[Asterisk-Users] Asterisk 'losing' upstream provider registration
state during small network outages.
Steve
asterisk at michiganbroadband.com
Fri Jun 24 22:18:49 MST 2005
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 => ##########:secret at sip.stanaphone.com/1000
> register => ##########:secret at sip.provider.net/4078
> register => ##########:secret at sip.provider.net/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 root at Vontage 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
>
>
>
>
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