cluster setup

[richardroi]

On my setup 1 db+web server and 1 telephony server, when I checked the recordings it is coming from telephony's ip, is this correct?

Thank you.

[richardroi]

I was redoing my telephony server, Now I'm having this error " Telephony server already exist, Sever was previously installed", how to correct this? and what if I will add one more telephony server, Is this issue will come up again?

[richardroi]

figured it out. You have to delete the server on main gui server.
How about adding new telephony server?

[rrb555]

install the same method you did with telephony 1

[richardroi]

ok will try again.

When im connecting on ftp
" slave:~ # ftp 192.168.150.153
ftp: connect: Connection refused" i have this error!

then on my mail " Can't call method "login" on an undefined value at /usr/share/astguiclient/AST_CRON_audio_3_ftp.pl line 447."

Firewall is disabled!

[williamconley]

did you put the ftp credentials in during installation or in astguiclient.conf after installation?

Without credentials, ftp won't work.

It is also required that ftp be installed and working properly.

So ... for a place to begin, is ftp installed in the server you want to "receive" the files?

[richardroi]

the configuration of ftp on archive server was configured during the installation. Was trying to redo the archive but it is saying " archive server found, cannot setup two " what to do with it. want to try again. this happen also when i redo the telephony server. after installation before pressing reboot, error"duplicate entry", any advice here. thank you.

[richardroi]

I addded new telephony server. Installation went fine, the issue is, the new telephony server cannot use the trunk, and when i tried calling the extension on the other telephony it is saying " Call from '120' (192.168.150.115:15882) to extension '103' rejected because extension not found in context 'default'", they should connect each other right? ext to ext.
I put ips on the " active twin ip" still no go.

[richardroi]

Trunk now working. Copy the trunk and point to the 2nd telephony server ip.
The only issue now is calling ext to ext.

Thank you.

[richardroi]

anyone? extension to extension calling still no go.
and one more thing, every time I redo a Telephony server with the same ip, I'm having this error " error duplicate 192.XXX.XXX.XXX entry serverconf"

Thank you.

[williamconley]

When you add a new dialer, you must also add a new carrier and assign the carrier to the new dialer.

[richardroi]

Carriers no issue, calling extension from dialer A to dialer B is the issue. they cannot call each other.

[geoff3dmg]

Is 'Generate Cross-Server Phone Extensions' set to '1' in Admin -> System settings? (Bear in mind you can not have duplicate extension numbers on different diallers if you enable this)

[richardroi]

Thank you geoff3dmg. already tried that. still no go.

[williamconley]

will need a sample output from an attempt (from asterisk cli).

[richardroi]

-- Executing [102@default:1] Dial("SIP/120-00000004", "SIP/gs102,60,") in new stack
[Feb 28 22:52:08] WARNING[4496]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Feb 28 22:52:08] == Everyone is busy/congested at this time (1:0/0/1)
[Feb 28 22:52:08] -- Executing [102@default:2] Goto("SIP/120-00000004", "default,85026666666666102,1") in new stack
[Feb 28 22:52:08] -- Goto (default,85026666666666102,1)
[Feb 28 22:52:08] -- Executing [85026666666666102@default:1] Wait("SIP/120-00000004", "1") in new stack
[Feb 28 22:52:09] -- Executing [85026666666666102@default:2] VoiceMail("SIP/120-00000004", "102,u") in new stack
[Feb 28 22:52:09] -- <SIP/120-00000004> Playing 'vm-theperson.gsm' (language 'en')
[Feb 28 22:52:10] -- <SIP/120-00000004> Playing 'digits/1.gsm' (language 'en')
[Feb 28 22:52:11] -- <SIP/120-00000004> Playing 'digits/0.gsm' (language 'en')
[Feb 28 22:52:12] -- <SIP/120-00000004> Playing 'digits/2.gsm' (language 'en')
[Feb 28 22:52:12] -- <SIP/120-00000004> Playing 'vm-isunavail.gsm' (language 'en')
[Feb 28 22:52:14] -- <SIP/120-00000004> Playing 'vm-intro.gsm' (language 'en')
[Feb 28 22:52:15] == Spawn extension (default, 85026666666666102, 2) exited non-zero on 'SIP/120-00000004'
[Feb 28 22:52:15] -- Executing [h@default:1] AGI("SIP/120-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Feb 28 22:52:15] -- <SIP/120-00000004>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Feb 28 22:52:25] > Refreshing DNS lookups.


Sir: extension on server A can call extension on server B, but not the other way around. 'Generate Cross-Server Phone Extensions' set to '1'

[richardroi]

Created new phone, but not working on both now when we used the new phone logins. cannot call each other again. but when we return it to the previous phones, server A can call server B but still not the no go calling back from B.
[Mar 1 00:27:22] NOTICE[1140]: chan_sip.c:23510 handle_request_invite: Call from '102' (192.168.150.116:9420) to extension '121' rejected because extension not found in context 'default'.

[williamconley]

In which case you'll need to find out why it is trying to dial 121 when 121 is apparently not a dialable extension. Do you have a phone that has a "Dial Plan Number" of 121? Is there an entry in extensions-vicidial.conf for "exten => 121"?

It's not magic, this all has a method to make the entries and when it all works you can dial. One piece missing fails. Find the piece (did the phone not get saved to the file? does the phone have a different dialplan number? did asterisk not "reload" to recognize the new file?)

[richardroi]

got it working. we deleted all the phones and added new ones....Thank you for everything.

"AUTO ftp of database to an ftp server" --- not yet working. can't make the transfer. only recordings are transferring...

[williamconley]

have you read the manual for setting this up? Or just "takin a shot at it" on the forums? LOL

[richardroi]

You will need to have (and enter) the following information:
# REPORT server connection information
VARREPORT_host => 10.0.0.4
VARREPORT_user => cron
VARREPORT_pass => test
VARREPORT_port => 21
VARREPORT_dir => REPORTS
This information is used in the following scripts (each script has a “help” option for how to use it):
 ADMIN_backup.pl
 AST_email_web_report.pl
 AST_sourceID_summary_export.pl
 AST_VDlist_summary_export.pl
 AST_VDsales_export.pl
 VICIDIAL_IN_new_leads_file.pl
 Vtiger_IN_new_accounts_file.pl
 Vtiger_OUT_sync2VICIDIAL_file.pl


im still lost on backing up dbase on archive server!

[richardroi]

solved. Just edit the credentials on astguiclient (reports). Thanks William.

[richardroi]

Question: On cluster setup 1 dbase/web server + 2 telephony server, which of the two should have public ip to access recordings outside? will be using 2 nics.

[boybawang]

the 2 asterisk servers is recommended to have 1 local ip and 1 public ip so yes 2 nics per asterisk server, the database server must be on the local network only to prevent intrussion

[richardroi]

Thank you boybawang!

[richardroi]

question: iburst keyword, how to use?

[williamconley]

server xxx.xxx.xxx.xxx iburst