search for: defaultexpiry

Hi, I've got several trunks in my 1.6.0.1 setup. One of them is asking for 1800 sec registrations. You can provide this value setting defaultexpiry to 1800 in sip.conf but how can you specify this duration to this specific trunk and not affect the others ? An option to register statement in sip.conf would be perfect ... Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/as...

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.

...ll my users will re-register too often. I 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...

...is 63e8aedc77335879c93123055d21211d Would this value match what chan_sip would pass as the md5 credentials? Our sip.conf looks like the following... [general] context = NECTEST bindaddr = 0.0.0.0 bindport = 5060 websocket_enabled = false srvlookup = no allowguest = yes debug = yes sipdebug = yes defaultexpiry = 480 deny = 0.0.0.0/24 permit = 10.100.102.0/24 permit = 192.168.9.0/24 canreinvite = yes callcounter = yes register = 3016:3016 at 10.100.102.82:5060/3016 [3016] type = friend qualify = no nat = no host = 10.100.102.82:5060 defaultuser = 3016 secret = 3016 incominglimit = 24 accountcode = 33 por...

...ice 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: - from time to time peers have lagged...

...unately, 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 just to get the sta...

...ved=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 qualify=60000 nat =...

...8121281- Max-Forwards: 70 Content-Length: 0 Here's the configs: subscribecontext = device-hints allowexternaldomains = yes allowguest = yes allowsubscribe = 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 rea...

I do not want set allowguest=yes. The problem is, there is no official list with ip addresses of Telekom Germany. But I think all ip addresses comes from the ip range 217.0.0.0/13. I have now the following addition to sip.conf. I think it is the only safe option. Or what would you say? [telekom](!) context=from-trunk type=peer defaultuser= authuser= remotesecret= fromdomain=tel.t-online.de

...y. Looking at the debug log the codec used was miscellaneously numbered 0, 4 and 8. I thought I'd disabled 4 (g.723) but it appears not. My sip.conf has this: general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = voip-sip defaultexpiry = 3600 register => 12345:secret@fwd.pulver.com/39 disallow=all allow=alaw allow=ulaw I was expecting this would stop g.723 from being even tried - am I missing something? Is there any config option for SJphone that clobbers g.723? Iain

...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] type=friend usernam...

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 have around 7 ses...

Guys, when registering I am getting this error message, my question is that if this could be the reason whay I am able to make calls but not to recieve call ? [Apr 3 11:24:31] WARNING[19578]: chan_sip.c:15104 handle_response_register: Got 423 Interval too brief for service +506phonenumber at domain.co.cr@host.ip.addr, minimum is 3600 seconds Thanks -- http://celord.blogspot.com/

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;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 -------------- An embedded and charset-un...

Hi All, I am running asterisk on Linux machine and trying to use confbridge application. Please have a look at Conf files. sip.conf ====== [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow = all allow=ulaw allow=alaw defaultexpiry=100 [5001] type=friend nat=yes host=dynamic canreinvite=no context= conferences disallow = all allow=ulaw allow=alaw [5002] type=friend nat=yes host=dynamic canreinvite=no context= conferences disallow = all allow=ulaw allow=alaw [5003] type=friend nat=yes host=dynamic canr...

Hi, Are sip.conf's defaultexpiry and maxexpiry global? Or can they be used on a per-extension basis? I'd like to "force" some extensions to re-register more frequently than others (server-side). Thanks, Vieri

...ut I am not able to determine until now. Any help? By the way: which paramter in the sip.conf can be used to determine the timeout of the sip registration (so the IP Phone should send the registartion packet to keepalive within this timeout, otherwise it will be considered not register)? Is it the defaultexpiry or something else? Also, the above warning, to what it could be related? Is it a bug? Regards Bilal

...ot configured? My sip.conf [general] context = default allowguest = no bindport = 5060 bindaddr = 0.0.0.0 allowexternaldomains = no allowoverlap = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = no checkmwi = 10 compactheaders = no defaultexpiry = 120 domain=sop-korniychuk domain=172.30.8.13 domain=172.30.8.13:5060 dumphistory = no externrefresh = 10 g726nonstandard = no notifyringing = yes srvlookup = yes t1min = 100 t38pt_udptl = no ;tos_audio = none ;tos_sip = none ;tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone =...