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. 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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