little issues

[xmerlin]

hi there

we have just gone live with our first real vicidial server in a running call center (after a few test machines) witha 10 agents and i am happy to report that for the main everyting went swimmingly.

I have given all the agents cheap kubuntu boxes that have been locked down a fair amount and this is all working just beautifully. The feedback from the agents (normally a disparaging rabble) is really positive, so muchos kudos to you Matt.

I really cannot thank the posters on this board and Matt in particular enough for helping me / us get to this point!

Anyway, enough gushing. I did come here to ask for help once again, so cap in hand, here goes:


For some reason, every now and again, an agent gets through to an answer machine (im no tsure if this is every time it hits answerphone) and all the agent can hear is static, this doesnt go away until the agent logs out and back in again. I have not had a chance to look into this issue properly yet as it has been rather a full day but i will look into it in the morning, in the meantime, if you've got any thoughts on what could be causing it im sure it would be helpful. I am not using answer machine detection as laid out in the manager manual...

the only other thing that strikes me is that there is no way of easily seeing what calls have been marked as a sale made. i will knock up a little php page that the call center manager can use in the meantime, but it would be cool if i've simply overlooked something and this functionalty is already within vicidial.

cheers


ps. Happy New Year

[enjay]

You should be able to go into Agent Performance Detail under reports and see what calls were dispositioned as what. There is a SALE default out of the box that is included with this though you can customize it according to your environment needs.

I hope this helps.

-Art

[xmerlin]

thanks, but whilst that page is handy its just a case of filtering out only the ones marked as a sale and giving the detail on those leads so that the manager can print them off periodically and pass tehm on to sales people...

not to worry, ill knock something up

the answerphone thing is a bit of a concern though...

[mflorell]

For your agent audio issue, what kind of phones are you using?

What shows up in the "calls in this session" link at the bottom of vicidial.php when this happens?

What kind of zaptel timer are you using?

As for the record detail display(displaying details of SALEs for instance) this is something I am working on for the next release.

[xmerlin]

Hi Matt,

Sorry i couldnt reply earlier, but its been a little insane recently and the audio issue hasnt been a huge problem thus far because by logging out and in again the crackle clears. but obviously it is somethign that could do with fixing.

one thing that has changed is that it is not now just affecting calls to answerphones, but other calls as well :/

the softphone being used is Ekiga, i went for this because it is one of the few that i found would run well on kubuntu and would start up in autoanswer and minimised, meaning no interaction needed from the agent.

the zaptel timer i am useing is the T110P timing from teh network. Having said that, i have just modprobed ztdummy and that hasn't seemed to make any difference.

the info in the in session calls looks to be correct, there is no discernable difference between that info on a crackling call and a normal call. If you can let me know more specifically what it is you're looking for and i'll see if i can find something.

the only error i am getting out of the CLI is :

Jan 5 12:47:36 WARNING[2479]: channel.c:781 channel_find_locked: Avoided deadlock for '0x9de9fd0', 10 retries!

and this does not neccessarily pop up during a crackling call.

I made myself a little page that displays the sale information and that is doing the trick for now

thanks for the help, once again.

[mflorell]

What is the loadavg of your server when this happens?

Are you using any VOIP connections with any codecs other than ULAW?

Are you doing any VICIDIAL recording of calls?

[xmerlin]

Hi Matt,

Below is the output of sip show peer of teh actual agent with teh problem and below that teh CLI output of one agent with crackling call, although im not sure this is of any help.

top shows very little system resources being used.

as you can see below the softphones are using both a-law and u-law, with the latter being the preference, however, what is confuddling me is that the initial conference is fine and teh agent can make and receive calls for a while until the crackle starts, which to me does not sound like a codec mismatch problem...

the testing we did was a little convoluted, but i will try and lay it out below:

agent 1 makes call (manually, but also happens in auto dialling mode)
crackle starts and caller and called party can hear crackle. (ie affects whole conference)
callee hangs up and there is still crackle on the line
agent 2 dials into the same conference and can hear crackle.
agent 1 hang up the call and the conference goes clear.

Having done this testing it does start to look more and more like a problem with the softphone randomly going mental. Does this sound right to you?

The next thing i am looking to do is to get a couple of Aastra hardphones up to them for testing to see if we can replicate the problem with these, although i will be very surprised if we get the same problem.

Does anyone have any good remcommendations for softphones running on kubuntu?




* Name : 205
Secret : <Set>
MD5Secret : <Not set>
Context : default
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : Yes
Callerid : "" <>
Expire : 2566
Insecure : no
Nat : RFC3581
ACL : No
CanReinvite : Yes
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID : No
DTMFmode : inband
LastMsg : 0
ToHost :
Addr->IP : 192.168.0.155 Port 5098
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 205
SIP Options : (none)
Codecs : 0xc (ulaw|alaw)
Codec Order : (ulaw,alaw)
Status : OK (4 ms)
Useragent : Ekiga/2.0.1
Reg. Contact : sip:205@192.168.0.155:5098;transport=udp

== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600061@default-af07,2", "8600061") in new stack
> Channel Local/8600061@default-af07,1 was answered.
-- Executing AGI("Local/8600061@default-af07,1", "call_log.agi|901753482965") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("Local/8600061@default-af07,1", "zap/g1/01753482965|55|rtTo") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/01753482965
== Manager 'sendcron' logged off from 127.0.0.1
-- Zap/1-1 is proceeding passing it to Local/8600061@default-af07,1
-- Zap/1-1 is ringing
-- Zap/1-1 answered Local/8600061@default-af07,1
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8600061, 1) exited non-zero on 'Local/8600061@default-af07,2'
-- Executing DeadAGI("Local/8600061@default-af07,2", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("Local/8600061@default-af07,2", "VD_hangup.agi|PRI-----NODEBUG-----0---------------") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
-- Hungup 'Zap/1-1'
== Spawn extension (default, 901753482965, 2) exited non-zero on 'Local/8600061@default-af07,1'
-- Executing DeadAGI("Local/8600061@default-af07,1", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("Local/8600061@default-af07,1", "VD_hangup.agi|PRI-----NODEBUG-----16-----ANSWER-----37-----27") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Refreshing DNS lookups.
== 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
== Manager 'sendcron' logged off from 127.0.0.1
localhost*CLI>

[mflorell]

I would suggest trying another softphone just to see if the audio gets better.

Also try softphones on other machines.

[xmerlin]

yeah, when i got a couple of aastra hardware phones sent up there it was impossible to recreate the fault. so i recon it was purely down to the softphone. so warning to people:

Ekiga on Kubuntu may cause problems!

anyone got any recommendations for softphones on nix?

Thanks