[asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug.

Ishfaq Malik ish at pack-net.co.uk
Fri Feb 10 02:50:56 CST 2012


Hi

To the best of my understanding this is the correct behaviour. When you
add a peer to the database and a device configured for that peer
registers, it enters that peer into the RealTime cache.

When you do a sip reload you fully clear that RealTime cache so the
asterisk process will lose knowledge of that peer until it registers
again and gets re entered into the RealTime cache, which most SIP phones
are set to do after a number of minutes.

The real question here is, if you are using RealTime architecture for
your peers, why are you doing a sip reload?



On Fri, 2012-02-10 at 10:55 +0530, DHAVAL INDRODIYA wrote:
> nobody facing any issue with this or nobody using real time
> architecture
> 
> On Thu, Feb 9, 2012 at 10:54 AM, DHAVAL INDRODIYA
> <dhaval.it01034 at gmail.com> wrote:
>         Hi Group.
>         
>         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
>         
> 
> --
> _____________________________________________________________________
> -- 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

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062




More information about the asterisk-users mailing list