Transfer button disabled on manual dial

[tusharcall4]

hi......evryone;

my vicidial version : 2.7 rc1
asterisk version :1.8.22

whenever i make a manual dial even without override option ;the transfer button,park button are disabled. but the same are enabled for inbound & autodial calls.
Also on manual dial, the "NO LIVE CALL" on the agent screen does not get updated to "LIVE CALL" even though the call is established. I think maybe both the issues are related.

i am using a sip trunk from another asterisk server with asterisk version 1.4.39 with digium TE121 pri card.
sip trunk configuration under carrier are as follows:

register string as :
Register=> cc120:test@192.168.1.7/662XXXXX

trunk configuration :

[662XXXXX]
disallow=all
allow=ulaw
allow=alaw
allow=gsm
type=friend
username=cc120
secret=test
host=192.168.1.7
fromuser=cc120
Authuser=cc120
insecure=invite
context=trunkinbound

global string :
TESTSIPTRUNKP = SIP/6620XXXX

dialplan entry for trunk under carrier :
exten => _XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXXX,2,Dial(${TESTSIPTRUNKP}/${EXTEN},60,To)
exten => _XXXXXXXXXX,3,Hangup

my cli shows following output on manual dial :
-- Executing [986XXXXXXX@default:2] Dial("Local/8600053@default-000000a0;1", "SIP/66XXXXXX/986XXXXXXX,60,To") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/66XXXXXX/986XXXXXXX
-- SIP/66XXXXXX-00000057 is making progress passing it to Local/8600053@default-000000a0;1
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [58600053@default:1] MeetMe("Local/58600053@default-000000a1;2", "8600053,Fmq") in new stack
> Channel Local/58600053@default-000000a1;1 was answered.
-- Executing [8309@default:1] Answer("Local/58600053@default-000000a1;1", "") in new stack
-- Executing [8309@default:2] Monitor("Local/58600053@default-000000a1;1", "wav,20130703-185437_986XXXXXXX") in new stack
-- Executing [8309@default:3] Wait("Local/58600053@default-000000a1;1", "3600") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/66XXXXXX-00000057 answered Local/8600053@default-000000a0;1
any help on this issue will be appreciated
thanks in advance.........

[williamconley]

Did this work properly before you upgraded to asterisk 1.8?

also: you listed your vicidial major version number, but nothing else. LOL

What method did you use to install? What is the vicidial build number? Perhaps we should start here:

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

[tusharcall4]

We are using scratch install with following configuration:

We are using free Manager Manual available on Vicidial Website.

AstguiClient VERSION: 2.7-401a BUILD: 130508-2256 | Asterisk 1.8.22 with Asterisk AddOns | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel XEON QUAD Core 8GB RAM, 500GB HD Tower Server. CentOS 6.3 and No clustering | Dahdi version 2.7 + 2.7 complete but no pri card.

We are using a SIP Trunk from our other server with configuration AstguiClient 2.4-364a BUILD: 120409-1136| Asterisk 1.4.39 With Asterisk AddOns | Digium Wildcard TE121 PRI Card | No Extra Software After Installation | Intel XEON Quad Core , 4GB RAM, 500GB HD, CentOS 6.3 | Dahdi 2.6+2.6 complete.

We are not using a virtual or cloud server for any of the server.

Our SIP Trunk Settings as entered in the Vicidial Carrier Configuration:

register string as :
Register=> cc120:test@192.168.1.7/662XXXXX

trunk configuration :

[662XXXXX]
disallow=all
allow=ulaw
allow=alaw
allow=gsm
type=friend
username=cc120
secret=test
host=192.168.1.7
fromuser=cc120
Authuser=cc120
insecure=invite
context=trunkinbound

global string :
TESTSIPTRUNKP = SIP/6620XXXX

dialplan entry for trunk under carrier :
exten => _XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXXX,2,Dial(${TESTSIPTRUNKP}/${EXTEN},60,To)
exten => _XXXXXXXXXX,3,Hangup

And my problem is that whenever i make a manual dial even without override option ;the transfer button,park button are disabled. but the same are enabled for inbound & autodial calls.
Also on manual dial, the "NO LIVE CALL" on the agent screen does not get updated to "LIVE CALL" even though the call is established. I think maybe both the issues are related.

We have used Earlier Astguiclient VERSION 2.4-364a BUILD: 120409-1136 with Asterisk 1.4.39 and everything works properly.
We are now trying the latest Astguiclient available along with asterisk 1.8 on our other server

[tusharcall4]

Hi william conley ;
I am waiting for your reply , i am in hurry because my project Deadline is coming close &i have to test many things.
Please reply on this...

[williamconley]

scratch installs are prone to "bad install" problems. what is the result of:

screen -list

If you do not have at least 7 and more likely 10 screens running ... your install is not done. go back and finish.

It's a good idea to install a "sandbox" server on a virtual machine for comparison. use Vicibox.com's .iso to get a good clean installation so you can compare across systems until everything is running as it should.

And change back to STOCK scratch install until that works. Then consider upgrading asterisk from 1.4 to 1.8.

[tusharcall4]

hey i hv downgrade asterisk to 1.4 from 1.8.NOw its work fine problem is with asterisk 1.8

[williamconley]

or it could be with some slightly old asterisk configuration files. asterisk 1.8 compatibility is fairly new, and changes/bug fixes are being made all the time. many techs who "upgrade" regularly do not allow new asterisk configuration files to be used each time and (especially in the case of an asterisk 1.8 install) really should be.