vicidial.org

Hello guys,

I have setup an inbound group and point an external number to take inbound calls but don't know why, inbound calls will always be answered by asterisk with "invalid" extension message and hangup. It will be very much grateful if anyone can help.

My system setup are:
Vicidial Version: 2.0.4-122 Build: 81011-0855
Asterisk: 1.2.30
OS: CentOS 5.2
Trunk interface: Digium AX400P (, 1 FXO on channel 4)

/etc/zapata.conf
[channels]
context=default
signalling=fxs_ks
channel => 1-4


campaign ID: TEST_IN
campaign name: Closer and inbound campaign
active: Y
allow closers: Y
hopper level: 5
dial method: RATIO
auto dial level: 1
next agent call: oldest call finish
local call time: 24hours


Here is extension setting
exten => 106,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 106,2,Dial(newsip:test@192.168.19.2:5060/${EXTEN:2},,tTor)
exten => 106,3,Hangup


Asterisk CLI>
-- Starting simple switch on 'Zap/4-1'
Jul 23 07:22:39 NOTICE[31370]: chan_zap.c:6248 ss_thread: Got event 18 (Ring Begin)...
Jul 23 07:22:40 NOTICE[31370]: chan_zap.c:6248 ss_thread: Got event 2 (Ring/Answered)...
== Starting Zap/4-1 at pstn_incoming,s,1 failed so falling back to exten 's'
== Starting Zap/4-1 at pstn_incoming,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on Zap/4-1
-- Executing Playback("Zap/4-1", "invalid") in new stack
-- Playing 'invalid' (language 'en')
-- Timeout on Zap/4-1
== CDR updated on Zap/4-1
-- Executing Goto("Zap/4-1", "#|1") in new stack
-- Goto (default,#,1)
-- Executing Playback("Zap/4-1", "invalid") in new stack
-- Playing 'invalid' (language 'en')
-- Executing Hangup("Zap/4-1", "") in new stack
== Spawn extension (default, #, 2) exited non-zero on 'Zap/4-1'
-- Executing DeadAGI("Zap/4-1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Hungup 'Zap/4-1'
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- Starting simple switch on 'Zap/4-1'
Jul 23 07:22:39 NOTICE[31370]: chan_zap.c:6248 ss_thread: Got event 18 (Ring Begin)...
Jul 23 07:22:40 NOTICE[31370]: chan_zap.c:6248 ss_thread: Got event 2 (Ring/Answered)...
== Starting Zap/4-1 at pstn_incoming,s,1 failed so falling back to exten 's'
== Starting Zap/4-1 at pstn_incoming,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on Zap/4-1
-- Executing Playback("Zap/4-1", "invalid") in new stack
-- Playing 'invalid' (language 'en')
-- Timeout on Zap/4-1
== CDR updated on Zap/4-1
-- Executing Goto("Zap/4-1", "#|1") in new stack
-- Goto (default,#,1)
-- Executing Playback("Zap/4-1", "invalid") in new stack
-- Playing 'invalid' (language 'en')
-- Executing Hangup("Zap/4-1", "") in new stack
== Spawn extension (default, #, 2) exited non-zero on 'Zap/4-1'
-- Executing DeadAGI("Zap/4-1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Hungup 'Zap/4-1'
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

pls can anybody help me out i'm stucked here?

You don't seem to have a pstn_incoming context in your dialplan.

Thanks for your response

I'v created pstn_incoming in the dial plan , but on dial into the vicidial from the PSTN i saw this info displayed on the asterisk CLI>


-- Starting simple switch on 'Zap/4-1'
Jul 23 23:20:36 NOTICE[19420]: chan_zap.c:6248 ss_thread: Got event 18 (Ring Begin)...
Jul 23 23:20:37 NOTICE[19420]: chan_zap.c:6248 ss_thread: Got event 2 (Ring/Answered)...
-- Executing AGI("Zap/4-1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Zap/4-1", "/|55|o") in new stack
Jul 23 23:20:37 WARNING[19420]: channel.c:2621 ast_request: No channel type registered for ''
Jul 23 23:20:37 NOTICE[19420]: app_dial.c:1076 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("Zap/4-1", "") in new stack
== Spawn extension (pstn_incoming, s, 3) exited non-zero on 'Zap/4-1'
-- Hungup 'Zap/4-1'
-- Starting simple switch on 'Zap/4-1'
Jul 23 23:20:50 WARNING[19447]: chan_zap.c:6324 ss_thread: CallerID returned with error on channel 'Zap/4-1'
-- Executing AGI("Zap/4-1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Zap/4-1", "/|55|o") in new stack
Jul 23 23:20:50 WARNING[19447]: channel.c:2621 ast_request: No channel type registered for ''
Jul 23 23:20:50 NOTICE[19447]: app_dial.c:1076 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("Zap/4-1", "") in new stack
== Spawn extension (pstn_incoming, s, 3) exited non-zero on 'Zap/4-1'
-- Hungup 'Zap/4-1'
Does anyone know the way out pls?

I see you created it, but what did you put in it?

This is my configuration
extension.conf
[pstn_incoming]
exten => s,1,AGI(agi://127.0.0.1:4577/call_log)
exten => s,2,Dial(${SIP/201:test@192.168.19.2:5060}/${EXTEN:2},55,o)
exten => s,3,Hangup

Hello Matt, thanks for your concern, I've finally solved my problem .
This is what i did to my extension.conf.

[pstn_incoming]
exten => s,1,Answer()
exten => s,2,dial(SIP/201,20,tr)
exten => s,3,Hangup
Firstly, the voice of the caller is not audible and its also cracking.
the dial plan only work for the extension in sip.conf , the question is how to synchronize it with the existing extensions in sip-vicidial.conf can anybody help me pls?