[Asterisk-Users] Asterisk 'losing' upstream provider
registrationstate during small network outages.
Steve
asterisk at michiganbroadband.com
Sat Jun 25 09:17:00 MST 2005
Thank you for your reply.
Did moving to a newer version fix the problem?
I am still asking here if it is a problem with asterisk or am I not doing
something right?
Running a cron job every hour just seems way to hokey for something that
as far as I know is supposed to work.
besides that the phone would be down for up to an hour while we wait for
the cron job...
That will never fly :-)
This is supposed to be a phone system.
Thanks & take care.
Steve
On Sat, 25 Jun 2005, jurczak wrote:
> 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: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] 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 => ##########: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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