Asterisk realtime + load balancing

yasir.hi.yagi · July 10, 2007, 9:00pm

I have two asterisk servers (version 1.4.4 and 1.4.6) running Realtime ARA+mysql and all my users are sip users . the users can register alright but only users registered in the same server can call each others, if a user registered in server A tries to call a user registered in server B, he gets no such host warning in the asterisk CLI. When a user get registered his info is plugged in the sip_buddies table except for the “full contact” field remains empty and when the user log off I can see his ip address changed to 0.0.0.0 in the database. when I do “sip show peers” I get nothing and the same goes for “sip show peer xxx”.
my extconfig.conf as follow:

extensions.conf => mysql,voiceone,ast_config
features.conf => mysql,voiceone,ast_config
iax.conf => mysql,voiceone,ast_config
meetme.conf => mysql,voiceone,ast_config
misdn.conf => mysql,voiceone,ast_config
musiconhold.conf => mysql,voiceone,ast_config
queues.conf => mysql,voiceone,ast_config
sip.conf => mysql,voiceone,ast_config
zapata.conf => mysql,voiceone,ast_config
iaxusers => mysql,voiceone,iax_buddies
iaxpeers => mysql,voiceone,iax_buddies
sipusers => mysql,voiceone,sip_buddies
sippeers => mysql,voiceone,sip_buddies
voicemail => mysql,voiceone,voicemail_users
extensions => mysql,voiceone,extensions_table

I am not using realtime extensions for now i.e. the extensions_table is empty.

to make long short, I want to run two or more servers from the database where the user can register at any server and be able to call any user registered at any other server.
HELP PLEASE.
Regards

yasir.hi.yagi · July 11, 2007, 10:42am

just to add here is output from the asterisk CLI when a call happens:

– Executing [1002@DefaultOutgoingRule:1] AGI(“SIP/1003-081be338”, “dial.php|entity=55&group=2&extension=1002”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dial.php
– AGI Script Executing Application: (macro) Options: (stdexten|SIP/1002)
– Executing [s@macro-stdexten:1] Set(“SIP/1003-081be338”, “EXTENSION=1002”) in new stack
– Executing [s@macro-stdexten:2] GotoIf(“SIP/1003-081be338”, “0?DefaultOutgoingRule||1”) in new stack
– Executing [s@macro-stdexten:3] GotoIf(“SIP/1003-081be338”, “0?s-BUSY|1”) in new stack
– Executing [s@macro-stdexten:4] Dial(“SIP/1003-081be338”, “SIP/1002|20|gtTwW”) in new stack
[Jul 11 13:27:59] WARNING[4397]: app_dial.c:1106 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-stdexten:5] Goto(“SIP/1003-081be338”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-stdexten,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-stdexten:1] Goto(“SIP/1003-081be338”, “s-NOANSWER|1”) in new stack
– Goto (macro-stdexten,s-NOANSWER,1)
– Executing [s-NOANSWER@macro-stdexten:1] GotoIf(“SIP/1003-081be338”, “0?DefaultOutgoingRule||1”) in new stack
– Executing [s-NOANSWER@macro-stdexten:2] GotoIf(“SIP/1003-081be338”, “0?skip-vm”) in new stack
– Executing [s-NOANSWER@macro-stdexten:3] MailboxExists(“SIP/1003-081be338”, “1002”) in new stack
– Executing [s-NOANSWER@macro-stdexten:4] NoOp(“SIP/1003-081be338”, “MailboxExists(1002): SUCCESS”) in new stack
– Executing [s-NOANSWER@macro-stdexten:5] GotoIf(“SIP/1003-081be338”, “1?:skip-vm”) in new stack

also here is the output for "realtime load sippeers name 1002"
asterisk2*CLI> realtime load sippeers name 1002
Column Name Column Value
-------------------- --------------------
id 20
name 1002
type friend
fullcontact
regseconds 1184151597
ipaddr x.x.x.x
context DefaultOutgoingRule
secret 1002
callerid mataz yagi <1002>
mailbox 1002
username 1002
host dynamic
port 5064

also here is the output for "sip show peers"
asterisk2*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]

also the output of "sip show peer 1002"
asterisk2*CLI> sip show peer 1002
Peer 1002 not found.

asterisk2*CLI> realtime mysql status
Connected to xxxxx@x.x.x.x, port 3306 with username xxxx for 1 hours, 57 minutes, 51 seconds.

yasir.hi.yagi · July 11, 2007, 11:50am

small correction to my extconfig.conf as follow:

extensions.conf => mysql,voiceone,ast_config
features.conf => mysql,voiceone,ast_config
meetme.conf => mysql,voiceone,ast_config
misdn.conf => mysql,voiceone,ast_config
musiconhold.conf => mysql,voiceone,ast_config
queues.conf => mysql,voiceone,ast_config
iaxusers => mysql,voiceone,iax_buddies
iaxpeers => mysql,voiceone,iax_buddies
sipusers => mysql,voiceone,sip_buddies
sippeers => mysql,voiceone,sip_buddies
voicemail => mysql,voiceone,voicemail_users
extensions => mysql,voiceone,extensions_table

yasir.hi.yagi · July 11, 2007, 12:09pm

asterisk2CLI> sip show settings
asterisk2CLI>

Global Settings:

SIP Port: 5060
Bindaddress: 0.0.0.0
Videosupport: No
AutoCreatePeer: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: No
Promsic. redir: No
SIP domain support: No
Call to non-local dom.: Yes
URI user is phone no: No
Our auth realm asterisk
Realm. auth: No
Always auth rejects: No
Call limit peers only: No
Direct RTP setup: No
User Agent: Asterisk PBX
MWI checking interval: 10 secs
Reg. context: (not set)
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
IP ToS SIP: none
IP ToS RTP audio: none
IP ToS RTP video: none
T38 fax pt UDPTL: No
RFC2833 Compensation: No
SIP realtime: Enabled

Global Signalling Settings:

Codecs: 0x8000e (gsm|ulaw|alaw|h263)
Codec Order: none
T1 minimum: 100
Relax DTMF: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No

Default Settings:

Context: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: (Defaults to English)
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk

Realtime SIP Settings:

Realtime Peers: Yes
Realtime Users: Yes
Cache Friends: No
Update: Yes
Ignore Reg. Expire: No
Save sys. name: No
Auto Clear: 120

supertle · July 13, 2007, 3:28pm

see my thread, you should search first also
i just added this into my sip.conf

rtcachefriends=yes
This will now allow you to see your sip users configured in ARA after they have registered when you run “sip show users”