Cisco 9971 Help

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I’m an asterisk hobbyist, and I’ve got my hands on some cisco 9971’s preloaded with SIP (I think they might only come in SIP flavour actually?). I am quite excited about the possibilities with this kit – especially video calls. Unlike the earlier Cisco phones (e.g. 79 series), these can’t be used standalone, and require a TFTP server to get their config. After many hours of faffing, I’ve got the basic config parsing OK, but now I am stuck. The phone won’t actually finish booting without access to a couple more files – a g4_tones.xml (which I’ve created from a g3_tones I found kicking about, following a tip on the net) and gd_sip.jar which seems to contain locale specific information or at least be locale specific. From what I can gather, the gd_sip.jar is contained in po-locale-en-xxxxx which is listed on Cisco’s website for download but requires a partner login.

I know that on the voip-info site there is reference to possibly getting access to the cisco site via a $12 maintenance contract, and I am not adverse to paying monies where due at all, but I’m based in the UK and haven’t found anyone who knows anything about this.

Also, even if I had a maintenance contract, I’m not sure it would give me access to the po-locale-en-xxxx build as that seems to be part of call minder distributions, which would presumably be outside the scope of an
‘end point’ maintenance licence??

Any suggestions on how I source, re-create or otherwise address the gd_sip.jar issues or get a paid login to the cisco site would be hugely helpful, offlist is fine if more appropriate. I will write up my experiences for the greater good – these phones have the potential to be awesome with asterisk.

Cheers Patrick

4 thoughts on - Cisco 9971 Help

  • Stoyan Marinov wrote:

    Looks promising – a later firmware load, so the file I was looking for was not present, but still hopeful!

    Many thanks for the tip,

    Patrick

  • Getting closer…

    As a heads-up, the files in the link do not include the locale information
    (gd-sip.jar), but I have tracked down something suitable for that…

    The phones now get all the files that are essential, but never register with asterisk (there is no network traffic). The phone logs show:
    8360 NOT 00:21:14.387613 CVM-ccsip_register_send_msg: Error: cc_cfg_table is null.

    Googling this seems to suggest that I am not alone here, and that possibly the SIP 1.9 build on the website below is broken, or if it isn’t broken, the format of the SEPxxxx.cnf.xml file has subtly changed, but its not clear how. Anyone got a 9971 working with 1.9.2-2SR1-9 want to share their SEPxxx.cnf.xml?

    I’d like to regress to SIP 1.8, as I think that may fix the problem, but so far I haven’t been able to locate, ahem, a copy.

    Cheers Patrick

  • Yes, bad form to follow up on my own post. Anyway, the secret sauce is indeed a “correct” SEPxxxx.cnf.xml, and a kind lister provided a working model for SIP 1.9. I now have the endpoint registered, should be downhill from here. Full write up will indeed follow. No doubt I will be back soon with more q’s Cheers Patrick