search for: registertimeout

...e 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 -------------- An HTML attachment...

...6361355,1,Playback(beep) exten => 17036361355,2,SayDigits(${EXTEN}) exten => 17036361355,3,Goto(testdtmf|s|1 ;Ring on Elle mobile phone. ;exten => s,1,Answer() ;exten => s,n,Dial(SIP/17037171234,150,r,t,) [general] register =>1112530146:albany!@#123 at sip3.voipvoip.com/1112530146 registertimeout=20 context=incoming allowoverlap=no bindport=5060 bindaddr=192.168.1.10 srvlookup=no ;context=incoming ; The SIP provider [voipvoip.com] canreinvite=no username=1112530146 fromuser=1112530146 secret=albany!@#123 context=incoming type=friend fromdomain=sip3 at voipvoip.com host=69.90.209.57 dtmfmod...

...nd shows de message: "Oct 14 16:48:22 NOTICE[4090]: chan_sip.c:8742 handle_response_register: Outbound Registration: Expiry for gvt.com.br is 60 sec (Scheduling reregistration in 45 s)" I?m using asterisk-1.2.0-beta and the sip.conf parameters about registration: defaultexpirey=1200 registertimeout=1200 There is any way to make asterisk follow the 1200 seconds I?m trying to tell? Could be something happening out of my unit but at the provider network? Thanks in advance, Ricardo Poppi.

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 but still have some...

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 == Extensio...

...gt;' I dont know what is the problem. 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=dyna...

...ect.sipgate.de/USER [sipconnect.sipgate.de] type = friend host = sipconnect.sipgate.de outboundproxy = sipconnect.live.sipgate.de port = 5060 username = USER fromuser = USER fromdomain = sipconnect.sipgate.de secret = PASS dtmfmode = rfc2833 insecure = port,invite ;insecure = very canreinvite = no registertimeout = 600 disallow = all allow = alaw allow = ulaw context = sipconnect.sipgate.de Auntie google is not very helpful for me. They all say, it looks like a firewall problem on the router. But I'm sure this is set up correctly. Any ideas? -- Michael Herrmann

...name?????? 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 insecure=port,invite co...

...[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...

...3 to 202. 103 happens to be the last listed in sip.conf and the first listed in 'sip show peers' (I have confirmed that this is dependent on the order in the conf file, not numeric order) sip.conf :- [general] port = 5060 bindaddr = 0.0.0.0 pedantic = no autocreatepeer = no context = sip registertimeout=20 localnet = 10.10.10.0/255.255.255.0 srvlookup = yes tos=0xb8 rtptimeout=300 rtpholdtimeout=1800 maxexpirey=3600 defaultexpirey=1200 [sip-101] ; Aastra 480i phones for general office type=peer insecure=very disallow=all allow=ulaw allow=alaw host=dynamic dtmfmode=auto canreinvite=no context=offi...

...ng realm=etatcritik.dyndns.org bindport=5060 bindaddr=0.0.0.0 srvlookup=no maxexpiry=3600 defaultexpiry=1800 videosupport=yes disallow=all allow=ulaw allow=ilbc allow=alaw allow=gsm musicclass=default language=fr useragent=Asterisk PBX dtmfmode = auto register => 09XXXXXXXX:SECRET@freephonie.net registertimeout=40 externip = 82.XXX.XXX.XXX localnet=10.XXX.XXX.XXX/255.255.255.0 qualify=60000 nat = yes [test] type=friend username=test secret=test host=dynamic context=home callerid =test <2222> dmtfmode=rfc2833 authuser=test fromuser=test allow=all [freephonie_outbound] type=peer allow=all host=freepho...

....0/255.255.255.0 externip=195.xxx.xxx.xxx srvlookup=yes [authentication] [eutelia-out] ;maxexpirey=360000 ;defaultexpirey=180000 type=friend allow=alaw context=inbound username=xxxx secret=xxxxx fromuser=number fromdomain=voip.eutelia.it host=voip.eutelia.it dtmfmode=inband realm=voip.eutelia.it registertimeout=300 canreinvite=no ;registertimeout=9999999999 qualify=200 insecure=very ,allow=alaw ,allow=ulaw ,allow=gsm [messagenet-out] auth=user:password at sip.messagenet.it ;auth=md5 realm=sip.messagenet.it qualify=yes ;maxexpirey=360000 ;defaultexpirey=180000 authname=user authuser=user canreinvite=no co...

...onstandard = 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 => 6087294351:<sip...

...no > qualify=yes > context=incoming > type=friend > insecure=invite,port > fromdomain=sip.internode.on.net > host=sip.internode.on.net > dtmfmode=rfc2833 > disallow=all > allow=alaw > allow=ulaw > allow=g729 > bindport=5060 > bindaddr=0.0.0.0 > nat=yes > registertimeout=5 > allowoverlap=no > srvlookup=no > ubscribecontext=from-sip > callcounter=yes [Company2] > ... > [Company3] > ... > [Company4] > ... And here is some of the "extensions.conf" file: [incoming] > ; Get the DID number from the TO header. > exten =>...

...Enable DNS SRV lookups on outbound calls disallow=all ; First disallow all codecs ;allow=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

...one calls to our clients sound muffled, like they are talking underwater. Reported for both the Snom 870, and the polycom ip650. Incoming calls sound ok. Could this be a codec problem? My dialplan looks like: [general] port = 5060 bindaddr = 0.0.0.0 srvlookup = no tos_sip = cs7 tos_audio = ef registertimeout = 1 relaxdtmf = yes context = testofidea disallow = all ;allow=gsm allow = ulaw allow = alaw allow = g722 dtmfmode=rfc2833 ;; allows use of pushbuttoms ;dtmfmode = inband nat = no localnet = 10.0.0.0/255.0.0.0 canreinvite = no Thanks for any help. Best Eddie -- Eddie H. Mikell Senior Systems...

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

...d subscriptions (seconds) minexpiry=60 ; Minimum length of registrations/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 reg...

...de.on.net <http://sip.internode.on.net> > host=sip.internode.on.net <http://sip.internode.on.net> > dtmfmode=rfc2833 > disallow=all > allow=alaw > allow=ulaw > allow=g729 > bindport=5060 > bindaddr=0.0.0.0 > nat=yes > registertimeout=5 > allowoverlap=no > srvlookup=no > ubscribecontext=from-sip > callcounter=yes > > [Company2] > ... > [Company3] > ... > [Company4] > ... > > And here is some of the "extensions.conf" file: > > [inco...