search for: registerattempt

...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 ------------...

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...

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-p...

...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 canreinv...

...???????? 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 ins...

...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 ju...

...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/ Whe...

...= 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 =&g...

...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...

...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

...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

...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

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

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

...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 a...

...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.n...

...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...

...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 ;calleve...

...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 insecu...