vicidial.org

Hello,
I have goautodial 2.0 and i have configured a sip trunk. the manual call work perfectly but when all agent are logged and i start the compagain the cunsumer answser and the call are hanguped immediatly and no agent receve call . ( with an iax trunk all work perfectly /manual call and crm )

Any one can help me plz
Thx

Hi.
I've read this before, I did some googling and found these two threads
viewtopic.php?t=24641
viewtopic.php?t=14679
maybe they will help you, if not then post the output from asterisk CLI when a call is failing, version/build, dialplan etc

Hello, it still not working.. i will put all info:



VERSION: 2.2.0-5 BUILD: 90508-0727

Dialplan/Carriers ==>


Carrier ID: trunk-ipcall
Carrier Name: trunk-ipcall
Carrier Description: trunk-ipcall
Registration String: register => user:pass@77.243.222.229

Account Entry:

[trunk-ipcall]
disallow=all
allow=ulaw
type=friend
secret=pass
username=user
host=77.243.222.229
dtmfmode=rfc2833
context=trunk-ipcall
insecure=very

Protocol: SIP

Globals String: trunkipcall = sip/trunk-ipcall
Dialplan Entry:
exten => _87733.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _87733.,2,Dial(${trunkipcall}/5040${EXTEN:3},,tToR)
exten => _87733.,3,Hangup

CLI ==>

[Oct 17 15:31:05] -- Executing [87733547142221@default:1] AGI("Local/87733547142221@default-553c,2", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 17 15:31:05] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 17 15:31:05] -- Executing [87733547142221@default:2] Dial("Local/87733547142221@default-553c,2", "sip/trunk-ipcall/504033547142221||tToR") in new stack
[Oct 17 15:31:05] -- Called trunk-ipcall/504033547142221
[Oct 17 15:31:06] == Parsing '/etc/asterisk/manager.conf': [Oct 17 15:31:06] Found
[Oct 17 15:31:06] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 17 15:31:06] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 17 15:31:06] ERROR[1272]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Oct 17 15:31:06] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 17 15:31:08] -- SIP/trunk-ipcall-00000058 answered Local/87733547142221@default-553c,2
[Oct 17 15:31:08] > Channel Local/87733547142221@default-553c,1 was answered.
[Oct 17 15:31:08] -- Executing [8368@default:1] Playback("Local/87733547142221@default-553c,1", "sip-silence") in new stack
[Oct 17 15:31:08] -- <Local/87733547142221@default-553c,1> Playing 'sip-silence' (language 'en')
[Oct 17 15:31:08] WARNING[1316]: file.c:1292 waitstream_core: Unexpected control subclass '-1'
[Oct 17 15:31:08] -- Executing [8368@default:2] AGI("Local/87733547142221@default-553c,1", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 17 15:31:08] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 17 15:31:08] -- Executing [8368@default:3] AGI("Local/87733547142221@default-553c,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Oct 17 15:31:08] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Oct 17 15:31:09] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 17 15:31:09] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Oct 17 15:31:09] -- Executing [8368@default:4] AGI("Local/87733547142221@default-553c,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Oct 17 15:31:09] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Oct 17 15:31:10] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Oct 17 15:31:10] -- Executing [8368@default:5] Hangup("Local/87733547142221@default-553c,1", "") in new stack
[Oct 17 15:31:10] == Spawn extension (default, 8368, 5) exited non-zero on 'Local/87733547142221@default-553c,1'
[Oct 17 15:31:10] -- Executing [h@default:1] DeadAGI("Local/87733547142221@default-553c,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Oct 17 15:31:10] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Oct 17 15:31:10] -- Executing [h@default:1] DeadAGI("Local/87733547142221@default-553c,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----5-----2") in new stack
[Oct 17 15:31:10] -- AGI Script agi://1s27.0.0.1:4577/call_log--HVcause ... ---5-----2 completed, returning 0
[Oct 17 15:31:10] == Spawn extension (default, 87733547142221, 2) exited non-zero on 'Local/87733547142221@default-553c,2'

any solution ?!! thxx

Can you post a bit more of the asterisk CLI, I dont see the part when the call gets hung up.

and you said that IAX2 trunk works but not SIP, right?
have you open the ports for RTP/RTMP? enable sip debug and look at the CLI when your try to dial
did you look at the topics I linked to? viewtopic.php?t=24641

Hi,
the hangup part is: Oct 17 15:31:10 please see the cli log

yes IAX2 trunk work perfectly manual call and vicidial crm( compaign work and agents recive call). SIP work only manual call with prefix 877... and on a compaign the agent don't receive call ..

Yes RTP/RTMP enabled
i looked the link but it still not working ...


thx

i enabled the sip debug and here on this link http : / / www . heypasteit . com / clip / 0J8C all the log ( i enabled 1 agent and predictif are working with Auto Dial Level:1 to test )
(plz remove spaces)

here is the log in anyone else want to look at it http://www.heypasteit.com/clip/0J8C
"Audio is at 192.168.1.253 port 11816"
I ask again, are the ports in the router/firewall for the audio open?

Hi
yes the audio port is open
NAT actived on 10000 -> 20000 to the sever vicidial
i think if the nat is not actived the manual call don't work is it it ? but the manual call work perferctly i can speak with cunsumer

I saw a user agent string like freeSWITCH, is that yours?

i don't have freeSWITCH i have a agent or sipphone connected (Counterpath bria like xlite)... i think freeSWITCH is getted from the trunk provider

hi, i send you a cli log a part where i put a manual call to see that the call i working:
(Agent logged--> pause--> MANUAL DIAL-->Dial Now )

see here http : // www . heypasteit . com /clip/0JBM thanks (remove space thx)

hi,
i use pfsense like firewall .. it can be the problem ? .. any one can help me please

Nope, I use pfSense too.
But I dont forward ports, I just allowed all traffic from the providers ip address. (my vicidial server got a external ip)
I dont know what the problem with your setup is, just hope someone else looks at this topic and might find some detail I missed.

you can try adding "nat=yes" to your account entry and look carefully at the sip debug

can you check with someother provider, so that you come to know whether it is problem with provider or from your side.

Hi,
yes striker i tested anodher sip provider and its work .

DomeDan I do the same think on my firewall so the vicidial server are all visible to the provider

( on the wan rules * 77.243.222.229 * * * MYVICIDIAL_IP none )
i added also
externip = my_public_ip ; Address that we're going to put in outbound SIP
localnet=192.168.1.0/255.255.255.0; All RFC

AND AFTER THAT i put 2 TEST MANUAL CALL (WORKING) and CALL via crm vicidial (CALL SENT .. ANSWERED AND HANGUPED)
i think the problem is on the provider... can you see the problem on the cli log ?.. Why the call work only on manual ??? what is the differance between manual and the automatic crm call ? can i change it to have a call like manual? thx
I have here the CLI Log
==>
http : // www . heypasteit . com /clip/0JHU

When you said it like that I remember that its was the first problem I encountered and had to ask about on this forum
viewtopic.php?f=8&t=16560
A reinstall solved my problem

The difference between auto and manual is the meetme plan I think... someone with better experience are welcome to fill in.

If an other sip provider worked then it sounds like you need to tweak the account entry,
ask the provider if they got any idea why your config wont work

@matrix1233
i do once faced same problem long before, after long chat with the provider , he said "your account is not in dialler mode".
once he enabled the voip account in dialler mode it start worked.
no idea what does it mean dialler mode, but it worked after that.