Some time ago (with previous releases of Asterisk) I had the same problem with broadvoice, so I added a cron job that reloads the sip every 1 hour. I know this is not the best solution, but at the time this seed fine.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Sent: Saturday, June 25, 2005 8:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 'losing' upstream provider registrationstate during small network outages. Still looking for some help here..... Is this problem due to asterisk, the two week old version of CVS-HEAD I'm running? Or is it that I simply have not configured it correctly? Any helpful hints would be greatly appreciated. I'm about to try starting all over again from scratch and do a reinstall/recompile of the latest CVS-HEAD only to most likely find the problem has not gone away. Thanks! Steve On Thu, 23 Jun 2005, Steve wrote: > Now that I have most everything actually working I've noticed that about > every 3-4 days on average..... and at worse... Once a day my asterisk box > seems to lose it's registered state with our sip provider and no longer will > take any incoming calls. > > The caller simply hears a fast busy (reorder) > > If I do a reload at the command prompt all is well for another few days..... > > What I'm looking for is a way to make asterisk stay registered even if the > network drops for 10 minutes.... > > Or more correctly I should probably say re-register automatically if > registration state is lost or has timed out at the outer end (our isp sip > provider) > > > Our cable (Internet Connectivity) service provider has been going down for > 10-30 minutes in the middle of the night lately and I keep losing my > registered (connected) state where I can accept inbound calls via sip from > our service provider. > > It seems that I read somwhere awhile back that this change was recently > incorporated to asterisk by default and is by design where it would not keep > trying forever to reconnect to a sip provider if the net was down. > > If this is correct this behavior seems to be a bad thing! I'd really like it > to re-establish it's registration automatically when the net is available > again :-) > > Is there a setting that I should be using to accomplish this? > > Reading the docs as I have so far seem to have revealed that I can set the > expiry times and re-register times for my own sip clients to the box but are > very unclear in how to make my asterisk box 'stay registered' or auto > re-register after a 15 or 20 minute network outage of my upstream ISP. > > Attached is the relevant part of my sip.conf (also seen before on a previus > thread) :-) > > I'm now running CVS-HEAD compiled about 2 weeks ago and it's probably about > time for an update. > With quick look at the changelogs I didn't notice anything regarding this > behavior. > > > Next tiem this happens I will also try and capture more detail. > sip debug generaly was showing nothing go by with an attempted incoming call. > > And (from memory) sip show peers looked normal as if ready for incoming > calls. > > Thanks Much! > > Steve (Still an Aterisk Newbie) > > > > > > > > ;-------------Testing------------------ > > > [general] > port = 5060 > bindaddr = 0.0.0.0 > allow=ulaw > ; dtmfmode=info > ; nat=yes > > > > ; This section is because i'm behind nat > externip = x.x.x.x ;Outside address > localnet = 10.73.73.133 ;Inside address > localmask = 255.255.255.0 ;Inside subnet > > context = sip ; Default context for incoming calls > register => ##########:[EMAIL PROTECTED]/1000 > register => ##########:[EMAIL PROTECTED]/4078 > register => ##########:[EMAIL PROTECTED]/4077 > > > [stanaphone-out] > > ;works!!! > host=sip.stanaphone.com > context=sip > type=friend > dtmfmode=rfc2833 > canredirect=no > disallow=all > allow=ulaw > insecure=very > username=secret > fromuser=secret > secret=secret > > > ;more testing broadvoice examples > ;THIS ONE WORKS!!! > > [our-sip-provider-out] > type = peer > host = sip.provider.net > secret = secret > user=phone ; I needed this to make it work (what tha ????) > fromuser = secret > username= secret > authname= secret > fromdomain = sip.provider.net > context = sip > insecure=very ; To allow registered hosts to call without re-authenticating > canreinvite = no > ; BV claims they support rfc2833, but for some reason passing digits > ; after a connected call only works with inband > dtmfmode = rfc2833 > ;dtmf=inband > > CVS-HEAD > Running Version: > Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-06-06 > 22:32:05 > > > *CLI> show version files > File Revision > ---- -------- > cdr_custom.c Revision: 1.11 > cdr_manager.c Revision: 1.6 > cdr_csv.c Revision: 1.16 > pbx_functions.c Revision: 1.3 > chan_zap.c Revision: 1.458 > chan_phone.c Revision: 1.52 > chan_modem_i4l.c Revision: 1.27 > chan_oss.c Revision: 1.49 > chan_features.c Revision: 1.12 > chan_skinny.c Revision: 1.78 > chan_local.c Revision: 1.47 > chan_iax2.c Revision: 1.303 > iax2-parser.c Revision: 1.45 > iax2-provision.c Revision: 1.12 > chan_mgcp.c Revision: 1.123 > chan_agent.c Revision: 1.136 > chan_modem_bestdata.c Revision: 1.16 > chan_sip.c Revision: 1.754 > chan_modem_aopen.c Revision: 1.15 > chan_modem.c Revision: 1.40 > io.c Revision: 1.10 > sched.c Revision: 1.19 > logger.c Revision: 1.74 > frame.c Revision: 1.57 > loader.c Revision: 1.45 > config.c Revision: 1.66 > channel.c Revision: 1.202 > translate.c Revision: 1.37 > file.c Revision: 1.68 > say.c Revision: 1.60 > pbx.c Revision: 1.254 > cli.c Revision: 1.86 > md5.c Revision: 1.14 > term.c Revision: 1.10 > ulaw.c Revision: 1.4 > alaw.c Revision: 1.3 > callerid.c Revision: 1.32 > fskmodem.c Revision: 1.7 > image.c Revision: 1.15 > app.c Revision: 1.66 > cdr.c Revision: 1.40 > tdd.c Revision: 1.6 > acl.c Revision: 1.45 > rtp.c Revision: 1.133 > manager.c Revision: 1.99 > asterisk.c Revision: 1.162 > dsp.c Revision: 1.43 > chanvars.c Revision: 1.8 > indications.c Revision: 1.25 > autoservice.c Revision: 1.12 > db.c Revision: 1.18 > privacy.c Revision: 1.5 > enum.c Revision: 1.26 > srv.c Revision: 1.13 > dns.c Revision: 1.14 > utils.c Revision: 1.47 > config_old.c Revision: 1.4 > plc.c Revision: 1.5 > jitterbuf.c Revision: 1.15 > dnsmgr.c Revision: 1.5 > > > Sorry for the LONG delay on this wrap up. > > > Take care! > > Steve > > > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
