Asterisk realtime doesn't update info automatically

I change the context of the number, the codecs allow and commit the info in the database but doesn’t make the modifications in the truth until I use the reload.

Can anyone help ?

You’ll need to provide more information about your environment, such as the configuration. If you are referring to chan_sip then it can be configured to cache the result, in which case the database is not queried.

I use a cloud server and access with the internet behind the nat.

use:
rtcachefriends=yes
qualify=yes



Global Settings:
----------------
  UDP Bindaddress:        0.0.0.0:5060
  TCP SIP Bindaddress:    Disabled
  TLS SIP Bindaddress:    Disabled
  Videosupport:           No
  Textsupport:            Yes
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        Off
  Match Auth Username:    No
  Allow unknown access:   No
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain support:     No
  Path support :          No
  Realm. auth:            No
  Our auth realm          asterisk
  Use domains as realms:  No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    Yes
  Direct RTP setup:       No
  User Agent:             Asterisk PBX 13.18.5
  SDP Session Name:       Asterisk PBX 13.18.5
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Trust RPID:             No
  Send RPID:              No
  Legacy userfield parse: No
  Send Diversion:         Yes
  Caller ID:              asterisk
  From: Domain:
  Record SIP history:     Off
  Auth. Failure Events:   Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          4294967295
  SIP realtime:           Enabled
  Qualify Freq :          60000 ms
  Q.850 Reason header:    No
  Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS0
  IP ToS RTP audio:       CS0
  IP ToS RTP video:       CS0
  IP ToS RTP text:        CS0
  802.1p CoS SIP:         4
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   6
  802.1p CoS RTP text:    5
  Jitterbuffer enabled:   No

Network Settings:
---------------------------
  SIP address remapping:  Enabled using externaddr
  Externhost:             <none>
  Externaddr:             xxx.xxx.xxx.xxx:0
  Externrefresh:          10
  Localnet:               172.31.0.0/255.255.240.0
                          192.168.0.0/255.255.255.0

Global Signalling Settings:
---------------------------
  Codecs:                 (nothing)
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          Yes
  Compact SIP headers:    No
  RTP Keepalive:          30
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         Yes
  Pedantic SIP support:   Yes
  Reg. min duration       60 secs
  Reg. max duration:      360 secs
  Reg. default duration:  120 secs
  Sub. min duration       60 secs
  Sub. max duration:      360 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Outbound reg. retry 403:No
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      No
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No
  Outb. proxy:            <not set>
  Session Timers:         Accept
  Session Refresher:      uas
  Session Expires:        1800 secs
  Session Min-SE:         90 secs
  Timer T1:               500
  Timer T1 minimum:       100
  Timer B:                32000
  No premature media:     Yes
  Max forwards:           70

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:     UDP
  Context:                incoming
  Record on feature:      automon
  Record off feature:     automon
  Force rport:            Yes
  DTMF:                   rfc2833
  Qualify:                0
  Keepalive:              0
  Use ClientCode:         No
  Progress inband:        No
  Language:
  Tone zone:              <Not set>
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   asterisk
  RTCP Multiplexing:      No

Realtime SIP Settings:
----------------------
  Realtime Peers:         Yes
  Realtime Regs:          Yes
  Cache Friends:          Yes
  Update:                 Yes
  Ignore Reg. Expire:     Yes
  Save sys. name:         Yes
  Save path header:       No
  Auto Clear:             120 (Disabled)

You have enabled caching using the rtcachefriends=yes and thus a reload is required to get the updated information.

but I am with rtcachefriends=yes

Yes, if you have that set to yes then information from the database is cached in memory.

oh, if I put no I cant use qualify=yes and the extension lost connections sometimes, big problem then…

jcolp, have anything to do to solve it without using reload?

That’s how chan_sip works I’m afraid.