Call status is not updated

[hemanshurpatel]

Hello
i have installed VICIdialer with asterisk
i have followe dall scratch install .

now what happen when say a agent logs in the agent phone rings, when agent picks up a meetme generets.its ok till now
i have manual dialing, so when agend clicks on dial now, it diales a no to PRI line, for testing purpose it is dialing my mobile no.
i got call on my mobile and pics it up, still in vicidial agents window it shows ringing, states incall is not updated.
it shows ringing and then timeout.

what can be the problem?

Regards
Hemanshu Patel

[mflorell]

Did you apply the cli concise patch to Asterisk before compiling it?

Asterisk verison?

astguiclient version?

[hemanshurpatel]

yes i did apply cli patch

asterisk version 1.4.X
Latest Vicidial version

Stll having the same problem

[hemanshurpatel]

hello
i have downgrade to asterisk version 1.2.23
and installed appropriate libpri and zaptel driver

Still condition is same.
Whne agent logs in he got ring back, and then the user's (customer) no is dialed.
Even if user pics that call up asterisk shows ringing only

Please help somebody, i am stuck with this.

Regards,
Hemanshu Patel

[mflorell]

So your carrier is not sending an Answer signal?

Do you see "Channel X answered" on your Asterisk CLI when this happens?

[hemanshurpatel]

yes
i do see that channels is answered message.
when my agent logs in he got ring back, meet me is created and other party, which is me via my mobile in this case is called and connected

Still that call status is showing wait for ringing only

Any Idea what could be the Problem?

[hemanshurpatel]

the vicidialer version i am using is 2.0.4

People who r using this version are having same problem or ur dialer is working fine?

[mflorell]

Do you have the AGI(agi://127.0.0.1...call_log--... entries prior to Dials in your extensions.conf?

Your problem is usually a configuration problem.

[hemanshurpatel]

yes dear
i do have that entry
i have not left even a single installation stuff, even though its not working
i have no clue what so ever regarding that.

Looking for help.

Hemanshu.

[hemanshurpatel]

asterisk CLI is:

== Manager 'sendcron' logged off from 58.68.36.182
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 58.68.36.182
== Refreshing DNS lookups.
> Channel SIP/100000000-0090b650 was answered.
-- Executing MeetMe("SIP/100000000-0090b650", "8600051") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Manager 'sendcron' logged off from 58.68.36.182
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 58.68.36.182
-- Executing MeetMeAdmin("Local/118600051@default-2648,2", "8600051|M|1") in new stack
-- Executing Hangup("Local/118600051@default-2648,2", "") in new stack
== Spawn extension (default, 118600051, 2) exited non-zero on 'Local/118600051@default-2648,2'
-- Executing DeadAGI("Local/118600051@default-2648,2", "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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 58.68.36.182
-- Executing MeetMe("Local/8600051@default-0ed7,2", "8600051") in new stack
> Channel Local/8600051@default-0ed7,1 was answered.
-- Executing AGI("Local/8600051@default-0ed7,1", "AGI(agi://127.0.0.1:4577/call_log") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/AGI(agi://127.0.0.1:4577/call_log
Failed to execute '/var/lib/asterisk/agi-bin/AGI(agi://127.0.0.1:4577/call_log': No such file or directory
-- AGI Script AGI(agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-0ed7,1", "Zap/g1/9898698835") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/9898698835
-- Zap/1-1 is proceeding passing it to Local/8600051@default-0ed7,1
== Manager 'sendcron' logged off from 58.68.36.182
Jan 21 10:37:35 NOTICE[12791]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 58.68.36.180
== Manager 'sendcron' logged off from 58.68.36.182
-- Zap/1-1 is ringing
-- Zap/1-1 answered Local/8600051@default-0ed7,1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 58.68.36.182
== Parsing '/etc/asterisk/manager.conf': Found

[hemanshurpatel]

[root@rencsc-3 asterisk]# screen -ls
There are screens on:
11119.ASTlisten (Detached)
12423.ASTfastlog (Detached)
11113.ASTupdate (Detached)
12421.ASTVDadapt (Detached)
11122.ASTVDauto (Detached)
11116.ASTsend (Detached)
6 Sockets in /tmp/screens/S-root.

[root@rencsc-3 asterisk]#

[hemanshurpatel]

Please help guys

My vicidialer is dialing calls to outside world, but somehow, the call status is not updated, the agent is not transferred the live calls, so what happening is i am paying for a call but my agent is not been able to talk.

[mflorell]

What kind of trunks are you using? POTS or PRI?

[hemanshurpatel]

I am using PRI for outgoing calls,
and for agents its SIP onlywith g729 codec
that is not problem i guess

because say if my agents phone no is 1000 then from a registered 1000 phone i can make calls to anyother phone using PRI line, it works
but when an agent logs in to vicidial server with 1000 phone it doesnt work
phone rings, and when my agent picks it us it still shows waiting for ringing and then after 60 secs timeout is there.
during that time u can make a call(manual dial), that call is established from PRI end , but from agents end his phone is still not conencted, it still shows waiting for ringing.

[mflorell]

Can you set the server's agi output to BOTH and post some agi-VDADtransfer output from the agiout.2008-xx-xx logfile?

[hemanshurpatel]

Hey i worked for me
surprisingly i didi nothing except changing the phone
in that phone i was not been able to hera a sound from other person, so i decided to change phone, and as fast as i change the phone my problem solved.

Now the next problem is AMD detection.
is AMD detection is for sip/iax trunks only, can it be detected on zap lines?
I have added all rewuired dila plan 8369 in extensions.conf file, have change VDAD extention to 8369 in campaing settings, and put AGIlog to both

still nothing happens
and not ADM extension calls?


Can u tell me what would have gone wrong?

[mflorell]

AMD works best on Zap T1/E1 lines actually.

As for whether it is working, did you patch your system with the app_amd.so before compiling?

What is the Asterisk CLI output when you place a call?