GOautodial Open Source Omni-channel Contact Center Suite

Check if you softphone has timeout enabled. Normally softphones disconnect after 30 seconds of inactivity. Also change the following in /etc/asterisk/sip.conf:

rtptimeout=60         ; Terminate call if 60 seconds of no RTP or RTCP activity

to:

;rtptimeout=60        ; Terminate call if 60 seconds of no RTP or RTCP activity

Let us know if this helps.

These aren't SIP phones - they are connected to our Samsung Officeserve 7400

Then you have something misconfigured with your integration between GOautodial and your Samsumng Officeserve. We won't be able to help if you don't give any details.

I will post the relevant parts of my extensions config when I get to a pc to see if anyone can spot my error. Thanks.

Relevant parts of config :

[default]
include => vicidial-auto

exten => _XXXXXXXXXXXX,1,Dial(${TRUNKPRI}/${EXTEN:2},,tToR)
exten => _XXXXXXXXXXXX,2,Hangup

; dialplan to dial the epabx extensions - in my case epabx extensions is 3 digi$
exten => _2XX,1,Dial(${TRUNKPBX}/${EXTEN},,TtoR)
exten => _2XX,2,Hangup()

(rest in default is all standard)

[from-pstn] ; context to receive incomming calls fromt the service provider ie:port1 of E1 card
; dialplan to route the incomming calls from the service provider
; my provider send last 3 digits from my DID ie: my did is 4004400 so he sends 400 , 401 ...

;exten => _XXX,1,Dial(SIP/${EXTEN}) ; rings my asteirsk sip extension which matches the 3 Digits.
;exten => _XXX,2,Hangup()
; if you want to ring the EPABX extension then use the below dialplan

;exten => _XXXXXX,1,Dial(${TRUNKPBX}/${EXTEN},,TtoR)
;exten => _XXXXXX,2,Hangup()

exten => _XXXXXX,1,AGI
exten => _XXXXXX,2,Dial(${TRUNKPBX}/${EXTEN},,TtoR)
exten => _XXXXXX,3,Hangup()

[from-epabx] inbound cotext to handle the incomming from the epabx as mentioned in dahdi-channels.conf
exten => s,1,Answer()

exten => s,n(start),Read(PHONE,,14,,,)
exten => s,n,GotoIf($[${LEN} = 4]?asterisk)
exten => s,n,GotoIf($[${LEN} > 8]?landline)
;exten => s,n,GotoIf($[${LEN} > 8]?mobile)
exten => s,n(end),Playback(invalid)
exten => s,n(try),Goto(start)
exten => s,n(asterisk),Goto(asterisk,${PHONE},1)
exten => s,n(landline),Goto(landline,${PHONE},1)
exten => s,n(mobile),Goto(mobile,${PHONE},1)

exten => i,1,Playback(invalid)
exten => i,n,Hangup()

exten => t,1,Playback(unavailable)
exten => t,n,Hangup

[asterisk]
exten => _XXXX,1,Dial(SIP/${EXTEN:1}) ; discard first digit as my asterisk is a 3digit extension.
exten => _XXXX,2,Hangup
[landline]
exten => _X.,1,Dial(${TRUNKPRI}/${EXTEN:1},,tToR)
exten => _X.,2,Hangup
[mobile]
exten => _X.,1,Dial(${TRUNKPRI}/${EXTEN},,tToR) ; to dial mobile sometimes need zero prefix
exten => _X.,2,Hangup

Here's a dump of a call which was hung up by the agent :

[Aug 19 09:15:42] -- Executing [8369@default:5] AGI in new stack
[Aug 19 09:15:42] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Aug 19 09:15:42] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Aug 19 09:15:42] -- Executing [192*168*004*043*8600051@default:1] Goto("DAHDI/9-1", "default|8600051|1") in new stack
[Aug 19 09:15:42] -- Goto (default,8600051,1)
[Aug 19 09:15:42] -- Executing [8600051@default:1] MeetMe("DAHDI/9-1", "8600051|F") in new stack
[Aug 19 09:15:44] Parsing '/etc/asterisk/manager.conf': [Aug 19 09:15:44] Found
[Aug 19 09:15:44] Manager 'sendcron' logged on from 127.0.0.1
[Aug 19 09:15:44] -- Executing [58600051@default:1] MeetMe("Local/58600051@default-8e41,2", "8600051|Fmq") in new stack
[Aug 19 09:15:44] > Channel Local/58600051@default-8e41,1 was answered.
[Aug 19 09:15:44] -- Executing [8309@default:1] Answer("Local/58600051@default-8e41,1", "") in new stack
[Aug 19 09:15:44] -- Executing [8309@default:2] Monitor("Local/58600051@default-8e41,1", "wav|") in new stack
[Aug 19 09:15:44] Manager 'sendcron' logged off from 127.0.0.1
[Aug 19 09:15:44] -- Executing [8309@default:3] Wait("Local/58600051@default-8e41,1", "3600") in new stack
[Aug 19 09:15:46] Parsing '/etc/asterisk/manager.conf': [Aug 19 09:15:46] Found
[Aug 19 09:15:46] Manager 'sendcron' logged on from 127.0.0.1
[Aug 19 09:15:46] Spawn extension (default, 8600051, 1) exited non-zero on 'DAHDI/9-1'
[Aug 19 09:15:46] -- Executing [h@default:1] DeadAGI("DAHDI/9-1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Aug 19 09:15:46] Manager 'sendcron' logged off from 127.0.0.1
[Aug 19 09:15:46] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Aug 19 09:15:46] -- Hungup 'DAHDI/9-1'
[Aug 19 09:15:46] Parsing '/etc/asterisk/manager.conf': [Aug 19 09:15:46] Found
[Aug 19 09:15:46] Manager 'sendcron' logged on from 127.0.0.1
[Aug 19 09:15:46] Manager 'sendcron' logged off from 127.0.0.1
[Aug 19 09:15:46] Parsing '/etc/asterisk/manager.conf': [Aug 19 09:15:46] Found
[Aug 19 09:15:46] Manager 'sendcron' logged on from 127.0.0.1
[Aug 19 09:15:46] Manager 'sendcron' logged off from 127.0.0.1
[Aug 19 09:15:46] Parsing '/etc/asterisk/manager.conf': [Aug 19 09:15:46] Found
[Aug 19 09:15:46] Manager 'sendcron' logged on from 127.0.0.1
[Aug 19 09:15:46] Manager 'sendcron' logged off from 127.0.0.1
[Aug 19 09:15:46] Spawn extension (default, 8600051, 1) exited non-zero on 'DAHDI/33-1'
[Aug 19 09:15:46] -- Hungup 'DAHDI/pseudo-433651915'
[Aug 19 09:15:46] -- Executing [h@default:1] DeadAGI("DAHDI/33-1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Aug 19 09:15:46] Spawn extension (default, 58600051, 1) exited non-zero on 'Local/58600051@default-8e41,2'
[Aug 19 09:15:46] -- Executing [h@default:1] DeadAGI("Local/58600051@default-8e41,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Aug 19 09:15:46] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Aug 19 09:15:46] -- Hungup 'DAHDI/33-1'
[Aug 19 09:15:46] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Aug 19 09:15:46] == Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600051@default-8e41,1'
[Aug 19 09:15:46] -- Executing [h@default:1] DeadAGI("Local/58600051@default-8e41,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Aug 19 09:15:46] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0

Any help on this? It's obviously hanging up the pseudo channel and the outbound PRI from the conference - I am sure this is wrong but no matter how much googling or digging I do no-one on the planet seems to have ever had this issue?!

Pretty sure the issue lies here :

[Aug 20 12:58:25] Parsing '/etc/asterisk/manager.conf': [Aug 20 12:58:25] Found
[Aug 20 12:58:25] Manager 'sendcron' logged on from 127.0.0.1
[Aug 20 12:58:25] Manager 'sendcron' logged off from 127.0.0.1
[Aug 20 12:58:25] Spawn extension (default, 8600051, 1) exited non-zero on 'DAHDI/2-1'
[Aug 20 12:58:25] -- Executing [h@default:1] DeadAGI("DAHDI/2-1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Aug 20 12:58:25] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Aug 20 12:58:25] -- Hungup 'DAHDI/2-1'
[Aug 20 12:58:25] Parsing '/etc/asterisk/manager.conf': [Aug 20 12:58:25] Found
[Aug 20 12:58:25] Manager 'sendcron' logged on from 127.0.0.1
[Aug 20 12:58:25] Manager 'sendcron' logged off from 127.0.0.1
[Aug 20 12:58:25] Parsing '/etc/asterisk/manager.conf': [Aug 20 12:58:25] Found
[Aug 20 12:58:25] Manager 'sendcron' logged on from 127.0.0.1
[Aug 20 12:58:25] Manager 'sendcron' logged off from 127.0.0.1
[Aug 20 12:58:25] -- Hungup 'DAHDI/pseudo-837752789'
[Aug 20 12:58:25] == Spawn extension (default, 8600051, 1) exited non-zero on 'DAHDI/34-1'
[Aug 20 12:58:25] -- Executing [h@default:1] DeadAGI("DAHDI/34-1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Aug 20 12:58:25] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Aug 20 12:58:25] -- Hungup 'DAHDI/34-1'

It is cutting the PRI conection (DAHDI/2-1), and then proceeding to disconnect the Phonesystem to Asterisk connection (DAHDI/34-1). I guess I just need to find out how to stop it doing this :(

Have you checked your T1 configurations? Might be the clock signalling which can cause odd behaviors.

# All T1/E1/BRI spans generate a clock signal on their transmit side. The
# <timing source> parameter determines whether the clock signal from the far
# end of the T1/E1/BRI is used as the master source of clock timing. If it is, our
# own clock will synchronise to it. T1/E1/BRI connected directly or indirectly to
# a PSTN provider (telco) should generally be the first choice to sync to. The
# PSTN will never be a slave to you. You must be a slave to it.

#span=1,0,0,esf,b8zs
#span=2,1,0,esf,b8zs
#span=3,0,0,ccs,hdb3,crc4

post your dahdi settings
/etc/dahdi/system.conf
/etc/asterisk/chan_dahdi.conf
/etc/asterisk/dahdi-channels.conf

System.conf :

#autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
#autogenrated on 2013-08-04
#Dahdi Channels Configurations
#For detailed Dahdi options, view /etc/dahdi/system.conf.bak
loadzone=us
defaultzone=us

#Sangoma A102 port 1 [slot:4 bus:13 span:1] <wanpipe1>
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
echocanceller=mg2,1-15,17-31
hardhdlc=16

#Sangoma A102 port 2 [slot:4 bus:13 span:2] <wanpipe2>
span=2,2,0,ccs,hdb3,crc4
bchan=32-46,48-62
echocanceller=mg2,32-46,48-62
hardhdlc=47

Chan_dahdi.conf :

;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
;autogenrated on 2013-08-04
;Dahdi Channels Configurations
;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

;Sangoma A102 port 1 [slot:4 bus:13 span:1] <wanpipe1>
switchtype=euroisdn
context=from-pstn
group=0
echocancel=yes
faxdetect=incoming
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

;Sangoma A102 port 1 [slot:4 bus:13 span:1] <wanpipe1>
switchtype=euroisdn
context=from-pstn
group=0
echocancel=yes
faxdetect=incoming
signalling=pri_cpe
channel =>1-15,17-31
context=default
group = 63

;Sangoma A102 port 2 [slot:4 bus:13 span:2] <wanpipe2>
switchtype=euroisdn
context=from-epabx
group=1
echocancel=yes
signalling=pri_net
channel =>32-46,48-62
context=default
group = 63

dahdi-channels.conf :

There isn't one in the location given, only /etc/wanpipe/wancfg_zaptel/templates/dahdi-channels.conf

That's got nothing of note in it.

Demian Biscocho wrote:

Have you checked your T1 configurations? Might be the clock signalling which can cause odd behaviors.

[...]

Should I have set the port that talks to the phone system (not telco) as a master?

This really depends on your T1 line. Most telco provides the master clock signaling. Have you checked with your Telco?

Can you provide a detailed diagram of your system? How does your phone connect to your GOautodial system via SamsungOfficeserve?

with ref to the confs posted above - there is a bit of overlap caused by text copy (there's only 1 span 1)

BT ISDN30 --> Port A Sangoma

Port B Sangoma --> P1 Samsung OfficeServ

All functions work apart from the fact it kicks you out of the conference after you click hangup, thus breaking the link between web gui and phone.

Both ports are set to NORMAL for the clock (assume makes pass through).

You need to have the call_log entry in your outgoing trunk for Vicidial to function properly:

exten => _XXXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXXXXX,2,Dial(${TRUNKPRI}/${EXTEN:2},,tToR)
exten => _XXXXXXXXXXXX,3,Hangup

Though I'm not sure if this is related to the issue you're having. You can also try disabling the campaign recordings and see if their is a difference.

ok, second port now set to MASTER. Dialplan changed as above, call recording off on the campaign. Still no change :(

External phone system - in order to make the webgui call the phone I have to name the extensions g1/xxx - I kow this must be wrong. If I just put 278 it says it can't open DAHDI/278 - which is correct as it doesn't which channel to connect on - how do I tell it that g1 is the phone system?