search for: defaultexpirey

...job. I contacted Grandstream and although they didn't have a specific answer, they alluded to the fact that lack of registration might be the cause. I performed the following: Increased the "Register Expiration" in the device to 60 minutes Increased "maxexpirey" and "defaultexpirey" to 3800 seconds Configured the device with a static IP address Added defaultip=172.16.1.123 and host=dynamic to sip.conf It has now been running for 48 hours without a single lockup. Here are some of the settings that I am using: In Grandstream: Register Expiration: 60 SIP Registration: Y...

I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop loosing registration? Thanks, Jerry

...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 you can tell me what...

...fault ; Default for incoming calls ;srvlookup = yes ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all ; Disallow all codecs allow=ulaw ; Allow c...

...terval is to long... > > Regards, > Stefan > > > -- > (o_ Stefan Gofferje | Linux Systems >Specialist > //\ Reg'd Linux User #247167 | Network Security >Specialist > V_/_ Linux is like a Wigwam - No gates, no windows, >Apache inside defaultexpirey=120 :Default length of incoming/outoing registration I believe that is the correct option. This site is your friend. Try searching... http://www.voip-info.org/wiki-Asterisk+config+sip.conf

...Any ideas? SIP.conf ====== [general] ; context=incoming-bogus-calls bindport=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) maxexpirey=3600 ; Must be larger than the re-register timeout on the router defaultexpirey=3600 notifymimetype=text/plain rtptimeout=60 rtpholdtimeout=300 disallow=all allow=ulaw ; ; This section is because i'm behind nat ; register=>6477235412:<mypassword>@sip.unlimitel.ca/6477235412 externip=<mystaticIPaddress> ;Outside address localnet=192.168.0.148/255.255.255.0 ;...

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

...cessful response, and 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.

I am getting messages on the console about Registered SIP ... expires 60 How do I increase that 60 to 3 minutes??? I have tried in [general] of sip.conf to set expirey=300 defaultexpirey=300 nothing seems to affect it. Thanks, Jerry

...ce.com/PHONE [Broadvoice] type=friend username=PHONE authuser=PHONE fromuser=PHONE secret=secret host=sip.broadvoice.com port=5060 context=default fromdomain=sip.broadvoice.com canreinvite=no dtmfmode=inband insecure=very permit=sip.broadvoice.com qualify=yes disallow=all allow=ulaw maxexpirey=180 defaultexpirey=160 videosupport=no exten => _9XXXXXXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT) exten => _91XXXXXXXXXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)

...s is what I have in sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = frode.dyndns.org localnet = 192.168.0.0/16 context=stighess tos=lowdelay maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=60 ; Default length of incoming/outoing registratio bandwidth=high disallow=all allow=ulaw ....... [stanaphone] type=friend username=91438xxxx fromuser=91438xxxxx secret=********* host=sip.stanaphone.com context=stighess fromdomain=216.128.82.18 insecure=very nat=yes canreinvite=no di...

...********* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw [1004] type=friend username=1004 secret= reinvite=no canreinvite=no host=dynamic dtmfmode=inba...

Hello, I am trying to REGISTER asterisk to a SIP server, which is listening on Port 6060 (not 5060). The sip.conf file contains register=18474201111:quintum@192.168.2.94:6060/18474201111 maxexpirey=3600 defaultexpirey=120 But the REGISTER message is sent to Port 6060, but the Request-URI still contains, 5060. This is being rejected by SIP server. REGISTER sip:192.168.2.94 SIP/2.0 The request line instead should be like this: REGISTER sip:192.168.2.94:6060 SIP/2.0 See the attached debug logs....

...; SIP compatibility ;tos=lowdelay ; IP QoS parameter, either keyword or value ; like tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow realm=asterisk ; Our global authentication realm ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; Disallow all codecs allow=all ; Allow...

I'm getting these messages every 7-10 seconds. -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP '532' at x.x.x.x port 56988 expires 60 -- Registered SIP '529' at x.x.x.x port 51444 expires 60 -- Registered SIP '529' at x.x.x.x port 64044 expires 60 -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP

...t, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here knows how to set it up ? Some more info: Asterisk CVS-HEAD as of 15-02-2005 My sip.conf [general] context=from-sip realm=vanbaak port=5060 bindaddr=0.0.0.0 srvlookup=yes maxexpirey=3600 defaultexpirey=120 musicclass=default allow=all language=en relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 ;trustrpid = no ;progressinband=no useragent=Asterisk nat=no externip=XXX.XXX.XXX.XXX localnet=192.168.2.0/255.255.255.0 promiscredir = no register => 7304502:my_sipgate_pass@sipgate.de/7304502 register =...

...rityjumping=no [eutelia] include => out_eutelia exten=>_XXXXXXXXXX,1,Dial(SIP/100@out_eutelia,20) exten => _XXXXXXXXXX,2,Hangup 2) sip.conf: [general] context=eutelia realm=voip.eutelia.it port=5060 bindaddr=0.0.0.0 srvlookup=yes defaultexpirey=8600 useragent=Asterisk_Eut localnet=192.168.1.1/255.255.255.0 [out_eutelia] type=peer context=eutelia secret=xxxxxx username=<username> fromuser=<username> fromdomain=voip.eutelia.it host=voip.eutelia.it nat=yes dtmfmode=inband usereqp...

...address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The configuration I have in * is the following: sip.conf ----------- [general] port=5060 context=sip maxexpirey=3600 defaultexpirey=60 disallow=all allow=ulaw allow=gsm [1000] contet=sip type=friend username=1000 secret=????? (not the real one) host=dynamic mailbox=1000 canreinvite=yes dtmfmode=rfc2833 I did not change the above configuration when I moved the budgetTone from the LAN to the Internet (Wan). I am not using a &qu...

...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 calls tos=lowdelay ; lowdelay,throughput,reliability,mincost,none maxexpirey=7200 ; Max length of incoming registration we allow defaultexpirey=3600 ; Default length of incoming/outoing registration videosupport=yes ; Turn on support for SIP video disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw allow=g729 allow=gsm rtcachefriends=yes rtnoupdate=yes rt...

...is 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=office-dial qualify=yes username=101 secret=xxxxxx mailbox=101 callerid="User 1" <101> sip show peers :- 103/103...