[Asterisk-Users] Asterisk 'losing' upstream provider registration
state during small network outages.
Steve
asterisk at michiganbroadband.com
Thu Jun 23 12:40:40 MST 2005
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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