Agent waiting issue

I am also facing the CALL WAITING ISSUE. The DROPPED PERCENT IS ONLY 2.22 and in the 8 hours of calling an agent will only get 150 calls to speak. I did not find any solution for this and since we were facing some other problem my we cleared the caches and junks in the asterisk and then it started working good.After doing this the DROPPED PERCENT came to 16 and each agent got 400 calls to speak… But after few day again the same problem exists…Can any one find a solution for this…We are not able to identify where the error would be…Please guide us …

thank u

What is the “call waiting issue”? You write as though everyone knows of it, but this is the first time I remember that name being used.

Also, are you sure that your problem lies with Asterisk and not with gotoautodial.

If it is a known Asterisk bug, the, the bug tracker entry will tell you how the fix is progressing. If not, you will generally need to provide fairly detailed logging and details of your configuration (in Asterisk terms, not in terms of any front end that you are using).

David thanks for the reply…I just want to tell you that I am new to this goautodial and asterisk.

AGENT WAITING ISSUE : This issue is nothing but, each and every agents wait for a long duration to get the next call even when the ratio is fixed high. After disposing a call the agent wait for 1 minute to 6 or 7 minutes to get the another call, due to which the ideal time of the agent increases and end of the day the agents end up speaking with less number of customers.
I am not sure whether the problem is with GOAUTODIAL OR ASTERISK… Please help me out to sort out the problem.
I am using GOAUTODIAL 3.0 & ASTERISK 1.4 …
I have pasted the error and warning message…

 -- Executing [8368@default:1] Playback("Local/99841486680@default-b2fa,1", "sip-silence") in new stack

[Aug 26 22:47:04] – <Local/99841486680@default-b2fa,1> Playing ‘sip-silence’ (language ‘en’)
[color=#FF0000][Aug 26 22:47:04] WARNING[20538]: file.c:1297 waitstream_core: Unexpected control subclass ‘-1’[/color]
[Aug 26 22:47:04] – Executing [h@default:1] DeadAGI(“Local/99841486680@default-b2fa,2”, “agi://–HVcauses–PRI-----NODEBUG-----16-----ANSWER-----32-----0”) in new stack
[Aug 26 22:47:04] – Executing [8368@default:2] AGI(“DAHDI/10-1”, “agi://”) in new stack
[Aug 26 22:47:04] – AGI Script agi:// completed, returning 0
[Aug 26 22:47:04] – Executing [8368@default:3] AGI(“DAHDI/10-1”, “agi-VDAD_ALL_outbound.agi|NORMAL-----LB”) in new stack
[Aug 26 22:47:04] – Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[color=#FF0000][Aug 26 22:47:04] ERROR[20538]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe[/color]
[Aug 26 22:47:04] – AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Aug 26 22:47:04] – Executing [1921680012008600052@default:1] Goto(“DAHDI/10-1”, “default|8600052|1”) in new stack
[Aug 26 22:47:04] – Goto (default,8600052,1)

What do you mean by “ratio”.

The warning is unusual, but seems to relate to a custom sound file, probably part of gotoautodial. I’d initially ignore that.

The error is going to be a problem with the AGI script, which is part of gotoautodial. It means the script didn’t terminate cleanly. A lot of AGI scripts don’t terminate cleanly, but don’t cause problems.

You seem to be using DAHDI. If you have analogue line, unless you have suitable disconnect supervision, the call may not clear until the agent clears it. If this is a an outbound centre, and I don’t really like helping those, most PSTN providers will not clear the call just because the callee puts the phone on hook, as they assume that the callee may pick the phone up on a different parallel wired extension. They will wait for a few minutes before doing a network initiated release of the call.

I think you need to treat this as a gotoautodial problem, untless and until the gotoautodial people can identify an Asterisk problem, and describe it in Asterisk terms.