2009/1/20 Simon Dixey <[email protected]> > > 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. > > > ------------------------------ > See all the ways you can stay connected to friends and > family<http://www.microsoft.com/windows/windowslive/default.aspx> > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
