<br><br><div class="gmail_quote">2009/1/20 Simon Dixey <span dir="ltr"><<a href="mailto:simon_qro@hotmail.co.uk">simon_qro@hotmail.co.uk</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<br>
Hi folks,<br>
<br>
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.<br>
<br>
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.<br>
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!!) <br>
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).<br>
<br>
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.<br>
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.<br>
<br>
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!)<br>
<br>
The only resolve I've found which is rather unacceptable is to reboot the Siemens base station.<br>
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'...<br>
<br>
My setup is as follows:<br>
<br>
- Asterisk 1.4.22<br>
- Base "1" has 3x handsets and 3x SIP accounts (providers) and the SIP accounts are individually assigned to each handset.<br>
- 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.<br>
- Both bases on latest available as of Jan 09 - 021400000000 / 043.00<br>
<br>
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.<br>
Registrar server defined and refresh time set to 180 seconds (Siemens default).<br>
SIP.conf has nat=no, qualify=yes</div></blockquote><div><br>what if setting qualify=10000 (or any very long value) for every handset but one for which qualify=1000 is used ?<br><br>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 ...<br>
<br> </div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div>. 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.<br>
<br>
<br>I've seen similar posts from other users on the exact same subject. Sources as follows:<br>
Voip-Info <a href="http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP" target="_blank">http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP</a> (see 'Discussions' tab).<br>
The Open Sourcerer: <a href="http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/comment-page-3/" target="_blank">http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/comment-page-3/</a><br>
Siemens Forums: <a href="http://www.siemens-gigaset-forum.com/de/posts/list/13788.page" target="_blank">http://www.siemens-gigaset-forum.com/de/posts/list/13788.page</a><br>
Siemens Customer care: <a href="http://www.siemens-gigaset-forum.com/de/posts/list/13788.page" target="_blank">http://www.siemens-gigaset-forum.com/de/posts/list/13788.page</a> (people with aged open support calls)<br>
<br>
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!)<br>
<br>
I'm guessing this is a firmware issue but intrigued to know if others are experiencing the same.<br>
Anyone else experiencing similar problems? Or indeed successes with a similar setup?<br>
<br>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)<br>
May have to resort to ditching the S685s and go with Aastra desk sets all round - shame to lose the flexibility of cordless though.<br>
<br>
<br>
Many thanks in advance.<br>
<br><br><hr>See all the ways you can stay connected <a href="http://www.microsoft.com/windows/windowslive/default.aspx" target="_blank">to friends and family</a></div>
<br>_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br>