66 thoughts on “Configuring a Cisco IP phone for VoIP using SIP”

1. After trying unsuccessfully to move a 7960G from SCCP v6.0 to SIP v8.2, it appears a soft reset using **#** is not sufficient and that a hard reset (i.e. unplugging the power cord) is required in order to get the phone to load a new firmware image.

2. File : error 2 in system call CreateFile The system cannot find the file specified. [25/07 00:41:32.672]

   Some possibilities for this error is that access is denied for that file (wrong permissions), a path reference string is wrong in another configuration file or that there is no write access (considering call is called “CreateFile”).

3. Or it could just be that the file doesn’t exist ;-)

   The SEP files should produce an error on a SIP phone but the universal loader will still work through its hunt algorithm until it finds a file that it knows what to do with.

4. Brilliant. Just the sort of information I need to finally nudge me into trying to get a Cisco 7940 working here in the Seattle area on VOIP service. Thanks for letting us learn from your experience.

5. hello, where can i get the SIP files for 7940? thanks

6. @Paul – Follow the link at the bottom of the article labelled VoIP Info: Configuring Cisco 79xx phones with Asterisk. On there you should see firmware information (a few paragraphs down).

7. Hello Mark, thanks for your excellent blog. Just wondering of the security issues of having your IP phone to directly communicate with a SIP provider and having to open all those ports. Maybe its better to install an inhouse IP-PBX system (e.g linksys) and use the PSTN instead.

8. @cisco-tips. Thanks for the compliment!

   I’m not concerned about opening those ports (my phone is on a different subnet to most of my IT and so the ports are closed on a secondary firewall) but it is a valid concern for some people. I also have my 7940 working with the PSTN via a Linksys SPA3102 but there’s something not quite right in the config as it’s randomly dropping calls mid-conversation.

   Another option would be to use something like Asterisk as an in-house PBX although it may still need the same ports open to communicate with the SIP provider.

   Using SIP for calls is certainly saving me money at peak times though (hour long conference calls in the middle of the day, for example) but I still use the PSTN at weekends/evenings (when most of the call charges are included in my line rental).

9. An excellent and useful guide, thanks!

10. Its an excellent and useful guide. just a quick question about cisco phones though. how many sip providers it can connect to?

11. Thanks Mark – as many as you have lines for – at the moment I have my 7940 configured with two providers – Sipgate and Call Centric. If I had a 7960 I could use 4 lines.

12. Sorry I am still confused :) – I have 8 sip providers – means I could connect all of them at the same time if I have 7940 or just the two of them?

    or four If I have 7960?

    Thanks for all your help, I really appreciate.

    Mark Johnson

13. @Mark – 8 SIP providers!!! Wow! As far as I know, you can only configure one provider per line. You have two choices that I can think of:

    1. Get more lines on your handset (I think Cisco do an extension unit of some sort) but that sounds expensive and is still limited.
    2. Set up an IP PBX (e.g. an Asterisk box) and use your phone as an extension on that, with it handling the routing to the various SIP providers.

    (I’m probably going to try option 2 at some point as I have also been trying to get the Cisco phone talking to the PSTN via a LinkSys SPA3102 gateway with some success but also some strange call disconnects…)

14. dude ;) – If you’re thinking of IP PBX, I ‘d suggest trying 3cx – I ‘ve been using it for about a year now without any problems. Its very easy to setup and call quality is flawless. The only reason why I want to get rid of it is because the pc has to be on all the time and its very energy consuming.

    Guess I’ll just buy a SNOM IP phone ;)

    Cheers
    Mark

15. Hi,
    I’m having tales of woe also with my 7940, and trying to convert from SCCP.

    I’ve followed your advice Mark, but am stuck. I’m doing the v6.3 upgrade first, and I’m getting an error from tftpd32…

    “Read request for file . Mode octet [12/03 12:03:06.379]
    File : error 2 in system call CreateFile The system cannot find the file specified. [12/03 12:03:06.379]””

    Despite the fact that THAT file IS in the tftp folder.

    Any ideas?!

16. Argh, ignore me, I messed up the filename in the .TXT file, I missed the last 0 off the line… Damn…

17. @Stewart – glad you fixed it!

18. @Mark – one of my colleagues (who got me into the whole VoIP thing in the first place) has moved over to My SIP Switch to handle his various VoIP accounts so he only uses one line on the 79xx phone for multiple providers.

19. Mark:

    I used mysipswitch for a while and it sucked bigtime. Either the site’s not forwarding calls properly or registration errors. yada yada yda – so i moved to 3cx n now m thinkin – lets switch bak to som SIP based phone to save energy – I think m gona buy gnom’s IP phone with miltiple sip providers option or a dreytek router with 12 sip registrars ;) – its not cheap though:(

20. Pingback: markwilson.it » Customising a Cisco 79xx IP Phone: directory services

21. Hi Mark, Must appreciate your help on the VOIP arena. I have a quick question for you if you can assist me. I have configured a Cisco7940 on Sipgate & I am using firmware 8.4. Can see the phone online & tried the test call to sipgate & it works. I am able to hear the pre-recorded voicemesage. However if I dial my sip number(Cisco7940) from another pc ( through x-lite) logged in via a different sipgate acount. The cisco phone ring but when I pick up, there is no voice both ways. its the same the other way round. So to try & figure this out I set a timeout for voicemail to 10 sec & dial the web account through the cisco phone, it rings my pc & goes to voice mail as expected. I can listen to the message I left through sipgates online account that works without any problem. If this works then how is it that I cannot hear anything when the call is started from Web account to Cisco phone or reverse.

    Please could you advice

22. @Mickey – The best advice I can give is to use something like Wireshark to examine the VoIP traffic and see what is going where… that’s pretty deep stuff though – I imagine there’s some sort of issue preventing the softphone from talking to the hard phone over SIP (maybe firewall restrictions?).

23. Pingback: markwilson.it » Building a low-power server for 24×7 infrastructure at home: Part 1 (hardware selection)

24. I was wondering what settings you’re using to connect to CallCentric. I have a 7960 and it keeps getting stuck in the REGISTERING state.

    I’m behind a cheapo DLink home router.

25. Hi Mark, great writeup. I myself struggled through all the same steps, and blogs like yours got me on my feet. I’ve just upgraded to firmware v. 8-12, and at least a couple of bugs have been ironed out.

    One thing that I just cannot understand (I know it’s slightly off-topic, but I may not be the only one experiencing this) is why the phone will display the caller id, but flatly refuses to match it to the entry in its own directory? Either I’m the only one with this problem (as a search on Google returns nothing), or no one else is bothered with a list of missed (and placed/received) calls, that makes no sense without a name.

    I’d appreciate hearing if anyone else has this problem, or if there is something that they’ve done. Thanks.

26. Thanks for the post Mark. I just got my CCNA and will be going after the voice and wireless speciality certs so this gives me a little information on what is coming ahead for me.

27. Hello,

    Sacha, same problem as yours, can’t display caller id from local directory…
    Any clue ?

    Thank you !

28. I can listen to the message I left through sipgates online account that works without any problem. If this works then how is it that I cannot hear anything when the call is started from Web account to Cisco phone or reverse.

29. Pingback: markwilson.it » Installing Windows from a network server without Windows Deployment Services

30. Hi,

    Great information, sadly Cisco has restricted access to the SIP 7940 / 7960 firmware. I’d apprecaite it if someone could share it???

31. @Robin – if you check out the link in the post to VoIP Info: Configuring Cisco 79xx phones with Asterisk, you’ll find SIP firmware download links.

32. Good compilation of information for these phones, nice work.

    As you mentioned the documentation for the 79×1 and other XML configured devices is somewhat lacking but has improved. I’ve recently put together a thorough guide on setting up various different Cisco IP Phone models (including the IP Communicator) with asterisk. Examples of all configuration files have been included for reference. If anyone finds the information helpful it can be found at http://minded.ca/2009-12-16/configure-cisco-ip-phones-with-asterisk/

33. Hi Tyler – thanks for the link to your great resource on this! I’ll add it to the main post.

34. Any idea why the phones each night loose their personal settings ?

35. @femte I hadn’t experienced that but if the phones lose their settings (e.g. due to power loss) then they should reload them from the TFTP server at bootup.

36. You can try to use CIP-SIP – it’s a auto-provisioning web based software from some company. You can easy managed your Cisco IP Phones (all kind of models), also you can create your own directory, upload logo to the phone, manage memos, Directory search etc etc. I’ve tried it, and it very nice software, yes, thet just start, but it’s enough for me now. I’ll use this software in the future, because they anounced a lot of additional functionality and recommend to try.

    I’ve found thos software via (one coder – he works in this company), so, I’ve wrote email to info_at_eranet.com.ua and ask account for me.

37. Hi Mark, your site is an immensely helpful guide for configuring these phones. I have what may be a simple problem but I haven’t figured a solution.

    My 7960 works perfectly when connected to my BT HomeHub router. When I connect it to my laptops it seems to fail to register with my voip provider Sipgate (I get the phone icon with an “x” next to it) .

    I’ve used a WinXP ThinkPad with 3GB memory and a W7 netbook with 2GB and have DHCP enabled on the phone. I imagine it’s a problem with Windows’ firewall settings but may be wrong and can’t figure out which settings.

    Any ideas what I can do? Much appreciated.

    Many thanks.

38. @Vlad, you say “when I connect it to my laptops”. The phone should be between the laptop and the switch (there are two ports – one to PC and the other to switch). If that’s the case, then I don’t think the laptop should make a difference although I guess the BT router might be getting confused with two devices on one port (it shouldn’t, but it’s designed as a consumer product so probably doesn’t expect it)?

    Does it work with the phone and laptop on separate connections to the router?

39. Hi Mark,

    First, thanks for the response. Second, I finally got it working but, as I’ll explain, with an uncomfortable quirk.

    My objective is to able to use the 7960 on the road where WiFi is more readily available than LAN access. The phone is hence getting its internet connection via the laptop’s WiFi connection.

    How? I’ve used the Windows Network Bridge and Internet Connection Sharing to bridge the WiFi connection (routerlaptop) to the 7960 (laptopphone).

    I finally managed to get it working however it only does so when I disable my firewall. AVG Internet Security apparently doesn’t like using the PC as a server and having the PC use DHCP configs seems to be a problem. I’ve posed the question to AVG with no response so am using Windows Firewall for now.

    If you’re also using Windows Firewall, can you tell me or refer me to how to set up the configs. Thanks.

40. By the way, the 7960 works fine when connected via ethernet to the BT Home Hub, even before I opened ports 5004, 5060, 5080 and 10000 and attributed them to both devices (the laptop and 7960). This leads me to conclude that the problem is likely the laptop’s firewalls or network settings.

    To clarify my last comment above, the 7960 persistently “works” only for outbound calls with Windows Firewall enabled even though Sipgate shows my Cisco device “online”. When I first boot and register the 7960 with Sipgate, the phone receives incoming calls but then stops receiving them.

    Thanks.

41. Hi Mark, I finally got it working so that my laptop serves my 7960’s incoming and outgoing calls.

    For the benefit of anyone else facing this task and issue under these circumstances, I’ll confirm what you suggested: the BT HomeHub is not reliable for VoIP. The jury’s out on whether AVG Internet Security works with the 7960 bridged since I haven’t yet tried to reenable its firewall.

    I have an older version of the BTHH but BT also confirmed that the newer BTHHs (in the post to me) also have port limitations that other routers don’t. I’m now using a Belkin router.

    Best,

    Vlad

42. Hi Vlad – disappointing to hear that the Home Hub has these limitations (guess I won’t be getting one then…) but glad you managed to get things sorted. FWIW, I don’t use AVG any more – the combination of Windows Firewall, Microsoft Security Essentials and a hardware router seems to protect me reasonably well. On the non-Windows machines, I just use the built-in software firewalls too.

43. Hi,

    I was wondering what all things you require to setup 7960 at home. will this work with WINDOWS XP? Can someone please put up a complete list of things and how do set it up?

    Regards,
    Billy

44. Thanks for the info Mark, this is exactly what i needed to get this done. Truthfully i never knew tftp existed, but once i set that up on my ubuntu box and used the config files you provided as a starter, i had this going very quickly

    only issue now is i cant seem to get inbound calls going properly, keeps going to voicemail. not sure if thats the setup or what…but its very odd. anyone else encountered this?

45. Thanks a lot mate, I’ve finally got my 7940 running after some 5 years thanks to your help!!!

    Much appreciated.
    Thanks again
    Thanks!!!
    Thanks!
    Dan

46. Another thing to be aware of is the list of errors in the UK dial plan files:

    http://www.aa-asterisk.org.uk/index.php/Errors_in_Cisco_UK_Numbering_Plan_Documents

    This is a list of errors in the documentation. Some listed items may also reflect as errors in the associated software itself.

47. Mark – will this procedure work with a Cisco IP 7945 phone?

48. Probably not – I haven’t tried, but the 7941 uses a different format for the config files (XML-based). I guess newer models do too…

49. Does anyone know what the time interval for a Cisco 7940 to automatically contact the TFTP provisioning server to see if there are newer settings?

50. Good day mark! Have a question about changing default language from English to Russian or any other once on Cisco 7940G. Is it possible to do it just via TFTP for SIP configuration. Several days of trying to search for an answer on the web didn’t bring me any answer on that question.
    Sincerely
    Vlad

51. Hi Vlad, I’m pretty sure that it should be possible, but it’s not something I’ve tried and it’s been a couple of years since I looked at this now. Some of the links at the bottom of my blog post go to some forums that I found helpful – maybe people there might have the answer? Sorry I couldn’t help, Mark

52. Thanx for reply Mark. As far as I can see you are good specialist with Cisco equipment. Would love to stay in keep in touch with ya in a future … and maybe could answer on some questions on your blog as well.
    Sincerely Vlad (from Russia)

53. Do anyone has or tried to build a software to create / modify these config files? I’m struggling editing these configs manually, especially if I need to do it on couple phones. I’m not that much software developer to complete whole package, but there shouldn’t be that much work to read text config file, parse it into the variables, place on the screen, and write back modified data. Plus, if package will include TFTP server and basic configs, it will make Cisco phones more popular (especially newer models) with SIP providers.

    SIncerely,

    Mike / CCNP, CCVP, CCDP, CCSP, CCIE Voice (written), etc.
    http://www.ciscoheadsetadapter.com

54. Does anyone know how to configure a Cisco 501G to receive calls from the BT Homehub 2?
    The IP Phone provider is BT also, and the BT number used to ring a wireless BT IP handset. Thanks.

55. Hi all,

    I am wondering if anyone can help me. I have CISCO 7960 which used to work with SIPgate.co.uk and stopped one day. I re-flashed it with latest SIP firmware but can not make it work now. My home router has UDP ports forwardings are set. Sipgate works from PCs, mobiles on the same network.
    ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
    XMLDefault.cnf.xml

    2000

    2427
    2428

    P0S3-8-12-00

    ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
    SipDefault.cnf

    # SIP Default Generic Configuration File

    # Image Version
    image_version: P0S3-8-12-00

    # Proxy Server
    proxy1_address: “sipgate.co.uk” ; Can be dotted IP or FQDN
    proxy2_address: 222.33.22.11 ; 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 Server Port (default – 5060)
    proxy1_port: 5060
    proxy2_port: 5069
    proxy3_port: 5060
    proxy4_port: 5060
    proxy5_port: 5060
    proxy6_port: 5060

    # Proxy Registration (0-disable (default), 1-enable)
    proxy_register: 1

    # Phone Registration Expiration [1-3932100 sec] (Default – 3600)
    timer_register_expires: 600

    # Codec for media stream (g711ulaw (default), g711alaw, g729a)
    preferred_codec: g711ulaw

    # TOS bits in media stream [0-5] (Default – 5)
    tos_media: 5

    # Inband DTMF Settings (0-disable, 1-enable (default))
    dtmf_inband: 1

    # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always – always avt )
    dtmf_outofband: avt

    # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)
    dtmf_db_level: 3

    # SIP Timers
    timer_t1: 500 ; Default 500 msec
    timer_t2: 4000 ; Default 4 sec
    sip_retx: 10 ; Default 10
    sip_invite_retx: 6 ; Default 6
    timer_invite_expires: 180 ; Default 180 sec

    ####### New Parameters added in Release 2.0 #######

    # Dialplan template (.xml format file relative to the TFTP root directory)
    dial_template: dialplan

    # TFTP Phone Specific Configuration File Directory
    tftp_cfg_dir: “” ; Example: ./sip_phone/

    # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
    sntp_server: “” ; SNTP Server IP Address
    sntp_mode: unicast ; unicast, multicast, anycast, or directedbroadcast (default)
    time_zone: GMT ; Time Zone Phone is in
    dst_offset: 1 ; Offset from Phone’s time when DST is in effect
    dst_start_month: April ; Month in which DST starts
    dst_start_day: “” ; Day of month in which DST starts
    dst_start_day_of_week: Sun ; Day of week in which DST starts
    dst_start_week_of_month: 1 ; Week of month in which DST starts
    dst_start_time: 19 ; Time of day in which DST starts
    dst_stop_month: Jan ; Month in which DST stops
    dst_stop_day: “” ; Day of month in which DST stops
    dst_stop_day_of_week: Saturday ; Day of week in which DST stops
    dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month
    dst_stop_time: 2 ; Time of day in which DST stops
    dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment
    time_format_24hr: 1 ; Enable(1 – 24Hr Default)/Disable(0 – 12Hr)

    # Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
    dnd_control: 0 ; Default 0 (Do Not Disturb feature is off)

    # Caller ID Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
    callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous)

    # Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
    anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls)

    # DTMF AVT Payload (Dynamic payload range for AVT tones – 96-127)
    dtmf_avt_payload: 101 ; Default 101

    # Sync value of the phone used for remote reset
    sync: 1 ; Default 1

    ####### New Parameters added in Release 2.1 #######

    # Backup Proxy Support
    proxy_backup: “sipgate.co.uk” ; Dotted IP of Backup Proxy
    proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)

    # Emergency Proxy Support
    proxy_emergency: “sipgate.co.uk” ; Dotted IP of Emergency Proxy
    proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060)

    # Configurable VAD option
    enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable

    ####### New Parameters added in Release 2.2 ######

    # NAT/Firewall Traversal
    nat_enable: 1 ; 0-Disabled (default), 1-Enabled
    nat_address: “” ; WAN IP address of NAT box (dotted IP or DNS A record only)
    voip_control_port: 5060 ; UDP port used for SIP messages (default – 5060)
    # start_media_port: 16384 ; Start RTP range for media (default – 16384)
    # end_media_port: 32766 ; End RTP range for media (default – 32766)
    nat_received_processing: 1 ; 0-Disabled (default), 1-Enabled

    # Outbound Proxy Support
    outbound_proxy: “proxy.live.sipgate.co.uk” ; restricted to dotted IP or DNS A record only
    outbound_proxy_port: 5060 ; default is 5060

    ####### New Parameter added in Release 3.0 #######

    # Allow for the bridge on a 3way call to join remaining parties upon hangup
    cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default)

    ####### New Parameters added in Release 3.1 #######

    # Allow Transfer to be completed while target phone is still ringing
    semi_attended_transfer: 1 ; 0-Disabled, 1-Enabled (default)

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

    ####### New Parameters added in Release 4.0 #######

    # XML URLs
    services_url: “” ; URL for external Phone Services
    directory_url: “” ; URL for external Directory location
    logo_url: “” ; URL for branding logo to be used on phone display

    # HTTP Proxy Support
    # http_proxy_addr: “” ; Address of HTTP Proxy server
    # http_proxy_port: 80 ; Port of HTTP Proxy Server (80-default)

    # Dynamic DNS/TFTP Support
    dyn_dns_addr_1: “” ; restricted to dotted IP
    dyn_dns_addr_2: “” ; restricted to dotted IP
    dyn_tftp_addr: “” ; restricted to dotted IP

    # Remote Party ID
    remote_party_id: 0 ; 0-Disabled (default), 1-Enabled

    ####### New Parameters added in Release 4.4 #######

    # Call Hold Ringback (0-off, 1-on, 2-off with no user control, 3-on with no user control)
    call_hold_ringback: 0 ; Default 0 (Call Hold Ringback feature is off)

    ####### New Parameters added in Release 6.0 #######

    # Dialtone Stutter for MWI
    stutter_msg_waiting: 0 ; 0-Disabled (default), 1-Enabled

    # RTP Call Statistics (SIP BYE/200 OK message exchange)
    call_stats: 0 ; 0-Disabled (default), 1-Enabled

    ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

    SIP001380FDC397.cnf

    image_version:P0S3-8-12-00

    # Line 1 Setup
    line1_name: “SipGate”
    line1_authname: “34343434”
    line1_shortname: “SipGate”
    line1_password: “xxxxxx”
    proxy1_address: “sipgate.co.uk”
    proxy1_port: “5060”
    line1_displayname: “VOIP Line 1”; # Line 1 Display Name (Display name to use for SIP messaging)

    # Line 2 Setup
    line2_name: “AnotherSip”
    line2_authname: “Voip2”
    line2_shortname: “Voip2”
    line2_password: “Voip2”
    line2_displayname: “VOIP Line 2”; # Line 2 Display Name (Display name to use for SIP messaging)
    proxy2_address: “222.33.22.11”
    proxy2_port: 5069

    # Phone Label (Text desired to be displayed in upper right corner)
    phone_label: “My 7960” ; Has no effect on SIP messaging
    phone_prompt: “MySIP Phone”

    # Phone Password (Password to be used for console or telnet login)
    phone_password: “cisco” ; Limited to 31 characters (Default – cisco)

    # User classifcation used when Registering [ none(default), phone, ip ]
    user_info: none
    telnet_level: 1

    ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

    What is wrong with my configuration as non of the lines works. My priority is to fix Line 1 i.e. Sipgate at the moment but ideally to make both work.

56. Mark! Many Thanks,

    I’d been flashing my phone for a week (new Cisgo 7960g (APP P0030301MFG2, Boot PC03010200 3.1(MFG@2).. till I read your blog…

    went back to version 6 of the SIP firmware as you suggested flashed 1st time then straight upgrade to P0S3-8-12-00 with no issues at all.. Happy camper I’d buy you a beer if you had paypal on this blog..

    Colm

57. Glad it helped you Colm :-) And thanks for the virtual beer – the thought is still good!

58. Pingback: markwilson.it » Configuring a Cisco IP phone for VoIP using SIP – revisited

59. Hi Mark, I am having issues with incoming calls on Cisco 7960G on sipgate.co.uk service.

    Outgoing call seems to work perfectly,

    For Incoming calls, Phone rings but when you answer it. no voice transmission at all, i cant hear the caller nor the caller can hear me..

    Whats your thoughts ?

    Thanks
    Sachin

60. Hi, i’m having a issue in Boot Load ID version in 7940 cisco phone. My requirement is

    APP LOAD ID: P00308010200
    BOOT LOAD ID: PC0303010100

    my app load is matching with firmware 8.1(2) but i’m unable to downgrade my BOOT LOAD ID that is PC0303010200 to PC0303010100.

    What i have to do now? Please help

61. Pingback: markwilson.it » Raspberry Pi infrastructure server (DNS, DHCP, TFTP)
62. Pingback: Compact and bijou - my home office! - markwilson.it
63. Pingback: Raspberry Pi FTP server - markwilson.it

64. Hi Mark, I appreciate your valuable blog, thanks. I have one problem, cisco 7942 ip phone registration is not happening, it keeps showing registering..nothing happening,I am using Asterisk box without NAT.

    Here is my TFTPD32 logs:-

    Connection received from 192.168.2.103 on port 50079 [03/06 21:26:12.725]
    Read request for file . Mode octet [03/06 21:26:12.725]
    File : error 2 in system call CreateFile The system cannot find the file specified. [03/06 21:26:12.725]
    Connection received from 192.168.2.103 on port 50136 [03/06 21:26:12.795]
    Read request for file . Mode octet [03/06 21:26:12.795]
    File : error 2 in system call CreateFile The system cannot find the file specified. [03/06 21:26:12.795]
    Connection received from 192.168.2.103 on port 49924 [03/06 21:26:12.875]
    Read request for file . Mode octet [03/06 21:26:12.876]
    File : error 2 in system call CreateFile The system cannot find the file specified. [03/06 21:26:12.876]
    Connection received from 192.168.2.103 on port 51636 [03/06 21:26:13.114]
    Read request for file . Mode octet [03/06 21:26:13.115]
    Using local port 61629 [03/06 21:26:13.115]
    : sent 14 blks, 7036 bytes in 0 s. 0 blk resent [03/06 21:26:13.134]
    Connection received from 192.168.2.103 on port 51847 [03/06 21:26:17.087]
    Read request for file . Mode octet [03/06 21:26:17.088]
    Using local port 61630 [03/06 21:26:17.088]
    : sent 1 blk, 298 bytes in 0 s. 0 blk resent [03/06 21:26:17.091]
    Connection received from 192.168.2.103 on port 49753 [03/06 21:35:44.128]
    Read request for file . Mode octet [03/06 21:35:44.129]
    File : error 2 in system call CreateFile The system cannot find the file specified. [03/06 21:35:44.129]
    Connection received from 192.168.2.103 on port 50000 [03/06 21:35:44.204]
    Read request for file . Mode octet [03/06 21:35:44.205]
    File : error 2 in system call CreateFile The system cannot find the file specified. [03/06 21:35:44.205]
    Connection received from 192.168.2.103 on port 49557 [03/06 21:35:44.273]
    Read request for file . Mode octet [03/06 21:35:44.274]
    File : error 2 in system call CreateFile The system cannot find the file specified. [03/06 21:35:44.274]
    Connection received from 192.168.2.103 on port 52450 [03/06 21:35:44.518]
    Read request for file . Mode octet [03/06 21:35:44.519]
    Using local port 51006 [03/06 21:35:44.519]
    : sent 14 blks, 7026 bytes in 0 s. 0 blk resent [03/06 21:35:44.539]
    Connection received from 192.168.2.103 on port 52472 [03/06 21:35:47.623]
    Read request for file . Mode octet [03/06 21:35:47.624]
    File : error 2 in system call CreateFile The system cannot find the file specified. [03/06 21:35:47.624]
    Connection received from 192.168.2.103 on port 52957 [03/06 21:35:48.968]
    Read request for file . Mode octet [03/06 21:35:48.969]
    File : error 3 in system call CreateFile The system cannot find the path specified. [03/06 21:35:48.969]
    Connection received from 192.168.2.103 on port 50570 [03/06 21:35:52.397]
    Read request for file . Mode octet [03/06 21:35:52.398]
    Using local port 58138 [03/06 21:35:52.398]
    : sent 1 blk, 298 bytes in 0 s. 0 blk resent [03/06 21:35:52.400]

65. Sorry Shameer, I haven’t any experience with the 7942s. One of the reasons I use a 7940 is because they are relatively easy to configure for this purpose but it’s been years since I got my hands on new Cisco hardware!

66. Pingback: SIP Gate – Calling 0800 on your Nokia or Cisco 7940… – Simkin's Musings

This site uses Akismet to reduce spam. Learn how your comment data is processed.