VICIDIAL versions and Asterisk 1.8

[bghayad]

Hello;

Can I know which versions of vicidial 2.8 that can work with asterisk 1.8?

I have two machines, one is working with asterisk 1.8 and the other has a problem that the call stay in the queue and does not reach to the agent although the agent is logged in.

The specification of the one that is working (the call reach to the telephone of the agent):

VERSION: 2.8-403a
BUILD: 130510-1350

The one that is not working (the call stay in the queue and the call does not reach to the telephone of the agent although the agent see that there is a call in the queue and the agent is logged in):

Version: 2.8-404a
BUILD: 130605-0841

To get out from these troubles (some version is work and some does not work), what installation I have to use? I need asterisk 1.8 and I need vicidial 2.8?

Regards
Bilal

[williamconley]

any later version should work as well as or better than the earlier version. it is likely you are experiencing a non-1.8 issue or are set up slightly differently and possibly exposing a bug not bumped into with the other configuration.

also: what installer did you use on these two installations? what version of asterisk (full version description, not just major revision)? and why are you not using "stock" installation with 1.4 asterisk?

[bghayad]

any later version should work as well as or better than the earlier version. it is likely you are experiencing a non-1.8 issue or are set up slightly differently and possibly exposing a bug not bumped into with the other configuration.

* Which logs (do I need to enable something) that will help me to discover the problem?

also: what installer did you use on these two installations? what version of asterisk (full version description, not just major revision)? and why are you not using "stock" installation with 1.4 asterisk?

* I used iso vicibox 4.0.3 standard and Asterisk 1.8.21.0-vici-beta. By the way, how I get two different builds while I used the same CD for the installation (vicibox 4.0.3)?

At the real time report, I can see that the agent is logged in to the queue and I can see that there is a call in the queue and the agent is ready (not paused).

In other words, the call is staying wait in the queue and not routed to the agent ! Where I can see the logs that can tell me what is happening?

The only thing that I did, I installed a lot of dependent libraries (even some maybe is not required) to be able to compile most of the asterisk channels (like gtalk, jingle, jaber, ... etc), so if this caused a conflict that caused an error, where I can see this error?

Regards
Bilal

[williamconley]

Inbound or outbound? Cuz inbound calls are not only related to Campaign and if the agent is not logged into the Ingroup, or the Campaign does not allow the Ingroup ... well, perhaps sharing the method of calling would be helpful. I'm not prepared to assume outbound autodial. LOL

When you install, depending on your choices, you can technically get any version of Vicidial using the Vicibox .iso. The OS and packages will all be the same, but the Vicidial revision level will be "latest" at the moment of install or a defined version (such as what was on the .iso or a specified revision level).

I note that you carefully did not specify the reason for asterisk 1.8 ... perhaps you could drop that while you're at it. LOL (ie: if you're getting Free Skype calls don't bother hiding it, we don't care ... but information is useful in troubleshooting).

If you use the "screen -r asterisk" screen, you can often see an error in AGI scripting. But other than that, the basic Asterisk CLI (asterisk -R) and the vicidial (/var/log/astguiclient) and asterisk (var/log/asterisk) logs are where to look.

[bghayad]

In the inbound.
I used asterisk 1.8.... Because I can have better commands in the consol (I can un-register a peer and some other commands), also it contains channel for SMS (the jingle channel). Really I don't have any calls to Skybe and did not do this, although I prefer to have gtalk or Jaber channels for future usage.

At the asterisk consol, I did not see any errors but I did not check yet the logs as I am outside.
Regards
Bilal

[williamconley]

You may do better to run the asterisk 1.8 in a separate server and switch back to a Stock installation of Vicidial with 1.4 from Vicibox. Then you'd know the problem (if there was one) was not related to Asterisk at all.

Nothing stopping you from having two servers running. Asterisk 1.8 will run in a virtual box quite nicely, so it does not even need to be a "real" computer for that portion of your requirement.

[bghayad]

I discover one point that might help:

If the Phone is not On-Hook (in other words, On-Hook = N), then it receives the calls. If the Phone is On-Hook = Y, then the problem happens (I see in there queue there is a call, and I see in the dashboard that the call sent for the Phone, but the call does not reach the Phone and nothing is appearing at the Asterisk CLI regarding this (except playing the waiting music but nothing is related to routed the call !!).


What could be the reason for this?
Regards
Bilal

[williamconley]

check the status of the peer. is it reachable?

sip show peers

[bghayad]

Ofcourse it is reachable, I can dial to and from it.

I am going to do fresh installation ... It seems that the problem was related to downloading un necessary libraries.

Or u have any idea that could help?
By the way, at the dash board (real time report), it appears that the call has been sent to the agent !! But the phone is not ringing !

I would if there is any method that I can use to troubleshoot and knows where is the problem.

Regards
Bilal

[williamconley]

did you have the agent log out and back in after you changed to on-hook?

[bghayad]

Of course I did this.

[williamconley]

have you always had this version of vicidial on the server, or has it been upgraded at some point? if so, did you upgrade the sample conf files during upgrade or say "No"?

if you have not upgraded, consider upgrading now (perhaps just install fresh on a VMware box) to test this function in the latest code. If it works in a fresh install with latest code ... upgrading may fix your problem.

[bghayad]

I did a fresh installation for vicibox 4.0.3 (the BUILD is: 130605-0841 for the fresh installation) and asterisk 1.8.21.0-vici-beta and again the same problem.

At the asterisk console, I do not see any attempt to send the call to the agent phone extension which is 501. Only I see the commands related to the music on hold for waiting (although the agent is ready to take the call).

This time, I captured logs from /var/log/astguiclient/agiout.2013-06-29 as it might help:

Code: Select all

2013-06-29 02:26:12|02:26:22|agi-VDAD_ALL_inbound.agi|RING-AGENT CALL SENT: |RINGAGENT00000000013|8331*13*Y6290226120000000760*agent002*501|0|15   SIP/505-0000000d|Y6290226120000000760
2013-06-29 02:26:12|02:26:22|agi-VDAD_ALL_inbound.agi|--    VDAD get agent: |360|0|10|20|0|0||0|update of vla table: 6abeebyInGroup|
|INSERT INTO vicidial_manager values('','','2013-06-29 02:26:12','NEW','N','192.168.10.2','','Originate','RINGAGENT00000000013','Exten: 8331*13*Y6290226120000000760*agent002*501','Context: default','Channel: Local/192*168*010*002*501@default','Priority: 1','Callerid: RINGAGENT00000000013','Timeout: 14000','','','','')|
2013-06-29 02:26:12|02:26:22|agi-VDAD_ALL_inbound.agi|NNNNN No available balance agent found
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|0|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and campaign_id = '6abeebyInGroup' and call_time < "2013-06-29 02:26:12" and lead_id != '760' and queue_priority >= '0' and agent_only='';|
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '760' and agent_only='' and ( (queue_priority > '0') or (queue_priority = '0' and call_time < "2013-06-29 02:26:12") );|
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|RING-AGENT TIMER:   14.75|15|15   SIP/505-0000000d|Y6290226120000000760
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|--    VDAD get agent: |360|0.25|11|20|0|0||0|update of vla table: 6abeebyInGroup|
|UNLOCK TABLES;|
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|NNNNN No available balance agent found
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|0|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and campaign_id = '6abeebyInGroup' and call_time < "2013-06-29 02:26:12" and lead_id != '760' and queue_priority >= '0' and agent_only='';|
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '760' and agent_only='' and ( (queue_priority > '0') or (queue_priority = '0' and call_time < "2013-06-29 02:26:12") );|
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|RING-AGENT TIMER:   14.5|15|15   SIP/505-0000000d|Y6290226120000000760
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|--    VDAD get agent: |360|0.5|12|20|0|0||0|update of vla table: 6abeebyInGroup|
|UNLOCK TABLES;|
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|NNNNN No available balance agent found
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|0|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and campaign_id = '6abeebyInGroup' and call_time < "2013-06-29 02:26:12" and lead_id != '760' and queue_priority >= '0' and agent_only='';|
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '760' and agent_only='' and ( (queue_priority > '0') or (queue_priority = '0' and call_time < "2013-06-29 02:26:12") );|
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|RING-AGENT TIMER:   14.25|15|15   SIP/505-0000000d|Y6290226120000000760
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|--    VDAD get agent: |360|0.75|13|20|0|0||0|update of vla table: 6abeebyInGroup|
|UNLOCK TABLES;|
2013-06-29 02:26:12|02:26:23|agi-VDAD_ALL_inbound.agi|NNNNN No available balance agent found

From the other side, I did another test:

I set the DID to send for the Phone 501 instead of going to the queue, the call reached at the Phone !! Actually the Phone is reachable from asterisk but it seems it is something related between the queue and the routing to the agent or the Phone of the agent.

Again, I am able to see at the agent desktop that there is a call in the queue, but the call is not coming to the agent even if the agent is ready.

In the above logs, the 505 is the extension that I placed the call from it (I was testing from a local extension .. actually I did a tests from sip trunk and from dahdi .. the same problem).

If the above logs are not enough, I hope if someone advise me for a specific log or specific thing that help me to determine the problem.

Is it a bug ?

I am going now to do a fresh installation, but this time with asterisk 1.4 (I will not use asterisk 1.8 version) .

I would like to say: I just did fresh installation for the same vicibox 4.0.3 and from the same media and using the asterisk 1.4 (which come with it), every thing is working FINE and I do not have a problem.

By the way, in the installation that used asterisk 1.8, I was compiling asterisk 1.8 before doing the vicibox-install ... But one time, it worked with me good, but I do not remember if I did the compilation and installation for asterisk 1.8 before the vicibox-install or after. Does this effect?

But, how I can determine exactly from where is the problem and if it is bug in asterisk-1.8.21.0-vici-beta or not?

Regards
Bilal

[williamconley]

try a very controlled asterisk cli output from an asterisk 1.8 "Fail" and an asterisk 1.4 "Success" and compare them side by side.

It is likely that there is a command difference between 1.4 & 1.8 that is causing the call to fail that was not anticipated in the asterisk 1.8 upgraded vicidial code.

It's likely very simple, could be a comma instead of a | needs to happen or something very arcane, but the only way to find it is to find the point of failure and then decipher that line.

It may also be helpful to compare to that successful call to the agent when the did goes direct. Perhaps a dial command has been "not updated to 1.8 requirements" somewhere.

You're doin' a good job of troubleshooting. At this level of effort you will be very likely to actually find it. Be sure to post it when you do ... to the Issue Tracker and link to it here so we can get it fixed in the next SVN update. (Assuming it's a bug, of course)

[bghayad]

But in the asterisk CLI, it does not show any dial command to the phone.

By the way: when there will be a vicibox installing asterisk 1.8?
Regards
Bilal

[williamconley]

check your /var/log/astguiclient logs

[bghayad]

A lot of logs files there, which files to look for it in a certain way?

Is it the agiout or congest or screenlog or vdautodial or vdautodial_FILL or action_full? Or which one?

Regards
Bilal

[williamconley]

delete them all. make your call. read what "pops up" and decide which are relevant