Callcenter module outgoing calls not working

Joined
Apr 17, 2011
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I also managed to get it work from a different SIP trunk.

I still cannot get it to work through my trunk going through AVAYA :(

I understand the problem was with my trunk but I can call exactly the same numbers manually from the same trunk using Avaya which the CC module is not able to do!!

What extra settings does call center module need to make it work through Avaya??
 
Joined
Feb 28, 2008
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There is no extra configuration...
Callcenter module, doesn't know if the target trunk is an avaya, a sip provider, or any other thing, it just dials through the trunk...
Paste the CLI output when the outbound campaign tries to dial out through yout avaya trunk.
 
Joined
Apr 17, 2011
Messages
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Hey,

I get continuous hold music till I log out, Below is my CLI:-

==================================================================================
== Agent '100' logged in (format ulaw/ulaw)
> doing dnsmgr_lookup for 'sip.pennytel.com'
> ast_get_srv: SRV lookup for '_sip._udp.sip.pennytel.com' mapped to host sip6.pennytel.com, port 5060
> doing dnsmgr_lookup for 'sip.pennytel.com'
> ast_get_srv: SRV lookup for '_sip._udp.sip.pennytel.com' mapped to host sip6.pennytel.com, port 5060
> doing dnsmgr_lookup for 'sip.pennytel.com'
> ast_get_srv: SRV lookup for '_sip._udp.sip.pennytel.com' mapped to host sip6.pennytel.com, port 5060
-- Stopped music on hold on SIP/6397-00000010
== Agent '100' logged out
== Spawn extension (from-internal, *8888100, 3) exited non-zero on 'SIP/6397-00000010'
-- Executing [h@from-internal:1] Macro("SIP/6397-00000010", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/6397-00000010", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/6397-00000010", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/6397-00000010", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/6397-00000010", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/6397-00000010", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/6397-00000010", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/6397-00000010' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/6397-00000010'
> doing dnsmgr_lookup for 'sip.pennytel.com'
> ast_get_srv: SRV lookup for '_sip._udp.sip.pennytel.com' mapped to host sip6.pennytel.com, port 5060
-- Remote UNIX connection
-- Remote UNIX connection disconnected
====================================================================================

Please Suggest...

Regards,
Sagar
 
Joined
Feb 28, 2008
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There is no attempt to use your trunk, it seems to me that you haven't configured correctly your outbound campaigns...
 
Joined
Apr 17, 2011
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Changing nothing except the trunk gets me through!!

This is the CLI



Asterisk 1.6.2.13, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': == Found
== Parsing '/etc/asterisk/extconfig.conf': == Found
Connected to Asterisk 1.6.2.13 currently running on Elastix (pid = 24912)
Verbosity is at least 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
> Channel SIP/6397-0000001d was answered.
-- Executing [*8888100@from-internal:1] Set("SIP/6397-0000001d", "AGENTNUMBER=100") in new stack
-- Executing [*8888100@from-internal:2] NoOp("SIP/6397-0000001d", "AgentNumber is 100") in new stack
-- Executing [*8888100@from-internal:3] AgentLogin("SIP/6397-0000001d", "100") in new stack
-- <SIP/6397-0000001d> Playing 'agent-pass.gsm' (language 'au')
-- <SIP/6397-0000001d> Playing 'agent-loginok.gsm' (language 'au')
-- Started music on hold, class 'default', on SIP/6397-0000001d
== Agent '100' logged in (format ulaw/ulaw)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
> doing dnsmgr_lookup for 'sip.pennytel.com'
> ast_get_srv: SRV lookup for '_sip._udp.sip.pennytel.com' mapped to host sip6.pennytel.com, port 5060
> Channel SIP/pennytel-0000001e was answered.
-- Executing [123@from-internal:1] Macro("SIP/pennytel-0000001e", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/pennytel-0000001e", "AMPUSER=61390952646") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/pennytel-0000001e", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/pennytel-0000001e", "1?Set(REALCALLERIDNUM=61390952646)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/pennytel-0000001e", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/pennytel-0000001e", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/pennytel-0000001e", "1?report") in new stack
-- Goto (macro-user-callerid,s,10)
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/pennytel-0000001e", "0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/pennytel-0000001e", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/pennytel-0000001e", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] Set("SIP/pennytel-0000001e", "CALLERID(number)=61390952646") in new stack
-- Executing [s@macro-user-callerid:20] Set("SIP/pennytel-0000001e", "CALLERID(name)=61390952646") in new stack
-- Executing [s@macro-user-callerid:21] NoOp("SIP/pennytel-0000001e", "Using CallerID "61390952646" <61390952646>") in new stack
-- Executing [123@from-internal:2] Answer("SIP/pennytel-0000001e", "") in new stack
-- Executing [123@from-internal:3] Set("SIP/pennytel-0000001e", "__BLKVM_OVERRIDE=BLKVM/123/SIP/pennytel-0000001e") in new stack
-- Executing [123@from-internal:4] Set("SIP/pennytel-0000001e", "__BLKVM_BASE=123") in new stack
-- Executing [123@from-internal:5] Set("SIP/pennytel-0000001e", "DB(BLKVM/123/SIP/pennytel-0000001e)=TRUE") in new stack
-- Executing [123@from-internal:6] ExecIf("SIP/pennytel-0000001e", "1?Set(_DIAL_OPTIONS=trM(auto-blkvm))") in new stack
-- Executing [123@from-internal:7] Set("SIP/pennytel-0000001e", "__NODEST=123") in new stack
-- Executing [123@from-internal:8] Set("SIP/pennytel-0000001e", "MONITOR_FILENAME=/var/spool/asterisk/monitor/q123-20110526-161327-1306393998.55") in new stack
-- Executing [123@from-internal:9] Queue("SIP/pennytel-0000001e", "123,t,,") in new stack
-- Started music on hold, class 'default', on SIP/pennytel-0000001e
-- Stopped music on hold on SIP/6397-0000001d
-- agent_call, call to agent '100' call on 'SIP/6397-0000001d'
-- <SIP/6397-0000001d> Playing 'beep.gsm' (language 'au')
> doing dnsmgr_lookup for 'sip.pennytel.com'
> ast_get_srv: SRV lookup for '_sip._udp.sip.pennytel.com' mapped to host sip6.pennytel.com, port 5060
-- Agent/100 answered SIP/pennytel-0000001e
-- Stopped music on hold on SIP/pennytel-0000001e
-- Executing [h@from-internal:1] Macro("SIP/pennytel-0000001e", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/pennytel-0000001e", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/pennytel-0000001e", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/pennytel-0000001e", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/pennytel-0000001e", "0?skipblkvm") in new stack
-- Executing [s@macro-hangupcall:8] NoOp("SIP/pennytel-0000001e", "Cleaning Up Block VM Flag: BLKVM/123/SIP/pennytel-0000001e") in new stack
-- Executing [s@macro-hangupcall:9] NoOp("SIP/pennytel-0000001e", "Deleting: BLKVM/123/SIP/pennytel-0000001e TRUE") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/pennytel-0000001e", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/pennytel-0000001e", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/pennytel-0000001e' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/pennytel-0000001e'
== Spawn extension (from-internal, 123, 9) exited non-zero on 'SIP/pennytel-0000001e'
-- Executing [h@from-internal:1] Macro("SIP/pennytel-0000001e", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/pennytel-0000001e", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/pennytel-0000001e", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/pennytel-0000001e", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/pennytel-0000001e", "0?skipblkvm") in new stack
-- Executing [s@macro-hangupcall:8] NoOp("SIP/pennytel-0000001e", "Cleaning Up Block VM Flag: BLKVM/123/SIP/pennytel-0000001e") in new stack
-- Executing [s@macro-hangupcall:9] NoOp("SIP/pennytel-0000001e", "Deleting: BLKVM/123/SIP/pennytel-0000001e ") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/pennytel-0000001e", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/pennytel-0000001e", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/pennytel-0000001e' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/pennytel-0000001e'
== Agent '100' logged out
== Spawn extension (from-internal, *8888100, 3) exited non-zero on 'SIP/6397-0000001d'
-- Executing [h@from-internal:1] Macro("SIP/6397-0000001d", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/6397-0000001d", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/6397-0000001d", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/6397-0000001d", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/6397-0000001d", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/6397-0000001d", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/6397-0000001d", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/6397-0000001d' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/6397-0000001d'

What else can I try?
 
Joined
Apr 16, 2011
Messages
30
Points
0
Hello there,
After a lot of effort managed to make the Callcenter module to work !!! But i have a problem that searched a lot and still found no solution...

When an agent log in to the agent console and the system makes the 1st call during that call after 31 or 32 secs the system auto log him out... and the call is cut off. How can i fix this ? IS there any solution to that ?

Thanks a lot !!!
 
Joined
Apr 18, 2011
Messages
92
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Problem is about your Softphone! I guess that you use X-lite.
I had the same problem.

See this link
Code:
This is what I found. First go to the options and choose Advanced. Then click the network button. Under the section
“Call Inactivity” take a tick out of the box “In times of network disruption, automatically hang up calls after:” (default setting is 30
seconds)
See this link
http://www.comprendo.co.uk/wordpress/co ... 0-seconds/
 
Joined
Apr 16, 2011
Messages
30
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0
i use eyeBeam actually not X-Lite... I search a little bit the options of eyeBeam but no luck ...

The problem occurs only on the callcenter module...When i place a call dialing directly i do not have a problem or when i have incoming call no problem there...

i will try though to use X-Lite and tell back !

thanks !!!!
 
Joined
Apr 18, 2011
Messages
92
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0
Dear friend,
When you run EyeBeam , click on the "Options" >> "Advanced" >> and "Network" and then UNCHECK the following option "in times of network disruption automatically hang up calls after.."

See the screenshoot!



after that If you still have the same problem then try Xlite or another softphone.
 
Joined
Apr 16, 2011
Messages
30
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0
Hello there,
Finally everything works smoothly... The problem with the auto disconnect was sloved by the above solution ...
It is wierd thought that the whole call center enviroment works 100% only with Chrome ... I do not care though... if it does not work with other browsers like Firefox... i will use Chrome for the center module !

Thanks guys for the support !!!
 
Joined
Jul 21, 2011
Messages
6
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0
Hello

I seem to be having the same problem, i have followed all the above and i think i have done everything ok, but still there is no change

Question1
When i log on to my aterix machine i use root - password, on the congiguration tab do i have to use admin - elastix456, and the ip i log in is 192.168.1.3. do i have to change that.

Question 2
Can i use txt file instead of csv

Any help

thankyou
 
Joined
Feb 28, 2008
Messages
5,737
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You need to use the correct information for AMI (callcenter configuration).
That means user: admin password: elastix456
The IP must be 127.0.0.1 NOT your elastix IP address.
You may use your txt files, but before uploading them, you must change them to csv
 
Joined
Jul 21, 2011
Messages
6
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Thankyou, i have changed the user and password, followed all steps but still seem to be having the same problem, when the campaign starts all i hear is music
 
Joined
Sep 30, 2010
Messages
10
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Hi Karma ,
can you give me the Links again seems i am not able to open it ....
 
Joined
Oct 9, 2008
Messages
4
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i have follow all the instructions but when the agent log in for the outgoing campaing all it does is listen to music on hold i dont call

here is the log


tail -f /opt/elastix/dialer/dialerd.log

[0] => Agent/1009
)

2013-05-14 18:22:04 : (AMIEventProcess) DEBUG: AMIEventProcess::_pingAgente: Agent/1009
2013-05-14 18:22:04 : (ECCPProcess) DEBUG: ECCPConn::procesarPaquete: requerimiento pingagent procesado luego de (sec): 0.0016369819641113
 

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