu Jan 07, 2010 10:19 am

swherdman wrote:ITS WORKING

well i can make calls but not recieve them, well as yet anyway, if anyone's intrested in details ill just dump you a copy of my tftp root
Thats good news swherdman, from my limited experience with these not being able to receive calls would be a port thing, what ports have you made available to sip?
Munka

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