Inbound Calls-Non-DID
I only have a single DID number for my Lync environment. I have configured the extensions for all users in for format of tel:did;ext=xxxx I also have my automated attendant as an extension as well. so to be clear no user has the DID number.
All numbers have an extension.
I need to have the main DID to go to the AA. I read that I needed to create a pool scope dial plan. I have done that but when I try to dial my DID it never connects.
Here is my nomination rule for my dial plan
^(1xxx7xx0394)$ --> +1xxx72xx0394;ext=1000
translation
+$1
I have not seen a clear example of how to get a single DID to work with Lync using only extensions with Exchange UM
You don't need to create a pool dialplan for this. If you've already got a global or site-level dial plan it will work. Your normalization rule isn't all that clear, but your rule should look like this:
Pattern: ^1xxx7xx0394$
Translation: +1xx7xx0394;ext=1000
This assumes the incoming number comes in EXACTLY as shown. If the incoming number already has a plus sign in front, then you need to get rid of it, because Lync won't normalize any number that has a plus sign already there.
Once you have your normalization rule setup, you need to use the OcsUMUtil.exe program (found in %CommonProgramFiles%\Microsoft Lync Server 2013\Support) to create the AA object for Lync. Make sure the TelURI is set to +1xx7xx0394;ext=1000.
If you've done all this right, it should work like a charm.
Ken Lasko | Lync MVP | http://UCKen.blogspot.com |
http://LyncOptimizer.com
Similar Messages
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SIP Trunk not accepting inbound calls
I have a CME setup using Engin as a SIP provider
I am able to dial out with no issue, however my inbound calls do not work, they divert to the Engin voicemail
My SIP registration is OK and the number is configured as the primary DN on one of my phones
Router#sh sip-ua register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
038682XXXX -1 1124 yes
101 20001 45 no
102 20003 18 no
103 20005 45 no
104 20006 45 no
I do see the call come in if I debug the dial peer, but it only seems to match an outgoing dp
I am seeing a couple of disconnect cause codes that I cant seem to find any relavent information on in the CCSIP debugs
Router#
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:4F947560
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:1800, SE Refresher:uas, Min-SE Value:1800, flags:2001
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:4F947DF8
Sep 8 18:15:15.025: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:403, container:4F947B38
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4C39C570
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0417XXXXXX
Called Number : 038682XXXX
Source IP Address (Sig ): 211.30.48.136
Destn SIP Req Addr:Port : 203.161.164.69:5060
Destn SIP Resp Addr:Port : 203.161.164.69:5060
Destination Name : 203.161.164.69
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 211.30.48.136
Source IP Port (Media): 17768
Destn IP Address (Media): 203.161.164.69
Destn IP Port (Media): 18314
Orig Destn IP Address:Port (Media): [ - ]:0
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 21
Disconnect Cause (SIP) : 403
Any Ideas
DougHi Tapan,
Firstly the topology is as follows
ISP/VOIP provider - Internet - Cable modem - 2800 CME router - IP Phone
The VM is provided by the ISP
debug ccsip messages
Sep 9 10:21:41.488: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Contact:
Supported: 100rel,timer
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: multipart/mixed,application/media_control+xml,application/sdp
Min-SE: 60
Session-Expires: 1800;refresher=uas
Max-Forwards: 9
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 317
v=0
o=BroadWorks 18275729 1 IN IP4 203.161.164.69
s=-
c=IN IP4 203.161.164.69
t=0 0
m=audio 18128 RTP/AVP 18 8 0 101
c=IN IP4 203.161.164.69
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=bsoft: 1 image udptl t38
Sep 9 10:21:41.508: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Sep 9 10:21:41.516: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=21
Content-Length: 0
Sep 9 10:21:41.544: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
CSeq: 633854439 ACK
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
Max-Forwards: 9
Content-Length: 0
Voice Config
Router#
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server expires max 3600 min 3600
localhost dns:mel.byo.engin.com.au
no call service stop
voice class codec 1
codec preference 1 g711ulaw
voice translation-rule 10
rule 1 /^0/ //
voice translation-rule 11
rule 1 /^.*/ /0386821234/
voice translation-profile PSTN_Outgoing
translate calling 11
voice-card 0
dsp services dspfarm
mgcp profile default
sccp local Vlan100
sccp ccm 10.1.100.1 identifier 1 version 7.0
sccp
sccp ccm group 1
bind interface Vlan100
associate ccm 1 priority 1
associate profile 1 register confdsp
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 4
associate application SCCP
dial-peer voice 99 voip
translation-profile outgoing PSTN_Outgoing
destination-pattern .T
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 100 voip
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number 0386821234
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 110 voip
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number .%
dtmf-relay rtp-nte
no vad
dial-peer voice 90 voip
description Melbourne 03 Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [89].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 91 voip
description National Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 0[278]........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 92 voip
description Vic/Tas 03 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [56].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 93 voip
description Mobile numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 04........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 94 voip
description 13XXXX numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 13[1-9]...
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 96 voip
description 1300/1800 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 1[38]00......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 98 voip
description Emergency 000
translation-profile outgoing PSTN_Outgoing
destination-pattern 000
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
sip-ua
credentials username 0386821234 password 7 XXXX realm voice.mibroadband.com.au
authentication username 0386821234 password 7 XXXX
nat symmetric role active
nat symmetric check-media-src
no remote-party-id
retry invite 2
retry register 10
timers connect 100
mwi-server dns:mel.byo.engin.com.au expires 3600 port 5060 transport udp unsolicited
registrar dns:mel.byo.engin.com.au expires 3600
sip-server dns:mel.byo.engin.com.au
connection-reuse
telephony-service
sdspfarm conference mute-on #1 mute-off #2
sdspfarm units 2
sdspfarm tag 1 confdsp
conference hardware
max-ephones 42
max-dn 144
ip source-address 10.1.100.1 port 2000
calling-number initiator
service phone videoCapability 1
service phone displayOnDuration 00:01
service phone displayOnTime 08:30
service phone displayOffTime 17:30
service phone displayIdleTimeout 00:01
service phone displayOnWhenIncomingCall 1
system message Cisco CME
load 7941 SCCP41.8-4-2S
load 7942 SCCP42.8-4-2S
load 7945 SCCP45.8-4-2S
load 7961 SCCP41.8-4-2S
load 7962 SCCP42.8-4-2S
load 7965 SCCP45.8-4-2S
load ata ATA030204SCCP090202A
time-zone 48
date-format dd-mm-yy
voicemail 90125200
mwi relay
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
moh music-on-hold.au
web admin system name cisco secret 5 $1$d8/H$glhLiCCWXmFSUp6BtwGho0
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 0.T
create cnf-files version-stamp 7960 Jul 06 2011 10:32:45
ephone-dn 1 dual-line
number 038682XXXX
label 101
name 7965
mwi sip
ephone-dn 2 dual-line
number 102
label 102
name 7941
ephone-dn 3 dual-line
number 103
label 103
name 7920
ephone 1
device-security-mode none
video
mac-address 0023.5EB8.6E4E
type 7965
button 1:2 2:1
ephone 3
device-security-mode none
mac-address 0019.0633.A933
max-calls-per-button 2
type 7920
button 1:3
ephone 10
device-security-mode none
mac-address 0019.E7B7.BAB3
max-calls-per-button 2
type ata
button 1:1 -
The inbound calls to our call center is drop after putting it on hold or transfer
Dear All;
Good day
The inbound calls to our call center is drop after putting it on hold or transfer the call to another agent. The MOH file is playing till 21 sec only then call drop . the agent cant resume the call again. The MOH file is running from Gateway (multicast).
No problem in outbound calls.
I urgent need you help
Should you require any more information , please do not hesitate to contact me.
Thanks & Best Regards,
Muhammad Fathy,
IT Network Manager
ALEXBANK
A subsidiary of Intesa Sanpaolo Group
Head office: B210-F1, Smart Village , KM 28 Cairo-Alex Desert Road, Egypt.
Cell: +201017288844.
Office: +202-35311300 Ext: 8090.
eMail: [email protected]
i To maintain a paperless environment, please don't print this e-mail unless you really need to.Typically you have a codec or media resource issue to track down. IE, MTP, region, location, gateway trunk to trunk to call or something in that area. Bypass UCCX and do the same call without this app... does it happen with a normal call?
-
How do I resolve the issue for answering inbound calls on my iPhone 5 when connected to Bluetooth in a holden commodore VE - 2012 model. Outbound calls connect ok.
I'm having the same problem with a 2013 GMC terrain. I don't know if the problem is the car or the iPhone 5. I have been seeing this problem more and more on threads. Can anyone help us with this? Is there any info online from Apple re: new updates that would solve this problem?
-
Debugging inbound call - need debug command
I am debugging what I believe to be a dial plan issue on PGW but want to run a debug command on the ITP to show the actual called number of the inbound call.
What is the best debug command to use to see the inbound called and calling numbers of a call.You are better off capturing an mdl trace from the PGW using per call tracing or using snoop and opening the call in Wireshark.
The ITP will only dump hex values and you will need to decode them yourself since it is not concerned with ISUP layer. You can use the cs7 paklog feature to attach an access list to a linkset and then send the raw data to a syslog server for decoding.
For example:
cs7 paklog x.x.x.x dest-port xxxx src-port yyyy
access-list 2700 instance x permit si all
debug cs7 mtp3 paklog 2700 -
Problem with OSB 11g - Unable to call non-static java methods
I have a problem in OSB. Unable to see any java methods when loading java callout. I have checked the java classes in the .jar file and they are all non-static java methods.
Is there any way where OSB is able to see this non-static java methods?
Need help urgently!
Thanks!Technical standpoint: Do you know why OSB is not able to call non-static methods?
This is by design. Ability to call non-static methods require Object creation which adds additional complexity. eg How to pass variables in constructor?. How/where to store created object for use across across pipeline instance?. Object life cycle (when and how to create object) etc. To avoid above complexites static methods are only supported.
"a lot of non-static method to call" just for my understanding what are the number involved?. If number is too high you can always request for a enhancement.
Thanks
Manoj -
Is it possible to call non-registered contacts via...
I am interested to use skype-to-go for business callls but like to know if you can also call non-registered contacts within skype-to-go? So my questions is: Could I simply call my skype-to-go number from my mobile/landline and then subsequently call any number/person (besides/outside my few registered contacts - I guess they only allow 10 different registered contacts)?
Otherwise I have to change my registered contacts all the time which is time consuming....and I also don't have internet at hand all the time.
cheers, Pieter
EDIT : title/message case changedpieterbedaux wrote:
1. Could I simply call my skype-to-go number from my mobile/landline and then subsequently call any number/person
2. - I guess they only allow 10 different registered contacts)?
Hello Pieter,
1. Yes - you can call worldwide - not just your registered contacts.
2. We have a maximum of 9 registered contacts.
http://www.skype.com/intl/en-us/features/allfeatures/skype-to-go-number/
TIME ZONE - US EASTERN. LOCATION - PHILADELPHIA, PA, USA.
I recommend that you always run the latest Skype version: Windows & Mac
If my advice helped to fix your issue please mark it as a solution to help others.
Please note that I generally don't respond to unsolicited Private Messages. Thank you. -
My Ipad will show all regular inbound calls, from Iphone, as Facetime calls. I would like only TRUE facetime calls to appear. I have switched audio/video on Ipad but it still happens. How can I set the Ipad to only register only true FT calls rather than all inbound calls that a re not facetime calls? Yes, you may have to reread this twice and slowly to understand..... Thanks!
Hi webn,
I understand that you do not wish to see phone calls on your iPad anymore. I have an article for you with some information on accomplishing this goal:
Connect your iPhone, iPad, iPod touch, and Mac using Continuity - Apple Support
https://support.apple.com/en-us/HT6337
Turn off iPhone cellular calls
To turn off iPhone cellular calls on your iPad or iPod touch, go to Settings > FaceTime and turn off iPhone Cellular Calls.
On your Mac, open the FaceTime app and go to FaceTime > Preferences. Click Settings and deselect the iPhone Cellular Calls option.
Thanks for coming to the Apple Support Communities!
Cheers,
Braden -
Inbound calls to response group - failed
Hi!
I have a problem with inbound calls to Response group from external phones. But calls from Lync client to Response Group are working without problem. And calls from external phones to Lync users perfectly working too.
I found this alert in event viewer
Event 31172 LS response group service
The workflow runtime encountered an error while connecting the call.
The workflow runtime encountered a critical error.
Failure occurrences: 2, since 27/06/2013 9:20:18 PM.
The last encountered error was from a workflow having the display name: Group2, the URI: sip:[email protected], and
the GUID: 12ee27bf-15ab-4cc9-8265-57a08f17ac5b.
Exception: Microsoft.Rtc.Signaling.OperationTimeoutException - This operation has timed out.
Inner Exception: -
I hope for your help!!! Thanks!!Hi,
Here is a similar case for your reference. Try to remove NAT and have the SIP line connect direct to Lync Server.
http://social.technet.microsoft.com/Forums/lync/en-US/d0e53ab2-42e7-4d79-be01-b8770d508133/calls-to-workflows-fail
Kent Huang
TechNet Community Support -
Blocking inbound call in CM with MGCP Voice gateways
I'm sure someone has posted on this: We have MGCP voicegateways and CM 4.02aSR2b. How can I setup CM to block a specific inbound call? If the gateways were H323 then no problem, but with MGCP we are stumped. Any suggestions? Thx!
It cannot be done with MGCP. It can be done with H.323 using reject translation rules on the GW.
With MGCP your only option is to route all calls via Unity or IPCC first where logic can be build to block calls.
HTH, please rate posts!
Chris -
Inbound calls getting answered automatically in Cisco CME
Hi All....
Please advise me the reason, why the inbound calls to CME got answered automatically.?
I am mentioning the call handling scenario here with..
Trunk Type : FXO
Connection PLAR to physical extension(200)
timeouts call-disconnect 1
timeouts wait-release 1
voice-port 0/0/0
trunk-group FXO
cptone AE
timeouts call-disconnect 1
timeouts wait-release 1
connection plar 200
caller-id enable
More over I tried with the below configuration also!!!!!!!!!!!! Bad luck !!
voice class custom-cptone UAE-TONE
dualtone disconnect
frequency 400
cadence 400 350 225 525
voice-port 0/1/0
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-TONE
timeouts call-disconnect 1
timeouts wait-release 1
dial-peer voice 202 pots
incoming called-number .
direct-inward-dialThe command connection plar 200 under voice port 0/0/0 directs the H323 gateway to automatically answer the inbound call on that FXO port and to attempt to transfer the call to DN 200.
If that is not what you want, then you need to remove the connection plar command and replace it with the command "secondary dialtone" -
HI.
I'm running CME on a 2851.
My issue is with inbound calls not failing totally, but then requiring the extension to be dialed. This only happens for calls where the calling number is unknown.
So all calls inbound work if the inbound calling number is known. Calls route nicely.
If the calling number is unknown the caller gets a tone, whereby they need to then dial the extension.
Ideally I'd like to have unknown calling numbers simply route the call the same way as with a known calling number.
FYI when i debug q931 caller id (if off) is shown as Calling Party Number i = 0x00A3, N/A
Appreciate any help as this is causing some problems!!!
Configuration attachedTimothy,
When the calls come in, a dial-peer match is made, and in this case, where the calling number is known, the dial-peer whose destination-pattern command matches the calling number is selected, and the call is routed.
For calls where the calling number is not available, there is no dial-peer matched, so the default dial-peer (dial-peer 0) is used, and two stage dialing is then required.
To resolve, a quick solution would be to create a new dial-peer that has the incoming called-number command (which matches on the called number), so that this dial-peer will always be used for all incoming calls regardless of whether the calling number is available or not.
Use the direct-inward-dial command as well in this dial-peer to force one stage dialing.
dial-peer voice 90 pots
incoming called-number .
direct-inward-dial
port 0/0/0:15.
See also
* http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#topic3
* http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800e00d0.shtml
Hope this helps.
Regards,
Michael. -
Hi,
I have a AS5400HPX device with a AS5400 T1 2 PRI DFC card installed and T1 link. I need to check the inbound calls on the T1 PRI link.
1) What commands can help me check verify if the call lands on the router, and what is the source number which is dialling?
2) Do all the Async interfaces on the device represent each of the client dailling in? How do I check which Async interface has been used by the client?
3) I will be using a local pool for assigning IP addresses to each of the client dialling in.Would this command suffice?
ip local pool <pool name> IP range
4) Also, what are these resource-pool group and resource-pool profile commands used for?
Appreciate your inputs.
Thanks
MikeyAppreciate if someone could reply to this.
Cheers
Mikey -
Calling non-static from servlet
Why the he** I cannot call non-static functions from other class.. I define my servlet like public class xxx extends HttpServlet and try to call "library" functions from normal class. Static is bad.. It messes up with my servlets when there are many users at the same time.
You can't call non-static functions (without an instance of the class) because the functions don't exist (without an instance of the class). Just because the library is sitting somewhere out there doesn't mean your program knows about it.
Static may mess-up with multiple users because static means only one exists. If each user is modifying it, it ain't gonna return the same thing to each user each time he accesses it. -
IPCC Enterprise: Outbound calls priority over inbound calls
How can I get outbound calls presented to the agents in SG-A even though there are calls queued to SG-A ?
My customer wish to service callbacks before queued calls.
To me it looks like the dialer doesn't issue the reservation call quick enough to get the agent reserved before the queueing mechanism is grabbing him.
Thanks
/ClausThe reason inbound calls take priority is because they are queued.
With personal callbacks, we try to reserve the agent even if it is currently busy so we have an opportunity to queue them.
With preview and predictive, we check the skill group once every 2 seconds and only attempt to reserve agents to see if there are available agents. So we won't attempt to reserve an agent if agents are busy with inbound calls. So if you have inbound calls queued up for a skill group that share the same agent pool as the outbound skill group, then those agents will likely be kept too busy with inbound calls. The Dialer won't try to make a reservation request if no agents are available and so there won't be an opportunity to add the calls to queue.
With Transfer to IVR campaigns, you can queue and route customer calls to agents. This would give you more flexibility, but there are some pros and cons.
PROS
You can make smart decisions in the routing script whether it is marked as Answering Machine or Live Voice.
You don't tie up an agent in reservation call waiting to find a customer.
You can prioritize the dialer calls over inbound calls in the scripting.
CONS
Customers will likely spend some time in queue, and may drop out.
The scripting to throttle calls is a little trickier.
The campaign reports and Dialer Detail records no longer tell you whether the call was handled by an agent or not. They stop tracking once the call is sent to the IVR. You only have the inbound call Termination Call Detail records and the inbound skill group and call type to track what happened ot the transfer to IVR calls.
David
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