2009/1/20 Simon Dixey <simon_qro at hotmail.co.uk>
>
> Hi folks,
>
> I wonder if any of you out there are using Siemens S685IP base station(s)
> (with S68H handsets) on Asterisk and experiencing problems with SIP
> registrations where the SIP extensions do not ring and peers become
> unreachable after a period of time.
>
> Symptoms are rather sporadic, but as described, SIP extensions being
> unreachable from Asterisk perspective. Also experienced 'not
possible'
> messages trying to dial using the handsets.
> In Asterisk - 'sip show peers' shows the SIP device status as
'UNKNOWN' and
> console output says chan_sip.c Peer 'xxx' is unreachable. (as one
would
> expect if it can't see it!!)
> So I'm thinking it's a problem with the base.. or.. some issue with
> qualifications and possibly the base station not responding (guessing
here).
>
> I'm finding that the Siemens web GUI reports messages similar to
'server
> not accessible' or 'registration failed'. These messages
appear randomly
> throughout the day following successful previous registration.
> Have enabled 'sip set debug ip xxx' for one of the bases and on the
Siemens
> web GUI disabling the SIP account and re-enabling it doesn't work
generally,
> and no SIP messages are being directed from the base to Asterisk. Leads me
> to think it's a base/firmware issue.
>
> Some times the phones are contactable for a day without fault, other times
> they're problematic at random intervals. It's not always all of
the SIP
> accounts assigned on the Siemens base, sometimes it's just one account,
> other times it's all accounts. (What a horrible situation to
debug/fault
> find! Glad my Aastra's are reliable!)
>
> The only resolve I've found which is rather unacceptable is to reboot
the
> Siemens base station.
> Upon doing so, the base re-registers all of the accounts to the Asterisk
> server and calls to/from handsets work for 'a period of time'...
>
> My setup is as follows:
>
> - Asterisk 1.4.22
> - Base "1" has 3x handsets and 3x SIP accounts (providers) and
the SIP
> accounts are individually assigned to each handset.
> - Base "2" has 5 handsets and 6 SIP accounts. 5 SIP accounts for
each
> handset, then the 6th SIP account is a 'group' extension which
rings all of
> the phones on the base station. MWI for VM is also used and works. Call
> waiting disabled on handsets.
> - Both bases on latest available as of Jan 09 - 021400000000 / 043.00
>
> The bases are set up with static IPs, info services off, etc. Am not using
> SIP domains, no NAT, all communicating on a LAN on same network so no
> routing or latency issues.
> Registrar server defined and refresh time set to 180 seconds (Siemens
> default).
> SIP.conf has nat=no, qualify=yes
>
what if setting qualify=10000 (or any very long value) for every handset but
one for which qualify=1000 is used ?
maybe, the base takes too long to reply to qualify messages for an unknown
reason and changing from qualify=yes to qualify=something might help to
confirm parts of diagnosis ...
> . host= is currently dynamic.. maybe I should set this to the IP of the
> base as they're using static IPs, but reading the specs of this setting
> describe set to 'dynamic' if the phone should register itself...
hmm.
>
>
> I've seen similar posts from other users on the exact same subject.
> Sources as follows:
> Voip-Info http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP (see
> 'Discussions' tab).
> The Open Sourcerer:
>
http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/comment-page-3/
> Siemens Forums:
> http://www.siemens-gigaset-forum.com/de/posts/list/13788.page
> Siemens Customer care:
> http://www.siemens-gigaset-forum.com/de/posts/list/13788.page (people with
> aged open support calls)
>
> Siemens support via phone were rather unhelpful and didn't grasp the
> technicalities of the issue I was conveying so drew a blank (have I
checked
> my router.. hmm!)
>
> I'm guessing this is a firmware issue but intrigued to know if others
are
> experiencing the same.
> Anyone else experiencing similar problems? Or indeed successes with a
> similar setup?
>
> Can anyone recommend a stable working DECT SIP phone for enterprise use
> with Asterisk? (The Snom M3 looks good but read about issues with
transfers
> which concern me)
> May have to resort to ditching the S685s and go with Aastra desk sets all
> round - shame to lose the flexibility of cordless though.
>
>
> Many thanks in advance.
>
>
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