GOautodial Open Source Omni-channel Contact Center Suite

To follow up on this, I can see this error on javascript console when I click "Login to Dialer".

POST https://10.10.0.226/GOagentJS.php 404 (Not Found)

Hi,

When you login do you hear a voice prompt confirmation "You are the only person in this conference". Can you also post your asterisk CLI log upon agent login and while attempting a call.

I dont hear the voice prompt confirmation when i login. Here is asterisk cli log upon agent login

-- Called 99991540835923@default
    -- Executing [99991540835923@default:1] Dial("Local/99991540835923@default-00000006;2", "SIP/1540835923@kamailio,,tTo") in new stack
== Using SIP RTP CoS mark 5
    -- Called SIP/1540835923@kamailio
[Dec  7 10:29:33] WARNING[16562][C-00000008]: chan_sip.c:24003 handle_response_invite: Received response: "Forbidden" from '"S1812071029328600051" <sip:[email protected]:5070>;tag=as2f5a57eb'
   Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [99991540835923@default:2] Hangup("Local/99991540835923@default-00000006;2", "") in new stack
   Spawn extension (default, 99991540835923, 2) exited non-zero on 'Local/99991540835923@default-00000006;2'
    -- Executing [h@default:1] AGI("Local/99991540835923@default-00000006;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CHANUNAVAIL---------------SIP 403 Not relaying)") in new stack
    -- <Local/99991540835923@default-00000006;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CHANUNAVAIL---------------SIP 403 Not relaying) completed, returning 0

An update.

I took the kamailio.cfg from https://goautodial.org/boards/3/topics/13730?r=14098 and then it worked, I did not get "You are only person..." but when I dial a number the number gets connected and I am able to hear the other party but I am not able to hear any ringing/busy audio.

Post asterisk cli log upon login and post the output of core show channels/sip show channels

Asterisk cli on login

Connected to Asterisk 13.17.2-vici currently running on vaglxc01 (pid = 8884)
   Manager 'sendcron' logged on from 127.0.0.1
    -- Called 99991540835923@default
    -- Executing [99991540835923@default:1] Dial("Local/99991540835923@default-00000012;2", "SIP/1540835923@kamailio,,tTo") in new stack
   Using SIP RTP CoS mark 5
    -- Called SIP/1540835923@kamailio
    -- SIP/kamailio-0000000b is ringing
    -- Local/99991540835923@default-00000012;1 is ringing
       > 0x7f229c00e170 -- Strict RTP learning after remote address set to: 10.10.0.52:30390
    -- SIP/kamailio-0000000b answered Local/99991540835923@default-00000012;2
    -- Local/99991540835923@default-00000012;1 answered
    -- Executing [8600051@default:1] Konference("Local/99991540835923@default-00000012;1", "8600051,R") in new stack
 == Manager 'sendcron' logged off from 127.0.0.1
    -- Channel SIP/kamailio-0000000b joined 'simple_bridge' basic-bridge <16a639b1-aa4c-4ade-bf50-fe8306d05b60>
    -- Channel Local/99991540835923@default-00000012;2 joined 'simple_bridge' basic-bridge <16a639b1-aa4c-4ade-bf50-fe8306d05b60>
       > 0x7f229c00e170 -- Strict RTP switching to RTP target address 10.10.0.52:30390 as source
       > 0x7f229c00e170 -- Strict RTP learning complete - Locking on source address 10.10.0.52:30390

Core show channels

vaglxc01*CLI> core show channels
Channel Location State Application(Data)
Local/99991540835923 99991540835923@defau Up Dial(SIP/1540835923@kamailio,,
Local/99991540835923 8600051@default:1 Up Konference(8600051,R)
SIP/kamailio-0000000 (None) Up AppDial((Outgoing Line))
SIP/kamailio-0000000 (None) Up AppDial((Outgoing Line))
Local/99991540835923 99991540835923@defau Up Dial(SIP/1540835923@kamailio,,
Local/99991540835923 8600051@default:1 Up Konference(8600051,R)
6 active channels
4 active calls
43 calls processed

SIP show channels

vaglxc01*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
127.0.0.1 1540835923 1e99b8ae1249684 (ulaw) No Tx: ACK kamailio
127.0.0.1 1540835923 0906ced81d3b74f (ulaw) No Rx: INFO kamailio
2 active SIP dialogs

I am not able to hear "You are only person..." msg, also DTMF does not work, in asterisk CLI i can see the DTMF is detected properly but its not passed on to the provider. DTMF is set to dtmfmode = rfc2833 and relaxdtmf=yes

Hi,

Can you please give us the output of "ps aux | grep listener.pl".

Thank you!

Hi,

The output should look like this.

[root@vaglxc01 ~]# ps aux | grep listener.pl
root 2686 0.0 0.3 196924 12388 pts/0 S 10:00 0:00 /usr/bin/perl /usr/local/bin/listener.pl
root 3098 0.0 0.0 112704 972 pts/0 S+ 10:04 0:00 grep --color=auto listener.pl

Thank you!

Hello Jackie

Sorry for the delay. I did not have listener.pl running. So i started it manually, but still the same, cannot hear "You are only person..." audio. Here is the output while starting

[root@vaglxc01 ~]# /usr/bin/perl /usr/local/bin/listener.pl
  Asterisk 13.17.2-vici semicolon hack, engage!
  [root@vaglxc01 ~]# ps aux | grep listener.pl
  root      9217  0.0  0.6 196928 12312 pts/8    S    19:55   0:00 /usr/bin/perl /usr/local/bin/listener.pl
  root      9233  0.0  0.0 112704   984 pts/8    S+   19:55   0:00 grep --color=auto listener.pl

Hello Jackie

Thanks, after starting listener.pl I can hear "You are only person...". Now only DTMF does not work, actually on debugging the server receives DTMF properly but on the remote end, one or 2 digits gets missed. Everytime 0 was not detected on the remote end but detected properly on the Goautodial box.

Hi,

Are you using the DTMF for outbound? if yes please provide us your carrier settings for us to review your config.

Thanks

Following are the carrier settings

[Twilio]
type=peer
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
insecure= port ,invite
host=xxxx.pstn.twilio.com
fromuser=+1XXXXXXXXXX
nat=yes
externhost=xxx.xx.x.xxx
localnet=10.10.0.0/24

GLobalString: DIAL9TRUNK = SIP/Twilio

Dialplan entry

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,n,Dial(${DIAL9TRUNK}/+1${EXTEN:2},,To)
exten => _91NXXNXXXXXX,n,Hangup

exten => _9NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXNXXXXXX,n,Dial(${DIAL9TRUNK}/+1${EXTEN:1},,tTo)
exten => _9NXXNXXXXXX,n,Hangup

I have installed Goautodial Version 4 using ISO.
had issue upon agent login no audio of "You are the only person in this conf"
Tried with kamalio.cfg from https://goautodial.org/boards/3/topics/13730?r=14098
but its not worked for me, so i reverted to default.

---
[root@go401 bal]# ps aux | grep listener.pl
root 686 0.0 0.0 112704 980 pts/8 S+ 05:26 0:00 grep --color=auto listener.pl
root 1965 0.0 0.0 196920 12484 ? S 2018 0:00 /usr/bin/perl /usr/local/bin/listener.pl
[root@go401 bal]#
----

Hi Bal Jidokasys,

Did you change the server IP? What are the changes you made after installing Goautodial Version 4?

Thanks

Hi Vinay Ramdas,

Try to change dtmfmode:

From:
dtmfmode=rfc2833

To:
dtmfmode=info

Thanks