search for: minse

I tried to use R version of package I noticed the original MatLab Pckage is much better documented. I had a look at the R demo code "gtm_demo" and found that variable Y is used in advanced of being created: I wrote my own few lines as follows: inDir <- "C:/Documents and Settings/Monville/Alanine Dipeptide/DBP1/DHA" setwd(inDir) T <-

Hi all... Does anyone know if it is possible to override sip.conf settings in extensions.conf (for example: session-minse=90) without needing to create an overarching peer in sip.conf and selecting it specifically in the dial plan? I'm on the 1.4 stable code base and looking to implement session-timers on certain call flows in a modular dial plan. Thanks, Josh Fuller josh.fuller at telus.com The views express...

Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username

...tbound calls limitonpeers=yes notifyringing=yes rtupdate=yes[authentication] [104] type=peer context=phones host=dynamic fromuser=104 secret=xxxxxx canreinvite=update directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=90 session-minse=120 session-refresher=uac register => 104:xxxxx at xxxxxx.com/104 defaultip=192.168.xx.xxx mailbox=104 disallow=all allow=ulaw,alaw artcachefriends=yes notifyhold=yes incominglimit=1 call-limit=3 Other information will be provided as asked for. Thanks in advance for any help...

...setup works well on my * > [104] > type=peer > context=phones > host=dynamic > fromuser=104 > secret=xxxxxx > canreinvite=yes > directrtpsetup=no > call-limit=3 > nat=yes > qualify=yes > register=no > session-timers=accept > session-expires=60 > session-minse=120 > session-refresher=uac > register => 104:xxxxx at yyyyyy.com/104 > defaultip=192.168.23.114 > mailbox=104 > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James Lamann...

..... but did not go through. Asterisk Side: sip.conf sample: [GENERAL] notifyringing=yes notifyhold=yes limitonpeers=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes t38pt_udptl = yes bindport=5070 externip=SERVER_IP rtptimeout=60 session-timers=originate session-expires=600 session-minse=90 session-refresher=uas rtpholdtimeout=120 rtpkeepalive=20 allow=gsm t38pt_udptl=yes sendrpid=yes trustrpid=no directrtpsetup=yes [USERNAME] deny=0.0.0.0/0.0.0.0 type=friend secret=PASSWORD qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes host=dynamic dtmfmode=rfc2833 disallow=all allow=gsm c...

...lerid: NULL amaflags: NULL callcounter: NULL busylevel: NULL allowoverlap: NULL allowsubscribe: NULL videosupport: NULL maxcallbitrate: NULL rfc2833compensate: NULL mailbox: NULL session-timers: NULL session-expires: NULL session-minse: NULL session-refresher: NULL t38pt_usertpsource: NULL regexten: NULL fromdomain: testers.com fromuser: 660 qualify: NULL defaultip: NULL rtptimeout: NULL rtpholdtimeout: NULL sendrpid: NULL outboundproxy: PU.BL.IC.IP...

Hi, I had a Trixbox 1.4.18 that I "yum update"d to 1.4.22.1. Now, I seem to have a huge problem with phones not staying registered (registrations worked perfectly at 1.4.18). I phone will register the first time I plug it in, and then once you make a call and hangup (or sometimes even during the call) all the lights will go orange meaning a lost registration. Every so often the lights

...manners with our SIP peer. Lastly, in case it matters, the general and peer-specific sections of my sip.conf are as follows: [general] dtmfmode = rfc2833 context=from-voipms srvlookup=yes register => myuserid:mypass at dallas.voip.ms:5060~600 session-timers=refuse session-expires=3600 session-minse=600 session-refresher=uas localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 localnet=169.254.0.0/255.255.0.0 stunaddr=stun01.sipphone.com allow=ulaw allow=gsm [voipms] canreinvite=yes context=from-voipms host=dallas.voip.ms secret=mypass type=friend username=myuse...

...match_auth_username=yes ; use 'authentication username' instead of 'username for authentication' (if available) session-timers=originate ; Request and run session-timers always session-expires=3600 ; maximum session refresh interval session-minse=600 ; minimum session refresh interval session-refresher=uas ; session refresher is user-agent-server ;allowguest=no ; Allow or reject guest calls (default is yes) notifyhold = yes ; Notify subscriptions on HOLD state (default: no...

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