search for: minexpiry

...e-register quickly. - 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. -------------- n...

...s 5060) bindaddr = 0.0.0.0 ?; ? ? ? ? ? ? ?IP address to bind to (0.0.0.0 binds to all) srvlookup = yes ?; ? ? ? ? ? ? ?Enable DNS SRV lookups on outbound calls context=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip=72.220.28.226 localnet=192.168.0.0 nat=yes maxexpiry=15 minexpiry=14 ;rtautoclear=no ;autofallthrough=yes register =><did>:<password>@69.59.234.67:5060/202 [vonage-out] username=<did> type=friend secret=<password> port=5061 nat=yes host=69.59.234.67 fromuser=<did> fromdomain=69.59.234.67 dtmfmode=rfc2833 auth=md5 context=from-ps...

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 problems: - from time to time...

...p = yes ; Enable DNS SRV lookups on outbound calls > > context=incoming > > disallow=all > > allow=ulaw > > allow=alaw > > allow=g729 > > allow=g723 > > externip=72.220.28.226 > > localnet=192.168.0.0 > > nat=yes > > maxexpiry=15 > > minexpiry=14 > > ;rtautoclear=no > > ;autofallthrough=yes > > > > register =><did>:<password>@69.59.234.67:5060/202 > > > > [vonage-out] > > username=<did> > > type=friend > > secret=<password> > > port=5061 > > nat=...

...SRV lookups on outbound calls context=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip=72.220.28.226 localnet=192.168.0.0 nat=yes maxexpiry=15 minexpiry=14 ;rtautoclear=no ;autofallthrough=yes register =>16194077214:<<password>@69.59.234.67:5060/202 [authentication] [3000] type = friend context = default...

...= yes alwaysauthreject = no autodomain = no callevents = no canreinvite = yes checkmwi = 10 compactheaders = no defaultexpiry = 120 dumphistory = no externip = 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_...

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

...rt is 5060)bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)srvlookup = yes ; Enable DNS SRV lookups on outbound callscontext=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip=72.220.28.226 localnet=192.168.0.0 nat=yes maxexpiry=15minexpiry=14;rtautoclear=no;autofallthrough=yes register =>16194077214:<<password>@69.59.234.67:5060/202 [authentication][3000]type = friendcontext = defaultusername = 3000host = dynamicmailbox = 3000dtmfmode = rfc2833[3001]type = friendcontext = defaultusername = 3001host = dynamicmailbox = 3001...

...(0.0.0.0 binds to > all) > srvlookup = yes ; Enable DNS SRV lookups on outbound calls > context=incoming > disallow=all > allow=ulaw > allow=alaw > allow=g729 > allow=g723 > externip=72.220.28.226 > localnet=192.168.0.0 > nat=yes > maxexpiry=15 > minexpiry=14 > ;rtautoclear=no > ;autofallthrough=yes > > register =>16194077214:<<password>@69.59.234.67:5060/202 > > [authentication] > [3000] > type = friend > context = default > username = 3000 > host = dynamic > mailbox = 3000 > dtmfmode = rfc2833 &g...

I have logs full with this messages... I must have qualify turned on, because phone is behind firewall, main problem si, that phone is each hour about one hour unavailable! :'( I tried to modify minexpiry/maxexpiry sip.conf timeouts, but nothing help me. I'm using latest firmware 8.4 in phone, will be better to downgrade? to what version? (latest asterisk 1.4branch) [Dec 7 00:36:56] NOTICE[19226] chan_sip.c: Peer '108' is now UNREACHABLE! Last qualify: 205 [Dec 7 01:36:23] NOTICE...

Hi, I've noticed that if I disconnect or reconnect a phone from the net, Asterisk take long time to realize that (even more then 10 minutes). Is there a way to reduce this time, working on the configuration files? Thank you. silvia ------------------------------------------------------ Passa a Infostrada. ADSL e Telefono senza limiti e senza canone Telecom http://click.libero.it/infostrada

Hi All, I am having some problems with Asterisk on static IP and Sipura-1001 on dynamic IP. Is there any solutions to in the Asterisk configuration or Sipura-1001 to re-register when the router change IP dynamic IP? Thanks. Regards, Kengie -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have 8 aastra phones that are loosing registration. On the phone gui it says 408 as the registation error after a minute or say they register. In the cli it eill say the phone is now unreachable then it will show it registering then available. At first they did it every hour all the phones. After messing with the experation it does it every 15 nin. Any ideas on how to troubleshoot this? I tried

Hello list, I have in sip.conf : /maxexpiry=60 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) defaultexpiry=120 ; Default length of incoming/outgoing registration ;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------ registertimeout=240 ; r...

Am 23.06.2020 16:22, schrieb Marek Greško: > It seems your problems lie in something other. Most probably it is not > mtu problem. All my suspections are contradicted. If it is true you > have inter vlan voice quality problems, it is definitely something > different. Formerly I assumed you were trying only LTE vs LAN using > internet. I'm not sure what you mean with the last

...lient code. The menu works fine, but sometimes I have DTMF duplication that prevent proper code entry. All DTMF come twice. my sip.conf ----------- [general] context=default allowguest=no allowoverlap=no allowtransfer=yes udpbindaddr=0.0.0.0 tcpenable=no tlsenable=no srvlookup=yes maxexpiry=3600 minexpiry=60 defaultexpiry=120 qualifyfreq=60 disallow=all allow=alaw language=fr relaxdtmf=no dtmfmode=rfc2833 videosupport=no dynamic_exclude_static=yes canreinvite=no rtcachefriends=yes rtsavesysname=yes rtupdate=yes rtautoclear=yes ignoreregexpire=no [xxxxx] type=peer host=xxxxxx.xxxxxxx.com secret=XxXx...

...speex allow=g722 allow=h264 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 can...