[asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug
Ishfaq Malik
ish at pack-net.co.uk
Wed Feb 15 02:42:31 CST 2012
Hi
Nothing will stop the behaviour you are seeing. A SIP reload will clear
the realtime cache thus stopping the asterisk server knowing where the
realtime sip endpoint is until the endpoint re-registers.
The question here is, why are you doing SIP reloads? Once you are using
RealTime architecture for SIP, sip reloads become unnecessary unless you
are making modifications to the general section of your sip.conf and why
would you need to do that regularly?
Regards
Ish
On Wed, 2012-02-15 at 12:52 +0530, DHAVAL INDRODIYA wrote:
> 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
>
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Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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