i tried it and it wont work with rtcachefriend=yes On Fri, Feb 10, 2012 at 11:56 PM, JR Richardson <[email protected]>wrote:
> > I am facing an issue with Peer registration in my asterisk server . > > > > I am using asterisk version 1.8.5.0 and using SIP real-time > > architecture.when i am doing registration it registered fine on asterisk > > as peer is available in Database. > > > > But now i am doing 'sip reload' or 'reload' due to some reason my peer > > registration is going out and i cannot able to call that peer even though > > in SIP client it shows me 'registered'. > > > > Can any body elaborate on this issue which settings i need to put in > > sip.conf. > > > > I also tried to follow this patch > > https://issues.asterisk.org/view.php?id=14196 But it allready applied in > > code base so why it wont work? > > > > Here is my sip.conf settings. > > > > [general] > > context=from-internal ; Default context for incoming cal > > rtcachefriends=no > > rtupdate=yes > > rtautoclear=yes > > rtsavesysname=yes > > callcounter = yes > > callevents=yes > > bindport=5060 ; UDP Port to bind to (SIP standard port is > 5060) > > srvlookup=yes ; Enable DNS SRV lookups on outbound calls > > pedantic=yes ; Enable slow, pedantic checking for Pingtel > > tos=184 ; Set IP QoS to either a keyword or numeric val > > tos_sip=cs3 ; Sets TOS for SIP packets. > > tos_audio=ef ; Sets TOS for RTP audio packets. > > tos=lowdelay ; lowdelay,throughput,reliability,mincost,none > > maxexpiry=3600 ; Max length of incoming registration we allow > > defaultexpiry=120 ; Default length of incoming/outoing > registration > > preferred_codec_only=yes > > disallow=all ; First disallow all codecs > > allow=ulaw ; Allow codecs in order of preference > > allow=alaw > > insecure=invite > > language=en ; Default language setting for all > > users/peers > > rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP > > activity > > useragent=dhaval ; Allows you to change the user agent > string > > dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. > Default: > > rfc2833 > > qualify=yes > > nat=yes > > ;canreinvite=yes > > directmedia=yes > > directrtpsetup=yes > > > > And here is DB fields snapshots. > > > > id: 1 > > name: 201 > > ipaddr: 172.18.100.243 > > port: 53624 > > regseconds: 1328716180 > > defaultuser: 201 > > fullcontact: NULL > > regserver: dhaval > > useragent: CSipSimple r1133 / b > > lastms: 554 > > host: dynamic > > type: friend > > context: from-internal > > permit: NULL > > deny: NULL > > secret: 201 > > md5secret: NULL > > remotesecret: NULL > > transport: NULL > > dtmfmode: NULL > > directmedia: yes > > nat: NULL > > allow: ulaw > > disallow: g729 > > insecure: invite > > callerid: NULL > > rfc2833compensate: NULL > > mailbox: NULL > > session-timers: NULL > > session-expires: NULL > > session-minse: NULL > > session-refresher: NULL > > > > Kindly help me to resolve this. > > > > Thanks > > Dhaval > > > > The first thing I would try is 'rtcachefriends=yes', that should do it. > > JR > -- > JR Richardson > Engineering for the Masses > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
