Contact Queue not routed to an agent

I found something new in CRS CSQ configuration, something interesting a unknown for me. The problem was regarding 2 CSQs – one was with criteria Most Skilled and second with Most Skilled by Weight. Additionally agent had another queues configured and those had been receiving calls with no problem. Those was no routed. Long long discoveries showed that if …

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0x829A – Non-selected user clearing

One another solution probably fast, but remember it doesn’t have to be the perfect for any case. If you see in logs “0x829A – Non-selected user clearing”, at the beginning check if all related dial-peers are OK. Check presence of “direct-inward-dial”, if destination-pattern doesn’t block.

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DSP module test

Found in the net: test voice driver and select 0 (for DSP slot in my case), 2 (DSP) then 16 (Display DSP Keepalive Status) Result as below DSP Keepalive Status Display: ============================= DSP 1 is Alive, State: 4, Keepalive Sent: 130, Skip 131 DSP 2 is Alive, State: 4, Keepalive Sent: 97325, Skip 97329 DSP …

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Another DSP issue and error 0x80AA

I had to solve another problem with switching congestion equipement. Again Cause i = 0x80AA – Switching equipment congestion However dspware version from “show voice dsp detail” was OK I tried: test dsp device 0 all reset but it hasn’t changed my status. Resetting router aka reload and no change at all. What I found …

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Contact Queue not routed to an agent 0x829A – Non-selected user clearing DSP module test Another DSP issue and error 0x80AA

Dec
07

Contact Queue not routed to an agent

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I found something new in CRS CSQ configuration, something interesting a unknown for me.

The problem was regarding 2 CSQs – one was with criteria Most Skilled and second with Most Skilled by Weight. Additionally agent had another queues configured and those had been receiving calls with no problem. Those was no routed. Long long discoveries showed that if there are CSQs with more than one skills and those skills are assigned to others CSQs, an agent will not recieve a call until all that queues will be assigned with minimum competence to agent.

So if there is a CSQ with skills a,b,c,d from queues A,B,C,D and agent has got assigned skills a,b,e,f then an agent will not receive any call from CSQs A,B until it will be assigned with CSQs A,B,C,D. In that case if faced such issue then add agent to proper CSQ or reconfigure CSQ by removing skills c,d. Of course assure, it will not touch call routing for others agents.

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Oct
20

0x829A – Non-selected user clearing

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One another solution probably fast, but remember it doesn’t have to be the perfect for any case.
If you see in logs “0x829A – Non-selected user clearing”, at the beginning check if all related dial-peers are OK. Check presence of “direct-inward-dial”, if destination-pattern doesn’t block.

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Oct
17

DSP module test

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Found in the net:
test voice driver and select 0 (for DSP slot in my case), 2 (DSP) then 16 (Display DSP Keepalive Status)
Result as below
DSP Keepalive Status Display:
=============================
DSP 1 is Alive, State: 4, Keepalive Sent: 130, Skip 131
DSP 2 is Alive, State: 4, Keepalive Sent: 97325, Skip 97329
DSP 3 is Alive, State: 4, Keepalive Sent: 97326, Skip 97328
DSP 4 Not Exist

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Oct
17

Another DSP issue and error 0x80AA

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I had to solve another problem with switching congestion equipement.
Again
Cause i = 0x80AA - Switching equipment congestion
However dspware version from “show voice dsp detail” was OK I tried:
test dsp device 0 all reset but it hasn’t changed my status. Resetting router aka reload and no change at all.
What I found was here:
router# Show voice dsp group all
DSP groups on slot 0:
dsp 1:
State: FAIL, firmware: 23.8.0
Max signal/voice channel: 16/16
Max credits: 240
num_of_sig_chnls_allocated: 8
Transcoding channels allocated: 0
Group: FLEX_GROUP_VOICE, complexity: FLEX
Shared credits: 240, reserved credits: 0
Signaling channels allocated: 8
Voice channels allocated: 0
Credits used: 0

dsp 2:
State: UP, firmware: 23.8.0
Max signal/voice channel: 16/16
Max credits: 240
num_of_sig_chnls_allocated: 0
Transcoding channels allocated: 0
Group: FLEX_GROUP_VOICE, complexity: FLEX
Shared credits: 240, reserved credits: 0
Signaling channels allocated: 0
Voice channels allocated: 0
Credits used: 0

Another time another DSP module to replace.

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Oct
12

How to limit E1 channels

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If there is an idea that you will need guarantize a few channels for incoming calls there is one good method.
1. In global configuration mode create a label
trunk group OUTGOING_CALLS
2. For controller assign trunk group and limit channels. Remember, it depends from provider’s setup of channels selection. Bottom-to-Top or Top-to-Bottom.
controller E1 0/2/0
pri-group timeslots 1-31
trunk-group OUTGOING_CALLS timeslots 1-9
3. Assign trunk group to outgoing dial-peer
dial-peer voice 100 pots
trunkgroup OUTGOING_CALLS

If there is a need to limit an incoming calls, just contact telco provider.

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Oct
07

DSP issue and switching equipment congestion

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A few weeks ago I faced an issue with DSP module. I was looking for a solution but there was a poor description in the network. Even Cisco error decoder hadn’t help at all. I hope this will help someone.
The problem waswith calls establishing. Phones were calling but no voice channel was created. In logs just short info.
Cause i = 0x80AA - Switching equipment congestion
I checked translation-rules, dial-peers etc. but when no changes have been done it wouldn’t be so simple. Some more discoveries and diagnostic as below:
Sh voice dsp detailed
*
*
Warning! DSPs1, in slot 0 are using non-default firmware from flash:
This is not recommended, the default version is 23.8.0

----------------------------FLEX VOICE CARD 0 ------------------------------
*DSP VOICE CHANNELS*

CURR STATE : (busy)inuse (b-out)busy out (bpend)busyout pending
LEGEND : (bad)bad (shut)shutdown (dpend)download pending

DSP DSP DSPWARE CURR BOOT PAK TX/RX
TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT
===== === == ========= ========== ===== ======= === == ========= == ==== ============
C5510 001 01 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 02 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 03 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 04 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 05 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 06 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 07 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 08 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 09 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 10 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 11 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 12 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 13 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 14 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 15 None 0.0.0 bad idle 1 0 0 0/0
C5510 001 16 None 0.0.0 bad idle 1 0 0 0/0
C5510 002 01 None 23.8.0 idle idle 1 0 0 0/0
C5510 002 02 None 23.8.0 idle idle 1 0 0 0/0
C5510 002 03 None 23.8.0 idle idle 1 0 0 0/0
C5510 002 04 None 23.8.0 idle idle 1 0 0 0/0
C5510 002 05 None 23.8.0 idle idle 1 0 0 0/0
C5510 002 06 None 23.8.0 idle idle 1 0 0 0/0
C5510 002 07 None 23.8.0 idle idle 1 0 0 0/0
C5510 002 08 None 23.8.0 idle idle 1 0 0 0/0
Difference between DSP NUM 1 and 2 shows that number 1 was broken. Fast change corrected and issue and all came back to normal work.

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Oct
04

Reverse telnet – AUX to Console

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I needed some access from router to switch without running adn connecting directly to switch. An Idea was to check if it is possible to connect through AUX port on Cisco router to HP proCurve console port.
Well, It is possible and simple, so:
1. Connect Cisco console cable between AUX on router and Console port on switch
2. Configure router AUX port
line aux 0
modem InOut
transport input all
speed 9600 (default or more – the same as for switch)
3. If there is no loopback create one
4. Check communication lines
RTR#sh line
Tty Line Typ Tx/Rx A Modem Roty AccO AccI Uses Noise Overruns Int
0 0 CTY - - - - - 0 0 0/0 -
1 1 AUX 9600/9600 - inout - - - 1 0 29/71480 -
194 194 VTY - - - - - 6 0 0/0 -
* 195 195 VTY - - - - - 3 0 0/0 -
196 196 VTY - - - - - 0 0 0/0 -
197 197 VTY - - - - - 0 0 0/0 -
* marks the line you are connected through. My example is from router 1841
5. Whatever you found lines have different numbering on different router’s series, so just telnet to loopback address and port 2000+ AUX line. In my case 2001.
6. When you has finished, while use keys combination CTRL+SHIFT+6, then press X.
This will disconnect you from the AUX port (and switch in my case)
7. To reconnect to AUX, first it is needed to clear line:
clear line x (in my case x=1)

It is nice I will have access to legacy PBX via IP. Dial-up is history.

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Oct
03

MTU and Maximum Segment Size

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This is just for my own memory.
In the case with MTU going to WAN and other LANs try to configure an interaface with
ip tcp adjust-mss max-segment-size

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Sep
28

E1 leading number and Contact Center

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I found an issue with leading number. When calling E1 leading number ended with a phone call was established with no problem, but in the case when it should go to IVR and forward to agent it was impossible.
Solution was oddly simple – add exact translation rule 1-to-1. Usiing the rule for entire 100 numbers assigned to E1 was usless.

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Sep
03

Palm and webOS security

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As on many smartphones also on webos phones it is possible to get access to e-mail accounts. During first access it is obligatory to setup access pin or password to phone.
Here comes some little danger. 7 tries with wrong pin or password and phone starts to erase its entire memory. Unique issue is that it takes approximately 1 hour.
You remember your palm account credentials of course, so it is fast to turn back everything on phone. I use exchange account to save all business details like contacts, calendar etc. For private staff I use gmail account.
Phone resurecction takes some no short time but I think it is good way secure all fragile content.
Good advice – be carefull what you are doing with phone in your pocket.

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