[Asterisk-Users] Asterisk 'losing' upstream provider registration state during small network outages.

Brian West brian.west at mac.com
Sat Jun 25 09:42:42 MST 2005


update...

/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jun 25, 2005, at 12:18 AM, Steve wrote:

>
> 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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