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.