Steve
2005-Jun-23 12:40 UTC
[Asterisk-Users] Asterisk 'losing' upstream provider registration state during small network outages.
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@sip.stanaphone.com/1000
register => ##########:secret@sip.provider.net/4078
register => ##########:secret@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@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
Joseph
2005-Jun-23 13:58 UTC
[Asterisk-Users] Asterisk 'losing' upstream provider registration state during small network outages.
I was just testing Shaw Extreme Cable connection and had a similar problem but with IAX. After few hours I was losing IAX registration nor could I make a call to VoipJet using IAX. Rebooting the firewall restore the connection but after few hours it would happen again. Though, I think the problem is with Shaw Cable network as I re-connected the same firewall to DSL and the connection stays up. Iit would nice if there was some kind of "switch setting" that would allow to monitor registration with the provider every certain number of minutes, if it detect de-registration it should try to re-register if it fails drop an email to a user/administrator etc. -- #Joseph On Thu, 2005-06-23 at 15:40 -0400, 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@sip.stanaphone.com/1000 > register => ##########:secret@sip.provider.net/4078 > register => ##########:secret@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@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
Steve
2005-Jun-24 22:18 UTC
[Asterisk-Users] Asterisk 'losing' upstream provider registration state 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@sip.stanaphone.com/1000 > register => ##########:secret@sip.provider.net/4078 > register => ##########:secret@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@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 > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >