JR Richardson
2012-Feb-10 18:26 UTC
[asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug
> 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
DHAVAL INDRODIYA
2012-Feb-15 07:22 UTC
[asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug
i tried it and it wont work with rtcachefriend=yes On Fri, Feb 10, 2012 at 11:56 PM, JR Richardson <jmr.richardson at gmail.com>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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120215/965d183e/attachment.htm>