search for: maxexpiry

Hello is it possible to determine the SIP.conf parameters 'defaultexpirty' and 'maxexpiry' on a peer basis ? My default value is 300 seconds, but some specific SIP-clients can only send a SIP REGISTER every 3600 seconds. In current configuration these SIP peers now become "Unreachable" after 300 seconds. Or is there another way to differentiate ? Kind regards. ---...

...want that only lagged users to re-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 a...

...andard port 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 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 co...

...On the SIP device it shows I am RESISTED but when I do "sip show peers" it shows my sip endpoints are "UNREACHABLE". And it keeps on flapping "Peer '9099993456' is now UNREACHABLE!" and "Peer '9099993456' is now REACHABLE!"... I changed my maxexpiry and defaultexpiry to 3600 in sip.conf but still it didn't help. I am using Asterisk 1.2.18 with Real-Time config. Any help will be appreciated... Cheers, Nitesh

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: - fr...

...standard port 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 = dynamicma...

...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> > > p...

Hi, I'm getting this error when registering with SIP server using Asterisk 1.4.10 and Freepbx... I'm getting this error no matter what I try to setup in sip.conf : - I'm getting confused whether options are maxexpirey=36000 or maxexpiry=36000 ? - Can I solve this with some settings in sip.conf or is this problem harder ? - I've read something about Asterisk's bug on this error, but am not sure it really patching is necessary or can be avoided with different settings ? Thanks in advance, regards, Rob. --- (10 headers...

...XX: 5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '793bf24e5290d562787c8d9451baedd7@82.XXX.XXX.XXX' sip.conf [general] context=incoming 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 q...

...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 c...

...yes allowtransfer = 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 to...

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

...ddress 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 =>16194077214:<<password>@69.59.234.67:5060/202 > > [authentication] > [3000] > type = friend > context = default > username = 3000 > host = dynamic > mailbox = 3000 > dtm...

...l in on my voip in number i get rejected. if i use Sipura without asterisk i get in calls here is my sip.conf ---------------------------------------------- [general] useragent=nedi port=5060 context=default ;tos=lowdelay disallow=all allow=ulaw allow=alaw allow=gsm allow=g726 language=de maxexpiry=50 defaultexpiry=30 register => user:passw@sip.voipstunt.com/user [useruser] type=friend username=user secret=passw host=sip.voipstunt.com fromdomain=sip.voipstunt.com canreinvite=yes insecure=very nat=yes context=incomingsip.voipstunt.com dtmfmode=rfc2833 stun=stun.voipstunt.com:3478 [13]...

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[19226] ch...

I'm having trouble with Polycom 501 phones that asterisk forgets how to reach them. /etc/asterisk/sip.conf: [general] context=default MusicOnHold=default port=5060 bindaddr=0.0.0.0 srvlookup=no;yes language=en dtmfmode=rfc2833 maxexpiry=600 defaultexpiry=120 [502] type=friend username=502 secret=pass host=dynamic mailbox=502@rm callerid= "Operator" <502> context=rm dtmfmode=rfc2833 accountcode= setvar=DINTERNAL=1 In extensions.conf I have hints setup that is monitored from a 601 with the expansion module. I also...

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

I have a 536i expansion module attached to a 57i-CT. The BLF lights on the 536i will light up and work fine for a while... however after a bit they seem to loose their ability to see if someone is on a phone. They still work to dial, if I try to dial, however, they don't light up when someone makes a call, or if their phone rings. If I reboot the phone, the lights start working again

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:

...39; Native Bridging it's same problem. it's sip module bug ?? When capturing with wireshark, at the beginning of sound file, we see a break in sound. thank you in advance sip conf: [general] port=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no rtcachefriends=yes directrtpsetup=no maxexpiry=300 bridge=yes defaultexpiry=300 useragent=toto PJ: shema of call with wireshark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090626/a978406c/attachment.htm -------------- next part ------------...