Overhead Paging with Cisco Berbee InformaCast - Caller ID when paging
Hello, I'm using Cisco Berbee InformaCast with a Bogen overhead paging system. I have Call Manager 6.1.4 and would like to know if there is a way to broadcast the name/number of the IP Phone from which the page originated. If anyone knows of a means of achieving this I would greatly appreciate your assistance.
Thanks,
Dave
Well, not sure why I haven't received any suggestions from anyone. I find it
hard to believe that there isn't anyone else who has the same need. After going from TAC to SingleWire Support here's what I have:
there is a way to have InformaCast display within message text the extension that sent a page. The only catch we were running into with your current implementation was ReliCast. If you're using a ReliCast dial pattern to page, then all InformaCast would display is the ReliCast CTI port that "called" InformaCast to make the page. It was my hope that there was a way to have the source of that trigger displayed in the reporting feature of InformaCast 7. Unfortunately InformaCast would produce the same Information. It would see the calling device as the ReliCast CTI port and log/report that information.
A possible solution to this issue, a panic button. You could setup one of the line appearance buttons on a users phone as an emergency live all page. Then, we can just add a bit of "script" to it that would display the time, day, and source device that made the page.
For example, you can create a new page type using a combination of Text and Live Audio. In the Short Text section of the message you could place the following:
"Panic Button Triggered at ${time} on ${date} by ext ${senderInfo.directoryNumbers}"
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Show Run
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.02.27 15:14:52 =~=~=~=~=~=~=~=~=~=~=~=
sh run
Building configuration...
Current configuration : 12139 bytes
! Last configuration change at 06:35:59 UTC Tue Feb 25 2014
! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname eucamvgw01
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.151-4.M5.bin
boot-end-marker
card type e1 0 0
logging buffered 51200 warnings
no logging console
no aaa new-model
no network-clock-participate wic 0
no ipv6 cef
ip source-route
ip traffic-export profile cuecapture mode capture
bidirectional
ip cef
ip multicast-routing
ip domain name drreddys.eu
ip name-server 10.197.20.1
ip name-server 10.197.20.2
multilink bundle-name authenticated
stcapp ccm-group 2
stcapp
stcapp feature access-code
stcapp feature speed-dial
stcapp supplementary-services
port 0/1/0
fallback-dn 5428025
port 0/1/1
fallback-dn 5428008
port 0/1/2
fallback-dn 5421462
port 0/1/3
fallback-dn 5421463
isdn switch-type primary-net5
crypto pki token default removal timeout 0
voice-card 0
dsp services dspfarm
voice call send-alert
voice call disc-pi-off
voice call convert-discpi-to-prog
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 10.198.0.0 255.255.255.0
ipv4 152.63.1.0 255.255.255.0
address-hiding
allow-connections sip to sip
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
fax-relay ans-disable
sip
rel1xx supported "track"
privacy pstn
no update-callerid
early-offer forced
call-route p-called-party-id
voice class uri 100 sip
host 41.206.187.71
voice class codec 10
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 ilbc
codec preference 4 g729r8
codec preference 5 g729br8
voice class codec 20
codec preference 1 g729br8
codec preference 2 g729r8
voice moh-group 1
moh flash:moh/Panjo.alaw.wav
description MOH G711 alaw
multicast moh 239.1.1.2 port 16384 route 10.198.2.9
voice translation-rule 1
rule 1 /^012237280\(..\)/ /54280\1/
rule 2 /^012236514\(..\)/ /54214\1/
rule 3 /^01223651081/ /5428010/
rule 4 /^01223506701/ /5428010/
voice translation-rule 2
rule 1 /^00\(.+\)/ /+\1/
rule 2 /^0\(.+\)/ /+44\1/
rule 3 /^\([0-9].+\)/ /+\1/
voice translation-rule 3
rule 1 /^9\(.+\)/ /\1/
rule 2 /^\+44\(.+\)/ /0\1/
rule 3 /^\+\(.+\)/ /00\1/
voice translation-rule 4
rule 1 /^54280\(..\)/ /12237280\1/
rule 2 /^54214\(..\)/ /12236514\1/
rule 3 /^\+44\(.+\)/ /\1/
rule 4 /^.54280\(..\)/ /12237280\1/
rule 5 /^.54214\(..\)/ /12236514\1/
voice translation-rule 9
rule 1 /^\(....\)/ /542\1/
voice translation-rule 10
voice translation-rule 11
rule 1 /^\+44122372\(....\)/ /542\1/
rule 2 /^\+44122365\(....\)/ /542\1/
voice translation-rule 12
voice translation-rule 13
rule 1 /^\([18]...\)/ /542\1/
voice translation-rule 14
voice translation-profile MPLS-incoming
translate calling 10
translate called 9
voice translation-profile MPLS-outgoing
translate calling 11
translate called 12
voice translation-profile PSTN-incoming
translate calling 2
translate called 1
voice translation-profile PSTN-outgoing
translate calling 4
translate called 3
voice translation-profile SRST-incoming
translate calling 14
translate called 13
license udi pid CISCO2921/K9 sn FGL145110RE
hw-module ism 0
hw-module pvdm 0/0
username administrator privilege 15 secret 5 $1$syu5$DsxdOgfS7Wltx78o4PV.60
redundancy
controller E1 0/0/0
ip tcp path-mtu-discovery
ip scp server enable
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description internal LAN
ip address 10.198.2.9 255.255.255.0
duplex auto
speed auto
interface ISM0/0
ip unnumbered GigabitEthernet0/0
service-module ip address 10.198.2.8 255.255.255.0
!Application: CUE Running on ISM
service-module ip default-gateway 10.198.2.9
interface GigabitEthernet0/1
description to TATA NGN
ip address 115.114.225.122 255.255.255.252
duplex auto
speed auto
interface GigabitEthernet0/2
description SIP Trunks external
ip address 79.121.254.83 255.255.255.248
ip access-group SIP-InBound in
ip traffic-export apply cuecapture size 8000000
duplex auto
speed auto
interface ISM0/1
description Internal switch interface connected to Internal Service Module
no ip address
shutdown
interface Vlan1
no ip address
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 10.198.2.1
ip route 10.198.2.8 255.255.255.255 ISM0/0
ip route 41.206.187.0 255.255.255.0 115.114.225.121
ip route 77.37.25.46 255.255.255.255 79.121.254.81
ip route 83.245.6.81 255.255.255.255 79.121.254.81
ip route 83.245.6.82 255.255.255.255 79.121.254.81
ip route 95.223.1.107 255.255.255.255 79.121.254.81
ip route 192.54.47.0 255.255.255.0 79.121.254.81
ip access-list extended SIP-InBound
permit ip host 77.37.25.46 any
permit ip host 83.245.6.81 any
permit ip host 83.245.6.82 any
permit ip 192.54.47.0 0.0.0.255 any
permit icmp any any
permit ip host 95.223.1.107 any
deny ip any any log
control-plane
voice-port 0/1/0
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/1
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/2
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/3
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
no ccm-manager fax protocol cisco
ccm-manager music-on-hold bind GigabitEthernet0/0
ccm-manager config server 152.63.1.19 152.63.1.100 172.27.210.5
ccm-manager sccp local GigabitEthernet0/0
ccm-manager sccp
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 10.198.2.9 identifier 3 priority 3 version 7.0
sccp ccm 152.63.1.19 identifier 4 version 7.0
sccp ccm 152.63.1.100 identifier 5 version 7.0
sccp ccm 172.27.210.5 identifier 6 version 7.0
sccp
sccp ccm group 2
bind interface GigabitEthernet0/0
associate ccm 4 priority 1
associate ccm 5 priority 2
associate ccm 6 priority 3
associate ccm 3 priority 4
associate profile 1002 register CFB_UK_CAM_02
associate profile 1001 register XCODE_UK_CAM_02
associate profile 1000 register MTP_UK_CAM_02
dspfarm profile 1001 transcode
codec ilbc
codec g722-64
codec g729br8
codec g729r8
codec gsmamr-nb
codec pass-through
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 18
associate application SCCP
dspfarm profile 1002 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dspfarm profile 1000 mtp
codec g711alaw
maximum sessions software 200
associate application SCCP
dial-peer cor custom
name SRSTMode
dial-peer cor list SRST
member SRSTMode
dial-peer voice 100 voip
description *** Inbound CUCM ***
translation-profile incoming PSTN-incoming
incoming called-number .
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 500 voip
description *** Inbound TATA MPLS ***
translation-profile incoming MPLS-incoming
session protocol sipv2
session target sip-server
incoming called-number ....
incoming uri from 100
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 510 voip
description *** Outbound TATA MPLS ***
translation-profile outgoing MPLS-outgoing
destination-pattern 54[013-9]....
session protocol sipv2
session target ipv4:41.206.187.71
session transport udp
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 520 voip
description *** Outbound TATA MPLS ***
translation-profile outgoing MPLS-outgoing
destination-pattern 5[0-35-9].....
session protocol sipv2
session target ipv4:41.206.187.71
session transport udp
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 200 voip
description *** Inbound M12 *** 01223651081, 01223651440 - 01223651489
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 0122365....
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 201 voip
description *** Inbound M12 *** 012237280XX
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 012237280..
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 202 voip
description *** Inbound M12 *** 01223506701
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 01223506701
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 210 voip
description *** Outbound M12 ***
translation-profile outgoing PSTN-outgoing
destination-pattern +...T
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 211 voip
description *** Outbound ISDN for SRST and emergency ***
translation-profile outgoing PSTN-outgoing
destination-pattern 9.T
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 212 voip
description *** Outbound ISDN for emergency ***
translation-profile outgoing PSTN-outgoing
destination-pattern 11[02]
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 2000 voip
description *** Outbound to CUCM Primary ***
preference 1
destination-pattern 542....
session protocol sipv2
session target ipv4:152.63.1.19
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 2001 voip
description *** Outbound to CUCM Secondary ***
preference 2
destination-pattern 542....
session protocol sipv2
session target ipv4:152.63.1.100
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 2002 voip
description *** Outbound to CUCM Teritiary ***
preference 3
destination-pattern 542....
session protocol sipv2
session target ipv4:172.27.210.5
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 999010 pots
service stcapp
port 0/1/0
dial-peer voice 999011 pots
service stcapp
port 0/1/1
dial-peer voice 999012 pots
service stcapp
port 0/1/2
dial-peer voice 999013 pots
service stcapp
port 0/1/3
sip-ua
no remote-party-id
gatekeeper
shutdown
call-manager-fallback
secondary-dialtone 9
max-conferences 4 gain -6
transfer-system full-consult
ip source-address 10.198.2.9 port 2000
max-ephones 110
max-dn 400 dual-line no-reg
translation-profile incoming SRST-incoming
moh flash:/moh/Panjo.ulaw.wav
multicast moh 239.1.1.1 port 16384 route 10.198.2.9
time-zone 22
time-format 24
date-format dd-mm-yy
line con 0
login local
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 131
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
session-timeout 60
exec-timeout 60 0
privilege level 15
login local
transport input all
line vty 5 15
session-timeout 60
exec-timeout 60 0
privilege level 15
login local
transport input all
scheduler allocate 20000 1000
ntp server 10.1.30.1
end
eucamvgw01#
Sh SCCP
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.03.03 17:57:44 =~=~=~=~=~=~=~=~=~=~=~=
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
IPv4 Address: 10.198.2.9
Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.198.2.9, Port Number: 2000
Priority: 3, Version: 7.0, Identifier: 3
Call Manager: 152.63.1.19, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 4
Trustpoint: N/A
Call Manager: 152.63.1.100, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 5
Trustpoint: N/A
Call Manager: 172.27.210.5, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 6
Trustpoint: N/A
MTP Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1000
Reported Max Streams: 400, Reported Max OOS Streams: 0
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1001
Reported Max Streams: 36, Reported Max OOS Streams: 0
Supported Codec: ilbc, Maximum Packetization Period: 120
Supported Codec: g722r64, Maximum Packetization Period: 30
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: gsmamr-nb, Maximum Packetization Period: 60
Supported Codec: pass-thru, Maximum Packetization Period: N/A
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
Conferencing Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1002
Reported Max Streams: 16, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070080
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070081
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070082
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070083
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
eucamvgw01# -
Ask the Expert : Call Recording with Cisco Unified Communication Manager (UCM)
Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to learn and ask questions about Cisco Unified CM call recording solution that provides the ability to record customer conversations for compliance purpose. This topic will cover an overview, configuration and troubleshooting of the call recording feature.
Monday, January 19th, 2015 to Friday, January 30th, 2015
Harmit Singh is a technical leader with the High Touch Technical Services (HTTS) and Technical Assistance Center (TAC) Unified Communications teams based in Bangalore. He has broad experience in Cisco Unified Communications infrastructure solutions. He has 10 years of experience working with large enterprise and service provider networks. He also holds CCIE certifications (#20012) in Voice and Collaboration as well as Red Hat and VMware certifications.
Mohammed Noorulla Khan is a customer support engineer in High-Touch Technical Services (HTTS) Unified Communications teams based in Bangalore. His areas of expertise include Cisco Unified Communications Manager, Gateways, and Jabber. He has over 6 years of industry experience working with large enterprise and service provider networks. He also holds CCIE certifications (#35741) in Voice and VMware certifications.
** Remember to use the rating system to let Mohammed and Harmit know if you've received an adequate response. **
Because of the volume expected during this event, the experts might not be able to answer every question. Remember that you can continue the conversation in the Collaboration, Voice and Video community, subcommunity, IP Telephony, shortly after the event. This event lasts through January 30th 2015. Visit this forum often to view responses to your questions and those of other Cisco Support Community members.Hi Maheshwar,
Thank you for your query. Please find my response below:
1> Do we support recording with HCS environment and which 3rd party vendors are validated with HCS based call control 10.1.1?
Answer: Whether you use a standalone UCM cluster, UCCE or HCS, call recording would be supported across the board in the same manner.
Please refer to the following link:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/hcs-cc/10_0_1/Install_and_Config/CHCS_BK_ICC270D0_00_installing-and-configuring-cisco-hcs/CHCS_BK_ICC270D0_00_installing-and-configuring-cisco-hcs_chapter_011.html#CHCS_RF_T1105284_00
Option
Notes
Recording
All Recording applications that are supported by Unified CCE are supported on HCS for CC. For details, see Recording section in Agent and Supervisor Capabilities.
With respect to which 3rd party vendors have been validated, marketplace.cisco.com is a good place to crosscheck this info. You will find a Cisco Compatible Logo against the partners listed there. The logo is used to signify that the PARTNER product has undergone technical interoperability testing with the Cisco product specified. The interoperability testing is conducted by a third party laboratory based on testing criteria set forth by Cisco. PARTNER is solely responsible for the support and warranty of its product. Placement of the PARTNER product or information pertaining thereto, on the Cisco Marketplace website does not constitute an offer to sell the PARTNER product in any way. For further information on the PARTNER products, please visit the PARTNER company website.
Please refer to the following link and use the search field under Collaboration Technology:
https://marketplace.cisco.com/catalog/search?utf8=%E2%9C%93&search[q]=&search[technology_category_ids]=23%2C24%2C197%2C1940%2C1941%2C1921%2C1576%2C1897%2C1983%2C2418%2C26%2C198%2C1904&search[order]=tier&per_page=20&_=1421663854257&ts=1421663855441
2> Which end points are supported for recording via HCS call control?
Answer: The following link should help clarify this:
http://solutionpartner.cisco.com/web/sip/wiki/-/wiki/Main/Unified+CM+Silent+Monitoring+Recording+Supported+Device+Matrix
Please let us know if you have any follow up questions. Hope this is helpful.
Regards,
Harmit Singh. -
Is it possible to integrate Nokia E62 with Cisco Call Manager 4.x?
The Nokia E-61 can be integrated with Cisco Call manager 4.x using a Nokia Call Connect 1.0 client.
1.Is it possible to do the same with Nokia E62.If it can be done,what would be the requirements?
2.Is Nokia E-62 PDA a Cisco Compatible Extension device?Thanks a lot for the prompt reply.You are right in saying that nokia and cisco are increasing the number of models supported.Infact the E65 has already joined the club.But my point of interest is the nokia E62 in particular.This Nokia E62 device
1) does not support WLAN interface(while E61,E65 does).
2) It is not Cisco Commpatible extension device.(I did not find E62 in the list of Cisco Compatible extensions.chk out the attachments for more info).
3)It is not compatible with Nokia Intellisync CAll Connect 1.0(while E61 and E65 are compatible.chk out the attachment for more info).
I found this rather weird and i wondered if Nokia E62 can be integrated with Cisco Call manager ,especially a device which is not Cisco compatible.The only possibility i can see from your answer is that the Nokia E62 might not be SIP enabled(correct me if i am wrong here!)
My objective is to send a message from a CUAE script to a nokia E62 device.Please excuse me if my posting is not in the right place.i would be highly grateful to you if you could give me pointers to an appropiate location where i can get my queries answered. -
Any one using/deployed Telematrix phones with Cisco Call Manager
Any one using/deployed Telematrix phones with Cisco Call Manager
Yes, it does:
Cisco Unified IP Phone Series 7900
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/compat/ccmcompmatr.html#wp51474
Config on CUCM itself is no different from any other phones neither other requirements like DLUs.
Of course you should already have APs for connectivity.
HTH
java
if this helps, please rate -
Integration with Cisco Call Manager?
We have Cisco VOIP phones and we would like to continue managing calls with Cisco Call Manager. We currently have voicemail on Cisco Unity. We currently have Exchange 2010, but plan to start using Exchange 2013 soon.
However, we would like to migrate voicemail service from Cisco Unity to Exchange Unified Messaging and also want to take advantage of most of the voicemail-related features of UM such as voice to text sent to email, voice mail waiting indicator light on
the phones and sending voice mail audio attachments to email.
Is Exchange 2013 Unified Messaging compatible Cisco phones that are managed by Cisco Call Manager?
We have 500 users and 2 Exchange 2010 servers (1 CAS and 1 Mailbox Server). We now have a few Exchange 2013 licenses available in addition to our existing 2010 licenses that are already used on our existing 2 servers. Can we keep the 2 existing
Exchange 2010 servers as they are and create a new Unified Messaging server on Exchange 2013 in the same environment or do we need to upgrade all the 2010 servers to 2013 if we use Unified Messaging on 2013?Hi Sachin,
We do run a hybrid Callmanager to Nortel Meridian setup (works well).
Here are some great docs;
Cisco Unified CallManager
Case Study: Nortel 61C PBX to Cisco IP Telephony Migration
From this good doc;
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_white_paper09186a00801115e0.shtml
Nortel Meridian PBX and Cisco CallManager Integration
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a008011888c.shtml
Cisco Unified CallManager System Guide, Release 4.2(1)
Cisco DPA Integration
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a008055cd53.html
Hope this helps! Let me know if you need additional info.
Rob -
Cisco Unified WFO - Call Recording and Quality Management with Extension Mobility agents
Hi All,
We're considering Cisco Unified WFO - Call Recording and Quality Management for a customer running UCCX 8.0, agents on multiple WAN sites, all agents using extension mobility.
The documentation I've been able to find describes three different recording methods:
Using Desktop Recording service (Endpoint) to record from an agent’s desktop.
Server Recording - Uses SPAN (not so good for remote sites)
Network Recording - Uses CUCM recording service / SIP trunk / phone's built in bridge.
Network recording or Desktop recording should be suitable for the customer but it seems that Extension mobility is not supported. Extension Mobility is not mentioned in the 8.5 installation guide, it is mentioned as ‘not supported’ in the 8.0 guide as follows:
'Server Recording and Network Recording have the following limitations:
• Extension mobility is not supported.'
Neither version's documentation specifically mention extension mobility in relation to the desktop recording method, though I realise this is a similar approach to the 'server recording' method.
So the question I have is: Is extension mobility supported in any way on version 8.0, or version 8.5 for recording? And if so which recording method(s) are supported?
Thanks,
JonathanHi,
I had more luck asking questions over at the Calabrio forum - they make the software and Cisco re-brand a version of it - there is some good info on their portal (http://portal.calabrio.com), you have to register but it's fairly painless. The answer I got was:
"QM Desktop recording has always supported extention mobility as it determines the recorded user by the desktop user's login. Extention mobility was not supported for Server and Network recording until the Calabrio QM 8.6.2 release in April 2011 and will be added to Cisco QM starting with QM 8.5.2 in June 2011"
Regards,
Jonathan -
I'm having an issue whenever one of our employee's conferences in an external translator, as soon as they bridge the customer into the call with the translator QM stops recording. I can hear the intial conversation with the customer, and then the employee put the customer on hold and call the translator service. Only when the two are bridged together the call always stops recording.
We are performing all recording on the server side, and are not using Quality Manager Desktop agents.
Any help would be much appreciated. Thanks!
-ChrisSo I found the following information listed below. I don't manage the Cisco Unified CM portion of our telco system. Can we limited the ourselves to a single Codec, and would this even resolve the issue. Does this cause other issues if we didn't limit the devices that are recording to a single codec?
Recording IssuesThis topic explains how to diagnose and resolve problems that occur with contact
recordings.
Calls for devices configured for Network Recording drop when you try to conference
or transfer a call.
Symptom. When you try to transfer or conference a call and one or more devices on
the call configured for network recording, the transfer or conference fails and
parties drop off of the call.
Cause. Cisco Unified CM does not support codec changes for devices that are
configured for call recording. The codec must remain the same throughout the life
of the call. For conference calls, the conference bridge must support the codec
used before the conference completes.
Solution. Update the Cisco Unified CM configuration to ensure that no codec
changes occur for network recorded devices. -
Click to Call with Cisco WIM 4.4?
Hi eGain Experts,
Could anybody confirm that Click to Call with Cisco WIM 4.4 will work with UCCE? I understand Web Callback will work, however not too sure about Click to call. I know eGain CallTrack or ClickToCall can achieve this. Is anybody able to confirm if this feature is configurable with the Cisco WIM licensing arrangements? if so could somebody provide sample configuration on how to achieve would be very useful.
Thank you very much.
Regards,
YavuzShort answer no. All evidence points to UCCX only.
Sent from Cisco Technical Support iPad App -
ATA-186 with overhead paging systems
Since the ATA-186 supplies 600 Ohms, would I need a seperate device such as a Bogen TAM-B to supplie the audio for overhead paging, or could I just use the ATA-186?
ATA-186 can be used as an audio source.
-
Message from Berbee Informacast
Hi,
We are currently using Cisco Call Manager 5.1.2 and Berbee Informacast 5.1.1 [server:Windows 2003 R2 SP2]system. Once in a while all our IP phones display following broadcast message " Telephone Service Working " [ this message is on Informacast Administration Message page]. What is causing this broadcast? How can I prevent this broadcast? Do I need to take any further action to resolve this issue? Please let me know.
Thanks in advance.
Alex ThazhayilInformacast has been successfully tested with thousands of phones. The actual messages are broadcast via multicast while phone control is sent via unicast. The multicast message is made available on a stream on the network and must remain available until the last phone on the network has received the message. In an installation of thousands of phones the stream must remain available on the network until the last phone is activated via unicast phone control. This means that the first phones that are activated are forced to get the message for the entire time the stream is available, thus you get some phones that repeat the message.
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