Flashbelt Conference in Midwest

Robert Reinhardt. Danny Patterson. Moses Gunesch. Paul
Ortchanian. Sam Pastel. Chris Allen.
What do they have in common? They all be giving sessions at
the Flashbelt conference in Minneapolis this June along with 30
others great Flash and Flex related sessions.
The 5th Annual Flashbelt Conference is 3 weeks away. Adobe is
the Platinum Sponsor. It's the place to see rocks-star designers,
whiz-kid developers and best-selling authors all come together and
talk about Flash and related topics. The speaker line up is amazing
and the size (limited to 400 attendees) is sure to make for an
incredible conference experience.
http://www.flashbelt.com

Hi Tom,
There are quite a few good Flash conferences out there, but
not many that are just 100% training.
I produce the flashbelt conference, so I'm biased and think
it's a great event.
www.flashbelt.com
We have content along 3 tracks, with one focused on design
and concepts, one focused on development, and one that catches the
rest of the content like business, art, inspiration and other
software tools. About 75% of the sessions are design to teach
attendees skills on the spot. Plus we have 3 day-long workshops set
up this year. Learning ActionScript 3, Video for Flash, and
Papervision 3D.
Other conferences with good track records and content are
Flash on the Beach, and FITC.
For more involved general training you might need to look at
companies rather than conferences. www.easeltraining.com is the
best place I can think of here in the midwest.

Similar Messages

Actionscript 3 Workshop

Join Rich Shupe in Minneapolis for the day-long workshop
"Learning Actionscript 3". This one-day workshop will help get you
started using AS3 whether you have experience with AS1/AS2 or are
just getting started.
The 5th Annual Flashbelt Conference is taking place in
Minneapolis this June 8-11. Adobe is the Platinum Sponsor. It's the
place to see rocks-star designers, whiz-kid developers and
best-selling authors all come together and talk about Flash and
related topics. The speaker line up is amazing and the size
(limited to 400 attendees) is sure to make for an incredible
conference experience.
http://www.flashbelt.com

Might be an idea to post this in the Flex2 or AS3 forums
where people who
are interested in these can read it.
"JonJonMenendez" <[email protected]> wrote
in message
news:e3ouru$fm1$[email protected]..
> It's a late notice, but important to any Flash
professional out there. I
> just
> want everyone to know about this rare opportunity. Fly
out to LA if you
> have to!
> Flex Builder 2 & Action Script 3 Training Workshop
> Date: Thu, Fri, & Sat, May 11,12, & 13 (9a-5p,
~20 classroom hours)
> Location: Venice, California 90291
> Instructor: Zach Stepek
> Price: $690
> Description: Flex expert and Stateline (Illinois)
Macromedia User Group
> Manager Zach Stepek will lead an in-depth three-day
workshop on the new
> generation of Flex -- the first tool for the Flash
Platform that utilizes
> the
> awesome processing power of AS3! --
>

Did you know that the carrier Verizon offer the merge conference call or internet while on call on the iphone? Only AT

Did you know that you can't use the merge function on a iPhone 4s with Verizon Wireless? They blocked the function only AT&T has it, also Verizon doesn't allow browsing the Internet while on a call like AT&T, huge mistake Verizon. For people that conference call, don't get tricked into changing carriers until Verizon changes this feature.

Not sure about the merging of calls, but being able to browse the internet while on a call can't be done with CDMA technology -- so it's not that Verizon doesn't allow it, it just can't be done on their network. This is old news ...

Calling issue with Cisco 7937 conference station

Hi Friends,
I am facing issue wiht Cisco 7937 conference station, our customer have various branch offices accross the world. All branches are connected over MPLS through service provider( SIP service provider) . there is a centralized CUCM and remote office have SIP Voice gateways .
When making calls from once remote site to another using Cisco 6921 phones calls working fine
When making calls from once remote site to another using Cisco 7937 conference station to make call any phone at remote office, calls are getting disconneted, remote phone rings when calls, but its gets fast busy tone when other party picks up the phone and not able to talk.
I suspect the issue with Codec but we have configured transcoders in VG and registered with CUCM
Please help me if any one experience such issue earlier.
Regards
Siva

hi Basant,
1. Actually tow phones A and B are registerd with centralized CUCM, A and B are located in two different locations, RTP traffic between And B pass through service provider.
Call Flow --> Phone A ---->CUCMRouterpattern--> SIP trunk ----> Voice gateway--->Service provider cloud---> Respective Voice Gateway---> CUCM -- Phone B
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#

Why my iPhone wont let me add to conference call or hang up on a particular person?

Why my iPhone wont let me add to conference call or hang up on a particular person?

Hey D.j.,
Welcome to the Apple Support Communities. The following article will go over the Conference features and will provide some help.
iPhone: Using Conference
http://support.apple.com/kb/ta38608
If you continue to have issues with the Conference feature, the next step would be to reset settings or backup and restore the device and setup as new. To reset settings on the device: Settings > General > Reset All Settings
If you need to restore your iPhone, the following article will assist with this.
iOS: How to back up your data and set up as a new device
http://support.apple.com/kb/HT4137
Regards,
-Norm G.

HI I have an Iphone 3GS and the microphone jack works when I play music but not when making phone calls. please help as I use it for conference calls

Hi I use my phone for conference calls and use the microphone jack to connect to speakers. It doesnt work when making phone calls but when i play music on the phone the jack works with the external speaker.

Happy to have been of help. Hope all goes smoothly.
BTW, for the sake of others who may help you here, it's usually most polite if you mark the post that answered your question as being the one that "solves" it, not your own post. That's how people get points here, if a questioner marks their post as either "helpful" or "solving". It's not an issue for me, but for those just starting out in the forums it may be important to them. See:
https://discussions.apple.com/static/apple/tutorial/mark.html
and
https://discussions.apple.com/static/apple/tutorial/reputation.html
Regards.

To implement video conference in my website

Hi,
I want to implement an interactive Video conference in my
website.
I am Beginner in PHP, MYSQL and FLASH.
It would be great if any one could give me answers and
solutions for below questions:
1. Any list of webhosting sites which can support video
conferencing?
2. If there are seperate server sites which only handle with
video conferencing and i need to integrate with my seperate
webhosting provider where my website is getting hosted. Then how
can i do it? what would be the costs that that would i have to
invest for supoose there are 1 presenter and 100 viewers.
3. Alternatively, I would like to get help from anyone in
doing below steps.
a) Add a flash player in my webpage, so that it can record
the video,audio stream from a webcam, microphone.
b) send to webhosting or videoconferencing supporting
server.
c) publish the live video,audio stream to the people who are
requesting to see it live.
Ex: I m presenting a topic in front of a webcam and n number
of users in my site can attend that topic live.
currently i m taking n to may be 50 or so.
Immediate response is appreciated, as i need to implement it
as soon as possible .. i have a very strict deadline for the
same... anyone can help me asap will be of great thanks..
Thanks in advance....

Hi,
Here are some quick answers:
1.
http://www.adobe.com/products/flashmediaserver/fvss/
2. Your website and video player can be hosted by someone
different than the FMS host. Your FMS host will give you the
information to use in the video player (the server name/application
name, etc).
3.a There are lots of articles and tutorials to help you get
started. Here's one:
http://www.adobe.com/devnet/flashmediaserver/articles/beginner_vod_fm3.html
And check out the tutorials right here at fmsguru.com
b. See #2.
c. see 3a.
HTH,
Jody

I am trying to listen to a webcast from a conference. I am told I need a plug in for Firefox. I can listen on Explorer nine without a problem. What plugin do I need for 3.6?

I just purchased a new HP computer with windows 7. I installed 3.6. which I use as a default and had no trouble until I tried to listen to a webcast from the BMO metals and mining conference. When I clicked on listen to webcast I got a message tha tadditional plug ins were needed to display all the media on the page. I checked my system and was able to hear the test message, but I could not hear the actual webcast.
There was a box in the message saying click here to fins needed plug ins. When I clicked on it I got a screen saying no plug ins found. I have windows media player installed. There was a space to click on to test sound system, and I heard to message on the test, but when I tried to hear the actual webcast I got the same plug in needed message.
I then tried to listen on Chrome and got the same result. I tried to listen on Explorer and was able to hear the webcast and got no message about plug ins needed.
Is this just a case where they won't accommodate firefox, or is there a plug in or add on that I can install so I can use firefox.

Start Firefox in [[Safe Mode]] to check if one of the add-ons is causing the problem.
* Don't make any changes on the Safe mode start window.
See:
* [[Troubleshooting extensions and themes]]
It is hard to tell which plugin is needed to play that webcast. You appear to have Flash and WMP and Silverlight. Other popular but less used plugins are RealPlayer and QuickTime.
See:
* [[Popular plugins]]
* http://kb.mozillazine.org/Testing_plugins
You can check the page source to see which MIME type is specified for that content or post a link if that web page can be accessed publicly without authorization.

I am planning to buy an iMac 21.5 inch with Fusion drive and 16 GB RAM while i visit USA for a conference. My country of origin is India. Once i am back to India, will my iMac be covered by the Apple's Protection Plan?

I am planning to buy an iMac 21.5 inch with Fusion drive and 16 GB RAM while i visit USA for a conference. My country of origin is India and the option of configuring to my need is not avaialble here in India. My concern is once i am back to India, will my iMac be covered by the Apple's Protection Plan if something goes wrong here in India?
I recently came across some posts regarding International warranty related issues for iPhone and iPad....hence this crossed my mind...
Waiting for response from the support group members...

Thanks Ralph
This essentially means that if my iMac develops some problem (both software and hardware) in my country (India), it will be completeley attended (even the whole system requires replacement) by the AASP in India...is that what you say?

How many users in a group video conference on iChat

What experience have people had with a number of participants on an iChat video conference. What's an optimal number? What is the maximum number (if the bandwidth supports it)?
Thanks,
Czet

http://support.apple.com/kb/HT2020?viewlocale=en_US

Problems with conference calls(adding and merging calls) after upgrade to ios 8.1.2 on iphone 5s

I am having weird problem intermittently with adding calls or merging them or swapping them. The issue happens 8 out of 10 times whenever I try to do a conference call. This issue came up after I upgraded to 8,1.2. Never experienced this in 8.1.1 or previous version.
Some of the scenarios that I have experienced is:
1. While on the call, when I try to add a new contact, it just doesn't dial the second number.
2. Sometimes the second number gets dialed, cannot merge the calls.
3. After talking to the second person, cannot go back to the first call.
I am very sure that this is not a carrier issue as it worked flawlessly before upgrade.
Anyone experienced this before? Any suggestions?

Thanks for the above infromation. I did a complete restore of my iphone through iTunes connecting it to my computer via cable. iOS8.1.3 was re-installed. I then added back my settings and data which were backed up prior to restoring the phone. I still have the same issue with conferencing and adding a third caller. When I hit the [+] box, the existing caller is not put on hold. The phone appears to be making a call but it is not dialing to the 3rd party. Shortly thereafter, the Merge icon appears but there is no active call to merge with. I have tried this several times using my wife's iPhone and our home phone to set up a 3-way call. I know it is not working because the 3rd phone (our home phone) does not ring. Is this a hardware problem? My wife's iPhone 6 works with no problem and was purchased the same time I purchased my iPhone 6. Probably time for a visit to the Apple Store unless there is another solution.
iPhone 6; iOS 8.1.3; Carrier: Verizon

How do I set up a conference call that people can call into via phone?

Hi, I need to do the following things:
find some kind of provider that will let me register a toll free number to use with Connect
set up this provider in my Connect account
make a meeting where people can call the toll free number to join the conference
Could someone walk me through this step by step, like you would to a 2nd grader who was trying to set all of this up for the first time? Thank you.

1. There are more providers of teleconferencing than is reasonable to list here. You can do some searching on the Internet and find some or you can start with those who fully integrate with Connect. Those that fully integrate with Connect are MeetingOne, Intercall, Arkadin and Premiere Global. You can use any SIP enable provider though (I'd be shocked if you could find one that isn't SIP to be honest).
2. If dealing with one of the providers that fully integrate, they should be able to walk you through the process, it's pretty simple. If dealing with another provider, follow these instructions: https://www.connectusers.com/tutorials/2012/10/universal_voice_setup/index.php
3. Browse to the desired folder in the Meeting library of Connect. Click the New Meeting button and go through the wizard for creating a new room. At the end of step 1 of the wizard you will have 3 audio options. Choose the middle option and select the desired phone bridge that is associated with your account. The meeting should then prompt you to connect to the audio bridge every time you start a meeting.

One DSP for Conference and Voice ?

Dear all,
I have one DSP (PVDM2-16) in the R2801 "12.3.11T". Can I configure HW conference and voice port (VIC2-4FXO) at the same time.
Indeed, I cannot configure as follows:
"maximum session X" under the conference profile. I only found X is "0". Can I allocate the DSP into different purposes?
Thanks for your help

Another problem with this config is that the PVDM2-16 is only 1 dsp, and conferencing requires a reserved dsp:
PVDM2-16 16-Channel Packet Fax/Voice DSP Module
1 DSP (TI 2510)
Q. Can conferencing share the resources with transcoding or voice calls of a single DSP?
A. No. Conferencing needs a dedicated DSP resource. If a DSP is assigned for a conferencing session, then it cannot be used for transcoding or voice call initiations or terminations. Transcoding and voice calls can share the resource of a single DSP, however. Note that conferencing needs a dedicated DSP, but not a dedicated PVDM2 module. For example, PVDM2-64 contains 4 DSPs; if one of them is used for conferencing, the other three can still be used for other purposes.
http://www.cisco.com/en/US/products/hw/modules/ps3115/products_qanda_item0900aecd8016c6ad.shtml
Mary Beth

How Do I Find Where My Beehive Conference Is?

Hello,
I am using Beehive Conferencing, version 2.0.1.7.0, I just recorded my first conference with a customer.
But now, I cannot locate where that conference is stored.
I have checked under C:\Documents and Settings\auahmed2\Application Date\Oracle\Beehive, there are no asp files here.
I use MS Outlook, it is installed with Outlook Extensions for Beehive.
I looked under the folder Documents, and there are no files here either.
Your help is appreciated.
Anees.

Hi Aness,
The default location is "Your Personal Workspace > Documents". This will be shown in the "Recording Location" when you start the recording and users can change the location. If you cannot find it in "Documents", then you might have chosen another location. Try recording another conf and check the "Recording Location"
Thanks,
Jereen

East Coast Oracle Conference - April 25,26

Conference: Atlantic OTC, April 25, 26
Location: Washington Convention Center, Washington DC
Website: http://www.aotc-maop.org
Early Registration: $300 (save $100)
Catch top Oracle authors and presenters for a packed, 2-day agenda. Over 80 sessions will be presented by leading Oracle experts including Michael R. Ault, Steven Feuerstein, Kevin Loney, Tom Kyte, Paul Dorsey, Marlene Therriault, Dan Hotka, Bert Scalzo, Megh Thakkar, Oracle University and many others. Click below for a list of speakers and presentations.
http://aotc-maop.org/abstract/agenda.asp
If travel and training budgets are tight, this is a perfect, low-cost opportunity to get the latest Oracle information and save money. Oracle 9i new features, PL/SQL,XML and Java will be covered as well as numerous DBA topics. Registration includes beverages, refreshments, lunch and a reception after the event. Network with top industry experts and your colleagues. An exhibit hall and onsite bookstore will showcase the latest Oracle book titles and software solutions. Hope to see you there!
Best wishes,
Cam White,
AOTC Planning Committee

Hey Campbell,
Thanks for letting us know!!! I plan on attending! Please inform us of other events as they come up.
Thanks again,
LM

Flashbelt Conference in Midwest

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