Displaying 19 results from an estimated 19 matches for "registerattempts".
2011 Sep 14
1
Sip re-register / delay problem.
...kly.
- check from time to time all users but no too often to see if is logged and
can be called.
Overall i want only lagged users to reregister and users with good response
time to be check from time to time.
defaultexpiry = 900
defaultexpirey = 900
maxexpiry = 300
maxexpirey = 300
minexpiry = 60
registerattempts = 5
registertimeout = 5
rtpholdtimeout = 900
rtptimeout = 60
jbmaxsize = 60
jbresyncthreshold = 200
qualify = yes
qualify = 600
qualifyfreq = 60
Thank you.
P.S. If you consider that i use too much options you can tell me what to
drop. I use asterisk 1.8.6.0.
-------------- next part -------------...
2010 Jul 26
1
Optimize peers registration under jitter/delay.
Hello,
I want to optimize my registrations and calls of peers to my asterisk
with the following options in sip.conf:
---///---
qualify = yes
qualify = 500
qualifyfreq=5
registerattempts = 0
registertimeout = 10
maxexpiry = 60
minexpiry = 20
defaultexpiry = 600
---///---
Can someone more experienced with these settings to help me to
optimize connections from peers with mobile phone that using operator
Internet with delay/jitter conditions?
I chooses values above after many tests...
2012 Oct 08
1
Sip registration Asterisk 1.8
Hello,
I have a local Asterisk server that keep loosing its registration to main
Asterisk server. The local asterisk server is on the local subnet, it acts
as a client with extension 808.
Local server
Sip.conf
register => 808:password at as2.xxxxx.com
registertimeout=20
registerattempts=10
Main Asterisk Server sip.conf
[808]
type=friend
context=sip-phones
call-limit=99
callerid="child2" <808>
disallow=all
allow=ulaw
allow=alaw
username=808
secret=xxxxx
dtmfmode=rfc2833
host=dynamic
mailbox=808
nat=yes
qualify=yes
canreinvite=no
== Extension Changed 800[sip-ph...
2007 Aug 09
1
strange warning
...Can somebody explain me this? Below is my
client configuration.
[general]
bindport=9060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
context=incoming
allowexternalinvites=yes
register=> diet:pepsi at magnum.axvoice.com:9060
registertimeout=10 ;(default 20 secs)
registerattempts=10 ;set it to zero for infinit attempts
Following is the server sip account im using for my client asterisk to
register:
[diet]
username=diet
type=friend
secret=pepsi
qualify=no
nat=yes
mailbox=12129339033
insecure=invite,port
call-limit=2
host=dynamic
dtmfmode=rfc2833
context=local
canreinvi...
2017 Oct 10
2
Asterisk chan_sip registration attempts
...???????? dnsmgr Username?????? Refresh
State??????????????? Reg.Time???????????????? //
//X.X.X.X:5060??????????????????? N????? <LOGIN> ?????????? 105
Unregistered?????????? /*
*
This happens sometimes once per 4 hours, sometimes once per a week.
I don't see any patterns.
*sip.conf:*
registerattempts=0
registertimeout=20
*peer confifuration:*
[XXXX-friend]
disallow=all
host=192.168.1.1
defaultuser=<phone number>
fromuser=<phone number>
callerid=<phone number>
secret=<ISP secret>
type=friend
qualify=yes
allow=ulaw
allow=alaw
nat=no
rtpkeepalive=10
dtmfmode=rfc2833
inse...
2010 Aug 05
1
Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
...pbx_config]
Unfortunately, no matter how I configure extensions.conf or sip.conf,
the phone call always ends up saying: "Extension is unavailable.
Please leave your message after the tone".
sip.conf:
[general]
register => NPANXXZZZZ:PASSWORD at SERVICE_PROVIDER_IP
registertimeout=29
registerattempts=0
defaultexpiry=60
[DID_NUMBER]
type=peer
context=default
host=SERVICE_PROVIDER_IP
authuser=DID_NUMBER
fromuser=DID_NUMBER
fromdomain=SERVICE_PROVIDER_REALM
remotesecret=SERVICE_PROVIDER_PASSWD
secret=SERVICE_PROVIDER_PASSWD
dtmfmode=rfc2833
disallow=all
allow=ulaw
qualify=yes
I am attempting jus...
2011 Apr 18
2
Registrations stops after 403 FORBIDDEN
...ons/subscriptions (default 60)
defaultexpiry=120 ; Default length of incoming/outgoing
registration
;----------------------------------------- OUTBOUND SIP REGISTRATIONS
------------------------
registertimeout=240 ; retry registration calls every 20
seconds (default)
;registerattempts=0 ; Number of registration attempts before
we give up
; 0 = continue forever, hammering the
other server
; until it accepts the registration
; Default is 0 tries, continue forever/
When...
2010 Nov 03
1
inbound call issue...
...= 216.26.109.22
g726nonstandard = no
jbenable = yes
jbforce = no
jblog = no
localnet = internal subnet
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 60
mohinterpret = default
nat = yes
notifyringing = yes
pedantic = no
progressinband = never
promiscredir = no
realm = asterisk
recordhistory = no
registerattempts = 0
registertimeout = 20
relaxdtmf = no
sendrpid = no
sipdebug = no
t1min = 100
t38pt_udptl = no
tos_audio = none
tos_sip = none
tos_video = none
trustrpid = no
useragent = Asterisk PBX
usereqphone = no
videosupport = no
disallow = all
allow = ulaw,gsm
subscribecontext = device-hints
register =>...
2008 Nov 28
0
Calls drop after a couple of minutes.
...also sip.conf
[justvoip.com]
type=peer
host=sip.justvoip.com
fromdomain=sip.justvoip.com
progressinband=yes
disallow=all
allow=alaw ; only alaw works with sip1...
nat=no
canreinvite=no
qualify=yes
insecure=port,invite
username=imagi-justvoip
fromuser=00491785450880
secret=xxxxxxxxxxxx
registerattempts=0 ; keep trying to register (normally times out after
10 attempts)
context=from-external
from rtp.conf
rtpstart=19000
rtpend=20000
--
Simon Tennant _____________________________________________
fixed: .uk +44 20 7043 6756 .de +49 89 420 955 854
mob: .uk +44 78 5335 6047...
2010 Feb 23
2
SIP provider registration attempts
...e time lapse).
When the DSL is down I get:
"sip show registry":
Host Username Refresh State Reg.
Time
sip.provider.com:5060 xxxxxxxxxx 105 Request Sent Tue,
23 Feb 2010 18:06:42
I've read about the sip registerattempts = Number option but:
a) is this the option I'm really looking for?
b) what happens if the DSL lines come back up? does Asterisk re-attempt registration automatically?
Thanks,
Vieri
2010 Nov 13
0
problem registering to ekiga.net
...llow=g726
;allow=g729
allow=speex
allow=ulaw
allow=alaw ; Allow codecs in order of
allow=ilbc ; preference
allow=gsm
;allow=h261
localnet=10.0.0.0/255.0.0.0
register => magwas:mypassword at ekiga.net
registertimeout=20 ; retry registration calls every 20
seconds (default)
registerattempts=0
2013 Nov 08
1
11.5.0 - SIP registration not retrying after failures
...SIP registrations.
There was evidently a temporary authentication problem at the provider.
Does a Forbidden response cause Asterisk to decide never to retry?
Even if it is a rare situation, is it possible to configure Asterisk to
recover from it automatically by retrying?
I checked sip.conf, and registerattempts was left unset (defaults to 0=forever).
I couldn't see any other relevant settings.
Cheers
Tony
--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org
2006 Oct 11
3
asterisk 1.2.12 lost phone registrations today... why?
I lost my internet connection today for a short time.
During that time 1.2.12.1 stopped talking to my phones.
Asterisk was still working as I got 2 voicemails. I have TDM analog
cards for incoming calls.
Anyway my cisco phones had X's (lost registration) and my uniden phones
said "Registration error".
Why would phones loose registration to asterisk when the internet
connection
2010 Jun 04
1
originating a sip call from the CLI
Hello again!
I just got a SIP account and it seems - from a config on the net -, that
I've configured it correctly. But I get no call to the outside. Registration
was OK.
I tried:
channel originate sip/1/echo at iptel.org Application ...
I see the channel active for a while, but no call gets established.
In my config I have defined the section [iptel] for the outgoing call and I
2006 Mar 14
1
Codec Issue
...work and incoming don't.
I have included my sip.conf code and extensions.conf code below:
;sip.conf
[general]
bindport=5064
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
srvlookup=yes
canreinvite=no;
autocreatepeer=yes
nat=yes
;dtmfmode=info
;dtmfmode=rfc2833
insecure=very
registerattempts=0
;context=default
register => username@providerIP/1234
;To make outgoing calls specify this block
[providerIP]
type=peer
user=phone
host=providerIP
port=6060
fromdomain=providerIP
fromuser=username
secret=password
username=username
insecure=very
context=incomingpstn
authname=username
al...
2009 Aug 04
0
SIP server behind NAT
...t; ;notifyringing = yes ; Notify subscriptions on RINGING state
> ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
> ;regcontext=sipregistrations
> ;registertimeout=20 ; retry registration calls every 20 seconds (default)
> ;registerattempts=10 ; Number of registration attempts before we give up
> callevents=no ; generate manager events when sip ua performs events (e.g. hold)
> externip=The_IP_of_my_router ; Address that we're going to put in outbound SIP messages
> ;externhost=foo.dyndns.ne...
2016 Jan 20
2
Incoming webrtc call succeeds in Firefox but fails in Google Chrome
...allow=h263p
allow=h263
allow=h261
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscipher=ALL
tlsclientmethod=tlsv1
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
callevents=no
jbenable=no
videosupport=yes
allowguest=no
srvlookup=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
g726nonstandard=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
notifyringing=yes
notifyhold=yes
nat=yes
[1000]
deny=0.0.0.0/0.0.0.0
secret=6ff108122cce3b0b45e0abf374c14ef4
dtmfmode=rfc2833
canreinvite=no
context=from-internal
h...
2006 Dec 18
0
pap2/wrt54gs/asterisk
...of a section defined
; below.
;
; Examples:
;
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;registertimeout=20 ; retry registration calls every 20
seconds (default)
;registerattempts=10 ; Number of registration attempts before
we give up
; 0 = continue forever, hammering the
other server until it
; accepts the registration
; Default is 0 tries, continue forever
;calleven...
2006 Jan 04
0
confusion about contexts - SER
...below. I would appreciate any advice as to why these issues are
occurring.
Many thanks,
Aisling.
;sip.conf
[general]
bindport=5064
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
srvlookup=yes
canreinvite=no;
autocreatepeer=yes
nat=yes
dtmfmode=info
insecure=very
registerattempts=0
register => username:password@sip.blueface.ie/1234
;To receive incoming calls specify this and replace
"yourcontext-pstn" for your dial plan
[blueface-in]
type=peer
host=sip.blueface.ie
context=pstn
[1234]
type=friend
username=1234
canreinvite=no
context=pstn
insecur...