vicidial.org

by **ruben23** » Fri Oct 23, 2009 11:22 am

Hi, i have an existing servers for vicidial;

setup is Asterisk/vicidial Server and Apache/Mysql Server.

If i plan to add up Asterisk/Vicidial to load balance the 2 same servers.

questions:

1. Do i have to access two vicidial web interfaces like im having, http://192.168.2.3/vicidial/welcome.php & http://192.168.2.2/vicidial/welcome.php.?

2. My 2 asterisk/vicidial server will be accessing my single apache/mysql server..?

3.Do i setup the 2nd Asterisk/vicidial server like on a single server, what should i run with it. ive already tried it but im confuse how would it function and i follow the steps on the LOAD BALANCE docs.

Please any comment answers.. jut to make it more clearer.. Thanks

by **okli** » Fri Oct 23, 2009 12:33 pm

1) No
2) Yes
3) You need to understand the load balance document. When you hit something in particular which you do not understand, then ask for it specifically. At first glance the document seems confusing, but when you start implementing it it becomes clearer. To begin with - follow it blindly until you make it work. Later on you will start understanding how it works.

by **ruben23** » Fri Oct 23, 2009 1:03 pm

okli, yeah i already started implementing it,

What ive done now are, since my existing asterisk/vicidial & apace/mysql server is completely running and operational

-I added another server- i have setup it on the same process a i created my 1st asterisk/vicidial server:

-astguiclient.conf--> are pointed to Database server
- i created an iax.conf as ststed by load balance docs- now bot server have detected itself. and also i modified also the extensions.conf and added neceesary things form loadbalance docs, also ive setup the voip trunk on Sip. Its crontab also

-Im on a halt in this part what should be my next steps to used and test and configure..this where i stop

by **okli** » Fri Oct 23, 2009 2:05 pm

Quick search with terms "load balance" and the radio buton moved to "Search all terms":

http://www.vicidial.org/VICIDIALforum/v ... ad+balance
http://www.vicidial.org/VICIDIALforum/v ... ad+balance
http://www.vicidial.org/VICIDIALforum/v ... ad+balance

Hint- add the new server, conferences and phones via the admin page.

by **ruben23** » Fri Oct 23, 2009 3:28 pm

okli thanks for the 2nd time youve done this helping,

-just an add up, i already added 2 server now on the admin, how do i set the VICIDIAL AD exten, should they be the same or should i used the two extensions specified on the Loadbalance docs.
8367 or 8368 ( should be different for both server..?)

- the instruction on loadblance docs is to used on the VDAD exten on the campaigns only not specified on the server.

by **ruben23** » Fri Oct 23, 2009 4:10 pm

Now when i enable the load balance on both servers the Asterisk 2 (192.168.2.7) whihc i newly added begun to create log calls on the CLI, but it saying UNavalible channel:

Oct 23 13:47:12] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Oct 23 13:47:12] == Parsing '/etc/asterisk/manager.conf': [Oct 23 13:47:12] Found
[Oct 23 13:47:12] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 23 13:47:12] -- Executing [917184717890@default:1] AGI("Local/917184717890@default-6795,2", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 23 13:47:12] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 23 13:47:12] -- Executing [917184717890@default:2] Dial("Local/917184717890@default-6795,2", "SIP/APN1/17184717890||To") in new stack
[Oct 23 13:47:12] WARNING[12985]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Oct 23 13:47:12] == Everyone is busy/congested at this time (1:0/0/1)
[Oct 23 13:47:12] -- Executing [917184717890@default:3] Hangup("Local/917184717890@default-6795,2", "") in new stack
[Oct 23 13:47:12] == Spawn extension (default, 917184717890, 3) exited non-zero on 'Local/917184717890@default-6795,2'
[Oct 23 13:47:12] -- Executing [h@default:1] DeadAGI("Local/917184717890@default-6795,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
[Oct 23 13:47:12] == Parsing '/etc/asterisk/manager.conf': [Oct 23 13:47:12] Found
[Oct 23 13:47:12] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 23 13:47:12] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Oct 23 13:47:12] -- Executing [917184713972@default:1] AGI("Local/917184713972@default-183d,2", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 23 13:47:12] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 23 13:47:12] -- Executing [917184713972@default:2] Dial("Local/917184713972@default-183d,2", "SIP/APN1/17184713972||To") in new stack
[Oct 23 13:47:12] WARNING[12987]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Oct 23 13:47:12] == Everyone is busy/congested at this time (1:0/0/1)
[Oct 23 13:47:12] -- Executing [917184713972@default:3] Hangup("Local/917184713972@default-183d,2", "") in new stack
[Oct 23 13:47:12] == Spawn extension (default, 917184713972, 3) exited non-zero on 'Local/917184713972@default-183d,2'
[Oct 23 13:47:12] -- Executing [h@default:1] DeadAGI("Local/917184713972@default-183d,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
[Oct 23 13:47:12] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Oct 23 13:47:12] == Parsing '/etc/asterisk/manager.conf': [Oct 23 13:47:12] Found
[Oct 23 13:47:12] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 23 13:47:12] -- Executing [913479264740@default:1] AGI("Local/913479264740@default-73e9,2", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 23 13:47:12] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 23 13:47:12] -- Executing [913479264740@default:2] Dial("Local/913479264740@default-73e9,2", "SIP/APN1/13479264740||To") in new stack
[Oct 23 13:47:12] WARNING[12991]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Oct 23 13:47:12] == Everyone is busy/congested at this time (1:0/0/1)
[Oct 23 13:47:12] -- Executing [913479264740@default:3] Hangup("Local/913479264740@default-73e9,2", "") in new stack
[Oct 23 13:47:12] == Spawn extension (default, 913479264740, 3) exited non-zero on 'Local/913479264740@default-73e9,2'
[Oct 23 13:47:12] -- Executing [h@default:1] DeadAGI("Local/913479264740@default-73e9,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
[Oct 23 13:47:12] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Oct 23 13:47:15] == Manager 'sendcron' logged off from 127.0.0.1

question, since my asterisk1(192.168.2.5) have already exixting conference and Phone on my admin page,

How do add up conference and phone for my second Asterisk2(192.168.2.7)

should i copy the same conference number and phone ID on my asterisk1 adn just put the IP routed to my asterisk2(192.168.2.7).

Im on the admin configuration only now.

by **okli** » Fri Oct 23, 2009 5:27 pm

should i copy the same conference number and phone ID on my asterisk1 adn just put the IP routed to my asterisk2(192.168.2.7)

Yep, you basically replicate the data- phones, conferences but as IP you use the IP of the new server so Vicidial knows where to find that phone/conference.

by **mflorell** » Fri Oct 23, 2009 6:36 pm

With more than one Asterisk/ViciDial-Agent server you might also want to consider using Phone Aliases to allow for agent phone login load balancing.

by **ruben23** » Fri Oct 23, 2009 6:50 pm

mflorell phone alias means having single alias for the phone extension on the asterisk/vici1 and asterisk/vici2..?

when i add phones between the two server should i add phone exten cc100 on server 1( which is default) and c100 also for Server2.?

by **ruben23** » Fri Oct 23, 2009 6:57 pm

and also when i add up the conference on the vicidial admin, there is a section Show Conference and Vicidial Conference.

now i added until 8600049 with the new server, but when i add 8600051 going up, is it ok that it added on the show conference not on the vicidial conference as okli showed in his screenshot, does it matters...? between show conference & Vicidial Conference

:?:

by **mflorell** » Fri Oct 23, 2009 10:06 pm

You can have the same extensions and conferences on different servers, so cc100 on server 1 and cc100 on server 2 is just fine.

by **ruben23** » Fri Oct 23, 2009 10:14 pm

mflorell im getting this on my server2:

Oct 23 20:11:35] -- Added extension '_84244XXX' priority 1 to default
[Oct 23 20:11:35] WARNING[7907]: pbx.c:4702 add_pri: Unable to register extension '_8600XXX*.', priority 1 in 'default', already in use
[Oct 23 20:11:35] WARNING[7907]: pbx.c:4702 add_pri: Unable to register extension '_78600XXX*.', priority 1 in 'default', already in use

My server 1 ( 192.168.2.5), extensions.conf

TRUNKIAX2=IAX2/ASTtest1:test@192.168.2.7

[default]
include => vicidial-auto

; [Start] Load Balancing Wildcard Entries
exten => _192*168*002*005*.,1,Goto(default,${EXTEN:16},1)
exten => _192*168*002*007*.,1,Dial(${TRUNKIAX2}/${EXTEN:16},55,o)

exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)

My server2 (192.168.2.7), extensionsw.conf
TRUNKIAX1=IAX2/ASTtest2:test@192.168.2.5

; [Start] Load Balancing Wildcard Entries
exten => _192*168*002*005*.,1,Dial(${TRUNKIAX1}/${EXTEN:16},55,o)
exten => _192*168*002*007*.,1,Goto(default,${EXTEN:16},1)

exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)

; [End ]Load Balancing Wildcard Entries

what could be the problem.

by **okli** » Sat Oct 24, 2009 2:01 am

You have these lines with extensions _8600XXX*. and _78600XXX*. already present in your dialplan. Look for duplicated lines in extensions.conf.

Come on, you should have figured that out

The message from asterisk is informative enough:

Bolded part is the hint:

[Oct 23 20:11:35] WARNING[7907]: pbx.c:4702 add_pri: Unable to register extension '_8600XXX*.', priority 1 in 'default', already in use

by **ruben23** » Sat Oct 24, 2009 8:07 am

okli sorry about that, guess i dont need to input that details anymore..

How about the option of adding phone extensions, tried adding one just by changing the Asterisk server IP which is my server2. but it wont appear on the phone list after i added it up, tried couple of times still no phone extensions excess...

by **jojo1031** » Sat Oct 24, 2009 9:56 am

Guys,

Is it really necessary to create multiple conferences and Vicidial conferences (8600001 and 8600051 pointing to xxx.xxx.xxx.1 and 8600001 and 8600051 pointing to xxx.xxx.xxx.2 given you have 2 asterisk/vicidial servers) as what okli did based of the screenshot that he made?

Thanks,

Jojo

by **Op3r** » Sat Oct 24, 2009 11:51 am

Jojo,

Yes.

@ruben

You need to create a server entry on the vicidial web admin then create your phones, your conferences and vicidial conferences.

If you know about mysql, edit the first_server_install.sql to fit your needs.

by **ruben23** » Sun Oct 25, 2009 5:56 am

Op3r.. ive tried creating the phone exten of the new server, problem is it wont appear on the phone list on my admin page, tried several times same thing,
what might be wrong...

by **ruben23** » Mon Oct 26, 2009 12:07 pm

hi i was able to add phone extensions on the new server, in my load balance Asterisk/vicidial server. i just change the login of the new extension.

now when i tried setting the phone extension and let the agent login:

Aasterisk/vicidial:

Server1 ( 192.168.2.5)
Server2 ( 192.168.2.7)

agents registered to server2 can able to dial same as agent on server1,

problem is my server2 is getting this problem on the asterisk CLI;

[Oct 26 10:04:51] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 10:04:51] -- Executing [917184717090@default:1] AGI("Local/917184717090@default-6936,2", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 26 10:04:51] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 26 10:04:51] -- Executing [917184717090@default:2] Dial("Local/917184717090@default-6936,2", "SIP/APN1/17184717090||To") in new stack
[Oct 26 10:04:51] WARNING[23182]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Oct 26 10:04:51] == Everyone is busy/congested at this time (1:0/0/1)
[Oct 26 10:04:51] -- Executing [917184717090@default:3] Hangup("Local/917184717090@default-6936,2", "") in new stack
[Oct 26 10:04:51] == Spawn extension (default, 917184717090, 3) exited non-zero on 'Local/917184717090@default-6936,2'
[Oct 26 10:04:51] -- Executing [h@default:1] DeadAGI("Local/917184717090@default-6936,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
[Oct 26 10:04:51] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Oct 26 10:04:51] == Parsing '/etc/asterisk/manager.conf': [Oct 26 10:04:51] Found
[Oct 26 10:04:51] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 10:04:51] -- Executing [917184712576@default:1] AGI("Local/917184712576@default-1681,2", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 26 10:04:51] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 26 10:04:51] -- Executing [917184712576@default:2] Dial("Local/917184712576@default-1681,2", "SIP/APN1/17184712576||To") in new stack
[Oct 26 10:04:51] WARNING[23184]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Oct 26 10:04:51] == Everyone is busy/congested at this time (1:0/0/1)
[Oct 26 10:04:51] -- Executing [917184712576@default:3] Hangup("Local/917184712576@default-1681,2", "") in new stack
[Oct 26 10:04:51] == Spawn extension (default, 917184712576, 3) exited non-zero on 'Local/917184712576@default-1681,2'
[Oct 26 10:04:51] -- Executing [h@default:1] DeadAGI("Local/917184712576@default-1681,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
[Oct 26 10:04:51] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Oct 26 10:04:55] == Manager 'sendcron' logged off from 127.0.0.1

Name/username Host Dyn Nat ACL Port Status
APN2 208.74.74.252 N 5060 UNREACHABLE
APN1 208.74.75.250 N 5060 UNREACHABLE
rapidvox 64.21.13.41 N 5060 UNREACHABLE
cc126/cc126 192.168.2.23 D N 4170 OK (101 ms)
cc125/cc125 192.168.2.22 D N 51416 OK (103 ms)
cc116/cc116 (Unspecified) D N 0 UNKNOWN

- it seemed my server was not able to reach the VOIP telco which also i used on the Server1(working). Is it possible to used same VOIP telco on both load balance Asterisk/vicidial server..?

how do i resolve this problem..any suggestion.

by **okli** » Mon Oct 26, 2009 12:57 pm

Look at this thread
http://www.vicidial.org/VICIDIALforum/v ... highlight=

by **jojo1031** » Mon Oct 26, 2009 1:04 pm

both servers should have registration and dial-out cability to your provider.

Hope that helps

by **ruben23** » Mon Oct 26, 2009 1:11 pm

yeah the public ip im using have is laready registered, its just my second balance asterisk/vicidial server is just using single ethernet--routed to my gateway/router server:

What in my mind is maybe i can used Alias on my existing ethernet card and used two IP on it:

eth0= xxx.xxx.xxx.xx. public ip
eth0:0 = 192.168.2.7 local ip

same physical single Ethernet card.. Is this possible Mr. Okli....Thanks

by **ruben23** » Mon Oct 26, 2009 2:06 pm

hi i made it with Alias IP, now my asterisk/vicidial (server2) are having this output on the CLI is this doing load balance..?

[Oct 26 12:01:57] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 12:01:57] -- Executing [917184742340@default:1] AGI("Local/917184742340@default-58d0,2", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 26 12:01:57] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 26 12:01:57] -- Executing [917184742340@default:2] Dial("Local/917184742340@default-58d0,2", "SIP/APN1/17184742340||To") in new stack
[Oct 26 12:01:57] -- Called APN1/17184742340
[Oct 26 12:01:58] == Parsing '/etc/asterisk/manager.conf': [Oct 26 12:01:58] Found
[Oct 26 12:01:58] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 12:01:58] -- Executing [917186344161@default:1] AGI("Local/917186344161@default-d214,2", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 26 12:01:58] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 26 12:01:58] -- Executing [917186344161@default:2] Dial("Local/917186344161@default-d214,2", "SIP/APN1/17186344161||To") in new stack
[Oct 26 12:01:58] -- Called APN1/17186344161
[Oct 26 12:01:58] == Parsing '/etc/asterisk/manager.conf': [Oct 26 12:01:58] Found
[Oct 26 12:01:58] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 12:01:58] -- Hungup 'IAX2/ASTtest1-3752'
[Oct 26 12:01:58] == Spawn extension (default, 192*168*002*005*8600061, 1) exited non-zero on 'SIP/APN1-082865f0'
[Oct 26 12:01:58] -- Executing [h@default:1] DeadAGI("SIP/APN1-082865f0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----56-----56") in new stack
[Oct 26 12:01:58] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -56-----56 completed, returning 0
[Oct 26 12:01:59] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 26 12:02:00] -- SIP/APN1-082a75b8 is ringing
[Oct 26 12:02:00] -- SIP/APN1-081ffb40 answered Local/917186342920@default-2bff,2
[Oct 26 12:02:00] > Channel Local/917186342920@default-2bff,1 was answered.
[Oct 26 12:02:00] -- Executing [8368@default:1] Playback("Local/917186342920@default-2bff,1", "sip-silence") in new stack
[Oct 26 12:02:00] -- <Local/917186342920@default-2bff,1> Playing 'sip-silence' (language 'en')
[Oct 26 12:02:00] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 26 12:02:00] == Spawn extension (default, 917186342920, 2) exited non-zero on 'Local/917186342920@default-2bff,2'
[Oct 26 12:02:00] -- Executing [h@default:1] DeadAGI("Local/917186342920@default-2bff,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----20-----0") in new stack
[Oct 26 12:02:00] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --20-----0 completed, returning 0
[Oct 26 12:02:00] -- Executing [8368@default:2] AGI("SIP/APN1-081ffb40", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 26 12:02:00] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 26 12:02:00] -- Executing [8368@default:3] AGI("SIP/APN1-081ffb40", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Oct 26 12:02:00] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Oct 26 12:02:00] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Oct 26 12:02:00] -- Executing [192*168*002*005*8600054@default:1] Dial("SIP/APN1-081ffb40", "IAX2/ASTtest2:test@192.168.2.5/8600054|55|o") in new stack
[Oct 26 12:02:00] -- Called ASTtest2:test@192.168.2.5/8600054
[Oct 26 12:02:00] -- Call accepted by 192.168.2.5 (format ulaw)
[Oct 26 12:02:00] -- Format for call is ulaw
[Oct 26 12:02:00] -- IAX2/ASTtest1-3628 answered SIP/APN1-081ffb40

Now the issue i have is my phone extensions on my Server2(192.168.2.7) can login with its details but not able to get a conference( no ringback on login" your the only person in this conference)--> ive laready added the conference of my server2 including vicidial 860001 to 86000 to 860299. what i haved missed.

by **ruben23** » Tue Oct 27, 2009 11:52 am

hi do i have to add datas and tables of my second asterisk/vicidial on existing databased servver, so conference would work..? and copies other config for my asterisk/vicidial servers..?

any idea..?

by **ruben23** » Tue Oct 27, 2009 12:22 pm

Do i have to re run this on mysql server for my second server Asterisk/vicidial....?

> use asterisk;
> \. /usr/src/astguiclient/trunk/extras/MySQL_AST_CREATE_tables.sql
> \. /usr/src/astguiclient/trunk/extras/first_server_install.sql
> \. /usr/src/astguiclient/trunk/extras/sip-iax_phones.sql

any suggestion..

by **mflorell** » Tue Oct 27, 2009 3:42 pm

Have you looked at the second_server_install.sql file?

by **ruben23** » Tue Oct 27, 2009 3:47 pm

yes i looked it up should i edit it and change the IP of its existing, actually on my second server i never done adding any entries on my databased, what just i do is manually added the phone extensions and conference of the second server:

Problem it it seems not working coz agent cant login if they used phone extensions using second server. help mflorell---what should i do.. really would like to finished this up--its just im missing something..

by **mflorell** » Tue Oct 27, 2009 3:53 pm

What is the Asterisk CLI output on the second server?

by **ruben23** » Tue Oct 27, 2009 4:00 pm

This is the output when an agent tries to login using the phone extension of the second server.. at present the server 2 is actually load balanceing i think but only to aget who are log to the 1st server, cannot do with second server.

[Oct 26 17:14:54] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI---- -NODEBUG-----16--------------- completed, returning 0
[Oct 26 17:14:54] == Parsing '/etc/asterisk/manager.conf': [Oct 26 17:14:54] Found
[Oct 26 17:14:54] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 26 17:14:54] -- Executing [55558600055@default:1] MeetMeAdmin("Local/555586 00055@default-9ebb,2", "8600055|K") in new stack
[Oct 26 17:14:54] WARNING[21959]: app_meetme.c:2821 admin_exec: Conference number '8 600055' not found!
[Oct 26 17:14:54] -- Executing [55558600055@default:2] Hangup("Local/55558600055 @default-9ebb,2", "") in new stack
[Oct 26 17:14:54] == Spawn extension (default, 55558600055, 2) exited non-zero on 'Local/55558600055@default-9ebb,2'
[Oct 26 17:14:54] -- Executing [h@default:1] DeadAGI("Local/55558600055@default- 9ebb,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16----------- ----") in new stack
[Oct 26 17:14:54] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI---- -NODEBUG-----16--------------- completed, returning 0

what else you need just say it...

by **mflorell** » Tue Oct 27, 2009 8:44 pm

strange, looks like you have a space in there, is that really there is is the Forum software messing it up?

by **ruben23** » Wed Oct 28, 2009 10:30 am

sorry about that, thats my fault...actualy no space in between., so what you think could be the problem...?

by **okli** » Wed Oct 28, 2009 2:30 pm

Do you have those conferences declared in extensions.conf at server 2?

by **ruben23** » Wed Oct 28, 2009 2:37 pm

okli yes i have but what i do is i add up it manually to my vicidial admin, on conference and vicidial conference..same value of conference except its server IP i change to my server2 asterisk/vicidial..

Would that be automatically added to the mysql server..? or do i have to perform this..?

> use asterisk;
> \. /usr/src/astguiclient/trunk/extras/MySQL_AST_CREATE_tables.sql
> \. /usr/src/astguiclient/trunk/extras/first_server_install.sql
> \. /usr/src/astguiclient/trunk/extras/sip-iax_phones.sql

by **okli** » Wed Oct 28, 2009 2:52 pm

what i do is i add up it manually to my vicidial admin, on conference and vicidial conference..same value of conference except its server IP i change to my server2 asterisk/vicidial..

Did that fix the problem?
BTW- weren't we at that point like 10 posts above, on the screenshots? Did you forget to do it or there was something else?

Would that be automatically added to the mysql server..? or do i have to perform this..?

> use asterisk;
> \. /usr/src/astguiclient/trunk/extras/MySQL_AST_CREATE_tables.sql
> \. /usr/src/astguiclient/trunk/extras/first_server_install.sql
> \. /usr/src/astguiclient/trunk/extras/sip-iax_phones.sql

These files would be used if you wanted to automate adding entries for the second server, editing the files beforehand and changing the IPs with the new one.

As for the first file- it supposedly creates the needed tables in the database. You have them created when the first server was setup and for the second and so on you just populate them with new values.
So the answer is- you do NOT use the first file at all.
And no- you do not use the other files if you already manually filled the information in them via the admin interface.

by **ruben23** » Wed Oct 28, 2009 3:10 pm

okli yes i did it manually as i see it on you shared screenshots,

for my conference i added

8600001 192.168.2.7 MODIFY
8600001 192.168.2.5 MODIFY
8600002 192.168.2.5 MODIFY
8600002 192.168.2.7 MODIFY
8600003 192.168.2.5 MODIFY
8600003 192.168.2.7 MODIFY
8600004 192.168.2.7 MODIFY
8600004 192.168.2.5 MODIFY
8600005 192.168.2.7 MODIFY
8600005 192.168.2.5 MODIFY
8600006 192.168.2.7 MODIFY
8600006 192.168.2.5 MODIFY
8600007 192.168.2.5 MODIFY
8600008 192.168.2.7 MODIFY
8600008 192.168.2.5 MODIFY
8600009 192.168.2.5 MODIFY
8600009 192.168.2.7 MODIFY
8600010 192.168.2.5 MODIFY
8600010 192.168.2.7 MODIFY
8600011 192.168.2.5 MODIFY
8600011 192.168.2.7 MODIFY
8600012 192.168.2.7 MODIFY
8600012 192.168.2.5 MODIFY
8600013 192.168.2.7 MODIFY
8600013 192.168.2.5 MODIFY
8600014 192.168.2.5 MODIFY
8600014 192.168.2.7 MODIFY
8600015 192.168.2.7 MODIFY
8600015 192.168.2.5 MODIFY
8600016 192.168.2.5 MODIFY
8600016 192.168.2.7 MODIFY
8600017 192.168.2.5 MODIFY
8600017 192.168.2.7 MODIFY
8600018 192.168.2.7 MODIFY
8600018 192.168.2.5 MODIFY
8600019 192.168.2.7 MODIFY
8600019 192.168.2.5 MODIFY
8600020 192.168.2.5 MODIFY
8600020 192.168.2.7 MODIFY
8600021 192.168.2.7 MODIFY
8600021 192.168.2.5 MODIFY
8600022 192.168.2.5 MODIFY
8600022 192.168.2.7 MODIFY
8600023 192.168.2.5 MODIFY
8600023 192.168.2.7 MODIFY
8600024 192.168.2.5 MODIFY
8600024 192.168.2.7 MODIFY
8600025 192.168.2.7 MODIFY
8600025 192.168.2.5 MODIFY
8600026 192.168.2.5 MODIFY
8600026 192.168.2.7 MODIFY
8600027 192.168.2.7 MODIFY
8600027 192.168.2.5 MODIFY
8600028 192.168.2.7 MODIFY
8600028 192.168.2.5 MODIFY
8600029 192.168.2.5 MODIFY
8600029 192.168.2.7 MODIFY
8600030 192.168.2.7 MODIFY
8600030 192.168.2.5 MODIFY
8600031 192.168.2.7 MODIFY
8600031 192.168.2.5 MODIFY
8600032 192.168.2.7 MODIFY
8600032 192.168.2.5 MODIFY
8600033 192.168.2.7 MODIFY
8600033 192.168.2.5 MODIFY
8600034 192.168.2.7 MODIFY
8600034 192.168.2.5 MODIFY
8600035 192.168.2.7 MODIFY
8600035 192.168.2.5 MODIFY
8600036 192.168.2.5 MODIFY
8600036 192.168.2.7 MODIFY
8600037 192.168.2.7 MODIFY
8600037 192.168.2.5 MODIFY
8600038 192.168.2.5 MODIFY
8600038 192.168.2.7 MODIFY
8600039 192.168.2.7 MODIFY
8600039 192.168.2.5 MODIFY
8600040 192.168.2.5 MODIFY
8600040 192.168.2.7 MODIFY
8600041 192.168.2.7 MODIFY

for the vicidial conference same thing:

8600051 192.168.2.7 MODIFY
8600051 192.168.2.5 SIP/cc126 MODIFY
8600052 192.168.2.7 MODIFY
8600052 192.168.2.5 SIP/cc115 MODIFY
8600053 192.168.2.7 MODIFY
8600053 192.168.2.5 SIP/cc116 MODIFY
8600054 192.168.2.7 MODIFY
8600054 192.168.2.5 SIP/cc113 MODIFY
8600055 192.168.2.7 MODIFY
8600055 192.168.2.5 SIP/cc124 MODIFY
8600056 192.168.2.7 MODIFY
8600056 192.168.2.5 SIP/cc125 MODIFY
8600057 192.168.2.7 MODIFY
8600057 192.168.2.5 SIP/cc120 MODIFY
8600058 192.168.2.7 MODIFY
8600058 192.168.2.5 SIP/cc127 MODIFY
8600059 192.168.2.7 MODIFY
8600059 192.168.2.5 SIP/cc119 MODIFY
8600060 192.168.2.7 MODIFY
8600060 192.168.2.5 SIP/cc114 MODIFY
8600061 192.168.2.7 MODIFY
8600061 192.168.2.5 SIP/cc118 MODIFY
8600062 192.168.2.5 SIP/cc109 MODIFY
8600062 192.168.2.7 MODIFY
8600063 192.168.2.7 MODIFY
8600063 192.168.2.5 SIP/cc117 MODIFY
8600064 192.168.2.7 MODIFY
8600064 192.168.2.5 SIP/cc110 MODIFY
8600065 192.168.2.7 MODIFY
8600065 192.168.2.5 SIP/cc108 MODIFY
8600066 192.168.2.5 SIP/cc112 MODIFY
8600066 192.168.2.7 MODIFY
8600067 192.168.2.7 MODIFY
8600067 192.168.2.5 MODIFY
8600068 192.168.2.7 MODIFY
8600068 192.168.2.5 MODIFY
8600069 192.168.2.5 MODIFY
8600069 192.168.2.7 MODIFY
8600070 192.168.2.5 MODIFY
8600070 192.168.2.7 MODIFY
8600071 192.168.2.5 MODIFY
8600071 192.168.2.7 MODIFY
8600072 192.168.2.5 MODIFY
8600072 192.168.2.7 MODIFY
8600073 192.168.2.7 MODIFY
8600073 192.168.2.5 MODIFY
8600074 192.168.2.5 MODIFY
8600074 192.168.2.7 MODIFY
8600075 192.168.2.7 MODIFY
8600075 192.168.2.5 MODIFY
8600076 192.168.2.7 MODIFY
8600076 192.168.2.5 MODIFY
8600077 192.168.2.7 MODIFY
8600077 192.168.2.5 MODIFY
8600078 192.168.2.7 MODIFY
8600078 192.168.2.5 MODIFY
8600079 192.168.2.7 MODIFY
8600079 192.168.2.5 MODIFY
8600080 192.168.2.5 MODIFY
8600080 192.168.2.7 MODIFY
8600081 192.168.2.7 MODIFY
8600081 192.168.2.5 MODIFY
8600082 192.168.2.5 MODIFY
8600082 192.168.2.7 MODIFY
8600083 192.168.2.7 MODIFY
8600083 192.168.2.5 MODIFY
8600084 192.168.2.5 MODIFY
8600084 192.168.2.7 MODIFY
8600085 192.168.2.5 MODIFY
8600085 192.168.2.7 MODIFY
8600086 192.168.2.7 MODIFY
8600086 192.168.2.5 MODIFY
8600087 192.168.2.7 MODIFY
8600087 192.168.2.5 MODIFY
8600088 192.168.2.7 MODIFY
8600088 192.168.2.5 MODIFY
8600089 192.168.2.7 MODIFY
8600089 192.168.2.5 MODIFY
8600090 192.168.2.7 MODIFY
8600090 192.168.2.5 MODIFY
8600091 192.168.2.7 MODIFY
8600091 192.168.2.5 MODIFY
8600092 192.168.2.5 MODIFY

with the phone extensions for both servers also:

cc100 SIP 192.168.2.5 100 100 ACTIVE station 100 0 0 MODIFY | STATS
cc100 SIP 192.168.2.7 100 100 ACTIVE Station100 0 0 MODIFY | STATS
cc101 SIP 192.168.2.5 101 101 ACTIVE station 101 0 0 MODIFY | STATS
cc101 SIP 192.168.2.7 101 101 ACTIVE Station101 0 0 MODIFY | STATS
cc102 SIP 192.168.2.5 102 102 ACTIVE station 102 0 0 MODIFY | STATS
cc102 SIP 192.168.2.7 102 102 ACTIVE Station102 0 0 MODIFY | STATS
cc103 SIP 192.168.2.5 103 103 ACTIVE station 103 0 0 MODIFY | STATS
cc103 SIP 192.168.2.7 103 103 ACTIVE Station103 0 0 MODIFY | STATS
cc104 SIP 192.168.2.5 104 104 ACTIVE station 104 0 0 MODIFY | STATS
cc104 SIP 192.168.2.7 104 104 ACTIVE Station104 0 0 MODIFY | STATS
cc105 SIP 192.168.2.5 105 105 ACTIVE station 105 0 0 MODIFY | STATS
cc105 SIP 192.168.2.7 105 105 ACTIVE Station105 0 0 MODIFY | STATS
cc106 SIP 192.168.2.5 106 106 ACTIVE station 106 0 0 MODIFY | STATS
cc106 SIP 192.168.2.7 106 106 ACTIVE Station106 0 0 MODIFY | STATS
cc107 SIP 192.168.2.5 107 107 ACTIVE station 107 0 0 MODIFY | STATS
cc107 SIP 192.168.2.7 107 107 ACTIVE Station107 0 0 MODIFY | STATS
cc108 SIP 192.168.2.5 108 108 ACTIVE station 108 0 0 MODIFY | STATS
cc108 SIP 192.168.2.7 108 108 ACTIVE Station108 0 0 MODIFY | STATS
cc109 SIP 192.168.2.5 109 109 ACTIVE station 109 0 0 MODIFY | STATS
cc109 SIP 192.168.2.7 109 109 ACTIVE Station109 0 0 MODIFY | STATS
cc110 SIP 192.168.2.5 110 110 ACTIVE station 110 0 0 MODIFY | STATS
cc110 SIP 192.168.2.7 110 110 ACTIVE Station110 0 0 MODIFY | STATS
cc111 SIP 192.168.2.5 111 111 ACTIVE station 111 0 0 MODIFY | STATS
cc111 SIP 192.168.2.7 111 111 ACTIVE Station111 0 0 MODIFY | STATS
cc112 SIP 192.168.2.5 112 112 ACTIVE station 112 0 0 MODIFY | STATS
cc112 SIP 192.168.2.7 112 112 ACTIVE Station112 0 0 MODIFY | STATS
cc113 SIP 192.168.2.5 113 113 ACTIVE station 113 0 0 MODIFY | STATS
cc113 SIP 192.168.2.7 113 113 ACTIVE Station113 0 0 MODIFY | STATS
cc114 SIP 192.168.2.5 114 114 ACTIVE station 114 0 0 MODIFY | STATS
cc114 SIP 192.168.2.7 114 114 ACTIVE Station114 0 0 MODIFY | STATS
cc115 SIP 192.168.2.5 115 115 ACTIVE station 115 0 0 MODIFY | STATS
cc115 SIP 192.168.2.7 115 115 ACTIVE Station115 0 0 MODIFY | STATS
cc116 SIP 192.168.2.5 116 116 ACTIVE station 116 0 0 MODIFY | STATS
cc116 SIP 192.168.2.7 116 116 ACTIVE Station116 0 0 MODIFY | STATS
cc117 SIP 192.168.2.5 117 117 ACTIVE station 117 0 0 MODIFY | STATS
cc118 SIP 192.168.2.5 118 118 ACTIVE station 118 0 0 MODIFY | STATS
cc119 SIP 192.168.2.5 119 119 ACTIVE station 119 0 0 MODIFY | STATS
cc120 SIP 192.168.2.5 120 120 ACTIVE station 120 0 0 MODIFY | STATS
cc121 SIP 192.168.2.5 121 121 ACTIVE station 121 5 0 MODIFY | STATS
cc122 SIP 192.168.2.5 122 122 ACTIVE station 122 0 0 MODIFY | STATS
cc123 SIP 192.168.2.5 123 123 ACTIVE station 123 0 0 MODIFY | STATS
cc124 SIP 192.168.2.5 124 124 ACTIVE station 124 0 0 MODIFY | STATS
cc125 SIP 192.168.2.5 125 125 ACTIVE station 125 0 0 MODIFY | STATS
cc125 SIP 192.168.2.7 125 125 ACTIVE Station125 0 0 MODIFY | STATS
cc126 SIP 192.168.2.5 126 126 ACTIVE station 126 0 0 MODIFY | STATS
cc126 SIP 192.168.2.7 126 126 ACTIVE Station126 0 0 MODIFY | STATS
cc127 SIP 192.168.2.5 127 127 ACTIVE station 127 0 0 MODIFY | STATS
cc128 SIP 192.168.2.5 128 128 ACTIVE station 128 0 0 MODIFY | STATS
cc128 SIP 192.168.2.7 128 128 ACTIVE Station128 0 0 MODIFY | STATS
cc129 SIP 192.168.2.5 129 129 ACTIVE station 129 0 0 MODIFY | STATS
cc130 SIP 192.168.2.5 130 130 ACTIVE station 130 0 0 MODIFY | STATS

thats all i have and problem si when my agents log using the first server they are ok and there calls are load balance between the 2 servers /asterik/vicidial, now when they login through server2 ( 192.168.2.7) that errors come up on the CLI, no ring back on sofphones..

by **okli** » Wed Oct 28, 2009 4:08 pm

Just in case- adding those conferences via the admin page does NOT amend your extensions.conf, you need to look at it separately. Ensure everything in it at server 2 is correct, especially the conferences sections.

by **ruben23** » Wed Oct 28, 2009 4:16 pm

okli..what part of the extensions.conf the vicidial conference falls, can you give shots or just a part of its extensions to guide, thanks in advance.

by **okli** » Wed Oct 28, 2009 4:45 pm

Om a second thought- check meetme.conf as well.

by **ruben23** » Wed Oct 28, 2009 5:41 pm

hi okli, meetme.conf are all intact for both server, and alo i have check phpmyadmin both conference for both servers are intact also,

on the extensions.conf all setting are all default and the same, im looking to problem with adding the phone extensions...do i need to change setting for my second server with its details, or should it be the same setting for both servers,

like:

Park Exten: 8301
Conf Exten: 8302
VICIDIAL Park Exten: 8301
VICIDIAL Park File:
Monitor Prefix: 8612
Recording Exten:
VMailMain Exten

just a suggestion..

by **ruben23** » Wed Oct 28, 2009 5:47 pm

also for my load balance configuration i have this:

Server1 ( 192.168.2.5 )

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=SIP/APN1 ; Trunk interface
TRUNKX=SIP/APN1 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com ; IAX trunk interface
SIPtrunk=SIP/1234:PASSWORD@sip.provider.net ; SIP trunk
TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569 ; used for blind monitoring
TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569 ; used for testing
TRUNKIAX2=IAX2/ASTtest1:test@192.168.2.7

; [Start] Load Balancing Wildcard Entries
exten => _192*168*002*005*.,1,Goto(default,${EXTEN:16},1)
exten => _192*168*002*007*.,1,Dial(${TRUNKIAX2}/${EXTEN:16},55,o)

exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)

; [End ]Load Balancing Wildcard Entries

And for my server2 (192.168.2.7)

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=SIP/APN1 ; Trunk interface
TRUNKX=SIP/APN1 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com ; IAX trunk interface
SIPtrunk=SIP/1234:PASSWORD@sip.provider.net ; SIP trunk
TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569 ; used for blind monitoring
TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569 ; used for testing
TRUNKIAX1=IAX2/ASTtest2:test@192.168.2.5

[default]
include => vicidial-auto

; [Start] Load Balancing Wildcard Entries
exten => _192*168*002*005*.,1,Dial(${TRUNKIAX1}/${EXTEN:16},55,o)
exten => _192*168*002*007*.,1,Goto(default,${EXTEN:16},1)

; [End ]Load Balancing Wildcard Entries

--------> i did not add the this to server2, coz it would make a warning to my CLI warning:

exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)

----->Oct 23 20:11:35] -- Added extension '_84244XXX' priority 1 to default
[Oct 23 20:11:35] WARNING[7907]: pbx.c:4702 add_pri: Unable to register extension '_8600XXX*.', priority 1 in 'default', already in use
[Oct 23 20:11:35] WARNING[7907]: pbx.c:4702 add_pri: Unable to register extension '_78600XXX*.', priority 1 in 'default', already in use

thats my latest config:

by **okli** » Wed Oct 28, 2009 6:38 pm

Please remove any sensitive information from extensions.conf, put it in a zip archive and upload it to a free file hosting site, such as www.datafilehost.com or www.mediafire.com . Provide download link.

There must be something yo are overlooking.

vicidial.org

LOAD BALANCE query.....

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