Dial In Conference Bridge

Hello all. We have CCM 4.1(3)sr1 and Unity. We currently have a pub and a sub and have each configured as a software conference bridge for meetme's and adhoc's. Is it possible with this equipment to set up a permanent conference bridge that we can map and external DID that we can give to users to join a repeating weekly conference? If not, what is needed to make this happen, or can you direct me to the docs to find out? Thanks
Brian

Meetme number can be configured as its DID number itself (9725425900 for callers calling from outside. Telco sends 5900 to your PRI, 5900 is a meetme number)
or
you can create a Translation pattern which maps to a Meetme number. The Translation pattern could be a DID number. So your meet me number in this case will be a non-DID number say 5800. You create a translation pattern 5900 (which is a DID number) and translate that to 5800.
The conference host should dial 5900 at the right time of the conference so that people calling in will be added to the bridge.
HTH
Sankar.
PS: please remember to rate posts!

Similar Messages

Customized dial script for conference bridge meeting ID

Hello, first time poster, here.
I have the 8830 and want to peform the following actions:
open an apointment which contains a meeting ID for a conference call (when you dial into a bridge)
have an option with the 'blackberry' button to 'copy' that value
(similar to current built-in function to call a number (two 'bb' button pushes), but instead to autoselect on the word boundary and 'copy')
Then select the conference bridge dial-in number and, prior to dialing, insert a pause at the end of the phone number, then tack that ID number onto the end
Then dial the full number, including the ID value after the pause (allowing the autoattendant to answer), which will then feed it the ID and press pound
Is there a way to script this in the BB OS environment? Any info on tools to access these capabilities?
I'm a technical guy but new to the BB OS.
Thanks in advance.
Eric

In the Outlook invitation type the bridge numbers in this format. Use no more than 2 dashes in the bridge phone number. Skip one space. Type an x, the PIN and the # sign. The participants can sign on now with one BB click. The meeting organizer will still have to manually key in the start code, but can still one click the bridge to that point.
EXAMPLE: +1555-222-3456 x123456#

How to enter a conference bridge code?

I'm switching from Blackberry where I was able to enter a telephone number followed by the conference bridge code and all that I had to do to dial and enter the code was hit enter twice. It looked like this: 1-999-999-9999, #99999999*.
Is something similar available on iPhone?
Thanks.

Type the number in exactly the same, put a couple comas after the numer for pauses to allow the phone system to answer then enter the access code followed by a #, then press Send.

CUCM 9.x conference bridge not working

Hi all,
I'm new to CUCM and IP telephony, and as soon as I have set up my new CUCM I have encounter a problem, as it usually happens that way
Anyhow, I have install CUCM, insert phones in it, setup up router, all the dial peers, connect it via SIP to outside, and everything is working fine. Well, not everything. My conference calls can not be established. I know I need to set up dspfarm for conference with outside calls, however, I'm having problems even with creating conference inside my LAN with all phones being connected to same CUCM.
For example, I'm calling my college, she picks up, then I pres conference, call my other callege, she goes on waiting, he picks up, and I have line open with him, and she is on waiting... When I click conference again, I get the message on phone, no conference bridge.
I have Cisco IP Voice Media Streaming App up and running(i have tried reseting service, still nothing). Under conference bridge I have default conference bridge with default settings. I have also tried same thing on my trial CUCM 8.5x with demo licences and 3 softphones, and everything is working. However, on fully implemented CUCM 9.x and 7942 phones, get no conference bridge message all the time... Any help would be more then welcome

Hello Wesley,
It was only after I posted that I notice I have posted on wrong forum... My apologies to everyone, it was unintentional
Thank you kindly for your response, and although I have checked, and region is set to use G.711 and G.722, you still might be correct, casue, as I said, I'm new to all of this, and I might be looking into wrong thing... I will look a bit more into it, and see what I might have done wrong, and if I need further assistence, I will post it on right place next time
If I need to check correct answer, so this topic can be closed, please let me know, I will do so.
Once again, my apologies to everyone

Configure conference bridge in VG

Hi all,
I am configuring a conference bridge but i am not able to configure "maximum sessions 2 "
In my dspfarm profile also i do a no shut it asks me configure session but its not possible.
check config below
I am not able to configure maximum sessions 1 or 2 , 3 nothing
SiteC-RTR#sh run
Building configuration...
Current configuration : 6269 bytes
! Last configuration change at 20:04:27 GMT Mon Nov 18 2013
! NVRAM config last updated at 20:15:12 GMT Mon Nov 18 2013
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname SiteC-RTR
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
memory-size iomem 20
clock timezone GMT 0
network-clock-participate wic 2
dot11 syslog
ip source-route
ip dhcp excluded-address 10.10.202.1 10.10.202.119
ip dhcp excluded-address 10.10.202.130 10.10.202.254
ip dhcp pool SC
   network 10.10.202.0 255.255.255.0
   default-router 10.10.202.1
   option 150 ip 10.10.210.11 10.10.210.10
ip cef
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-net5
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
voice class h323 1
h225 timeout tcp establish 3
voice translation-rule 1
rule 1 /.+$....$$/ /\1/
voice translation-rule 4000
rule 1 /^4...$/ /+44207796\0/
voice translation-profile 4000
translate calling 4000
voice translation-profile 8digitANI
translate calling 4000
voice translation-profile strip
translate called 1
voice-card 0
dsp services dspfarm
archive
log config
hidekeys
controller E1 0/2/0
pri-group timeslots 1-4,16
interface Loopback0
ip address 10.10.110.3 255.255.255.255
ip ospf network point-to-point
h323-gateway voip bind srcaddr 10.10.110.3
interface Loopback1
ip address 10.10.115.1 255.255.255.0
ip ospf network point-to-point
interface FastEthernet0/0
no ip address
shutdown
duplex auto
speed auto
interface Service-Engine0/0
ip unnumbered Loopback1
service-module ip address 10.10.115.2 255.255.255.0
service-module ip default-gateway 10.10.115.1
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface FastEthernet0/1/0
switchport trunk native vlan 301
switchport mode trunk
switchport voice vlan 302
interface FastEthernet0/1/1
switchport trunk native vlan 301
switchport mode trunk
switchport voice vlan 302
interface FastEthernet0/1/2
switchport access vlan 301
switchport mode trunk
switchport voice vlan 302
interface FastEthernet0/1/3
switchport access vlan 301
switchport mode trunk
switchport voice vlan 302
interface Serial0/2/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn outgoing display-ie
no cdp enable
interface Serial0/3/0
ip address 10.10.114.2 255.255.255.0
encapsulation frame-relay IETF
ip ospf mtu-ignore
shutdown
fair-queue 64 256 2
frame-relay interface-dlci 102
ip rsvp bandwidth 64
interface Serial0/3/1
ip address 10.10.112.2 255.255.255.0
encapsulation ppp
ip ospf mtu-ignore
clock rate 2000000
interface Vlan1
no ip address
interface Vlan302
ip address 10.10.202.1 255.255.255.0
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
ip forward-protocol nd
ip route 10.10.115.2 255.255.255.255 Service-Engine0/0
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:GUI/admin_user.html
disable-eadi
control-plane
voice-port 0/2/0:15
translation-profile incoming strip
ccm-manager fax protocol cisco
sccp local Vlan302
sccp ccm 10.10.210.11 identifier 1 version 7.0
sccp ccm 10.10.210.10 identifier 2 version 7.0
sccp ccm group 1
bind interface Vlan302
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 3 register sc-cfb
associate profile 2 register sc-mtp
associate profile 1 register sc-xcoder
dspfarm profile 2 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 4
associate application SCCP
dspfarm profile 3 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
associate application SCCP
shutdown
dspfarm profile 1 mtp
codec g729r8
codec pass-through
rsvp
maximum sessions software 100
associate application SCCP
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
dial-peer voice 2 voip
destination-pattern 4...
voice-class codec 1
voice-class h323 1
session target ipv4:10.10.210.11
incoming called-number .
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 3 voip
preference 1
destination-pattern 4...
voice-class codec 1
voice-class h323 1
session target ipv4:10.10.210.10
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 999 pots
translation-profile outgoing 8digitANI
destination-pattern 999
port 0/2/0:15
forward-digits 3
dial-peer voice 8 pots
translation-profile outgoing 4000
destination-pattern 9[1-9].......
port 0/2/0:15
forward-digits 8
dial-peer voice 11 pots
translation-profile outgoing 4000
destination-pattern 90[1-9].........
port 0/2/0:15
forward-digits 11
dial-peer voice 900 pots
translation-profile outgoing 4000
destination-pattern 900T
port 0/2/0:15
prefix 00
num-exp 2...$ 90012025552...
num-exp 3...$ 90014083873...
gatekeeper
shutdown
telephony-service
srst mode auto-provision all
srst dn line-mode octo
max-ephones 4
max-dn 4
ip source-address 10.10.202.1 port 2000
time-zone 21
time-format 24
max-conferences 4 gain -6
web admin system name admin password cisco
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Nov 18 2013 19:20:30
ephone-dn 1
number 4002
description +442077964002
name +442077964002
ephone-dn 2
number 4000
description +442077964000
name +442077964000
ephone-dn 3
number 4001
description +442077964001
name +442077964001
ephone 1
device-security-mode none
mac-address 0017.95B0.D7CE
button 1:1 2:2
ephone 2
device-security-mode none
mac-address 0018.187C.1844
button 1:3 2:2
line con 0
exec-timeout 0 0
password 7 1511021F0725
login
line aux 0
line 66
no activation-character
no exec
transport preferred none
transport input all
transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
line vty 0 4
exec-timeout 0 0
password 7 121A0C041104
login
scheduler allocate 20000 1000
ntp server 10.10.110.1
end

Hi Jaime,
Thanks for your reply.. to be honest this is first time that i'm configuring this its lab purposes.
I've checked my voicegatewat vesion:
Cisco 2801 (revision 6.0) with 314368K/78848K bytes of memory.
Processor board ID FTX1036W2D0
6 FastEthernet interfaces
5 Serial interfaces
2 Serial(sync/async) interfaces
1 terminal line
1 Channelized/Clear E1/PRI port
1 Virtual Private Network (VPN) Module
1 DSP, 16 Voice resources
1 cisco service engine(s)
DRAM configuration is 64 bits wide with parity disabled.
191K bytes of NVRAM.
62720K bytes of ATA CompactFlash (Read/Write)
Configuration register is 0x2102
Will this answer your question in combination with the config i'v attached earlier?
@ gergely=====>
This is output of sh voice dsp voice
SiteC-RTR#sh voice dsp voice
edsp 0001 01 g729r8 p 0.1 IDLE 50/0/1.1
edsp 0002 01 g729r8 p 0.1 IDLE 50/0/2.1
edsp 0003 01 g729r8 p 0.1 IDLE 50/0/3.1
----------------------------FLEX VOICE CARD 0 ------------------------------
                           *DSP VOICE CHANNELS*
CURR STATE : (busy)inuse (b-out)busy out (bpend)busyout pending
LEGEND     : (bad)bad    (shut)shutdown (dpend)download pending
DSP   DSP                 DSPWARE CURR BOOT                         PAK   TX/RX
TYPE NUM CH CODEC        VERSION STATE STATE   RST AI VOICEPORT TS ABRT PACK COUNT
===== === == ========= ========== ===== ======= === == ========= == ==== ============
C5510 001 01 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 02 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 03 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 04 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 05 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 06 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 07 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 08 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 09 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 10 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 11 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 12 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 13 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 14 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 15 None          23.8.6 idle idle      0 0                 0          0/0
C5510 001 16 None          23.8.6 idle idle      0 0                 0          0/0
------------------------END OF FLEX VOICE CARD 0 ----------------------------

Video Conferencing with IOS Homogeneous Video Conference Bridge

Dears,
I have CUCM 8.6 and 3945 Cisco router with 2 x PVDM-256
I configured the IOS Homogeneous Video Conference Bridge. and I did the required configurations in the router. and I can see the conferencing bridge is registered.
I put the conference bridge in the existing MRG and MRGL. But I am not able to do video conferencing. I can get only the audio.
is there any thing missing ? any reolad should I do ?

Sure.
So on the gateway that you have the video conference configured, in which we have guranteed voice as the configuration, you will also add in an MTP profile. The MTP profile will be configured as software and have the appropriate codec configured that you may be using because of your regions. We have g722-64 and passthrough on one because we saw the E20's requesting MTP resources from the logs.
We also have additional MTP profiles configured for other codecs to include G729r8, etc.
Register the codecs to CUCM.
It depends on what you are trying to accomplish but we were shooting for w448p thus made sure that the "Region" for the device pool associated with the Media Resource Group for each MTP was set to 1MB for the video bandwidth.
I hope this makes since if not I will try to pull down a config for you

I have Cisco TelePresence SX10 and content sharing is not working when I am dialing through the bridge

I have Cisco TelePresence SX10 and content sharing is not working when I am dialing through the bridge, I can share the content if I drag and drop from RMX, but if schedule the call in Resource manager or manually dial in from device the content is not going to other hand, I have tried to turn off encryption as well but still same issue. can you please help me out with this. I am from Lion co and purchased sx 10 recently.
regards
Hemang

Can you please provide us with a little more information on your systems and configuration / topology, such as, what call control are you using (Cisco VCS, CUCM, other?). what type of "Bridge" are you using (is this a Cisco MCU, or Cisco TelePresence Server, or other device?), what versions of the software are on each of the devices, etc. The more information we have about your environment will help us assist you better. But saying that, if all your core equipment isn't Cisco, you may have more luck in the forums for the manufacturer of such equipment (ie Polycom's Support Community).
Wayne
Please remember to rate responses and to mark your question as answered if appropriate.

Problems in conference bridge after migrating to 4.1.3SR3 from 4.1.3

very recently i upgraded my CCM from 4.1.3 to 4.1.3 SR3. i use 3725 with DSP farm as conference bridge ( the ios code on 3725 is 12.3.11 T7 )
the conferencing stopped working for me as soon as press conference on my after callint the 2md party, both my calls get dropped.i see that the CCM registers the bridge and also in SCCP config of gateway i see the CCM registered.
for testing purpose i configured one my 3845( with same version) over WAN as my conference bridge and then conference worked fine for me.
there are few command difference in 3725 and 3845 . In 3845 i can configure CCM version as 4.1 and in 3725 i can do only 3.1 or greater. i am not sure if this is problem or not.
in CCM traces i got this unusal error which is attached
please let me know any pointers on this
thanks
manish

We are also having the same issue, 3725 used as conf resource, as soon as we press conf button all calls are dropped with fast busy. we are using CM 3.3.5 and IOS 12.3.7T12.
cisco TAC poked around and could not find anything wrong with config (of course, we have other sites with identical config all working ok). so they said it must be bad DSPs even thou there are no DSP errors on the router. i doubt it is DSP issue but i'll let you know when TAC figures it out.

IP communicator giving error as "No conference Bridge"

Hello This is Rahul,
I am having CUCM 8 and have installed three ip communicators (CIPC)in it. Now when i try to take all three communicators in a conference call i am getting error "No conference Bridge" on IP phone from which i am trying for conference call. Can anyone help here to solve this issue.

Rahul,
The Conference Bridge Configuration procedure is detailed here http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_0_2/ccmcfg/b04cnbrg.html
Please complete the steps then you should be able add the conf bridge to MRGL & thus will be available for phones to utilise.
Pls rate the post if it helps.
GP.

Type of conferences bridge

In the Cisco Call Manager version 6, the types of conferences bridges are: Bridge Hardware, Cisco IOS Conf_Bridge, Dixieland CFB, Subbit CBT and Flint Conf Bridge, of all this, What kind of type is the MCU 3545?

You should have the option of a video conference bridge.
Brandon

CallManager 5.1 no conference bridge available / cannot complete conference

CM 5.1, Conferencing was working at one point, the servers were rebooted and now conferencing does not work, received no conference bridge available error message when trying to complete conference, found the CM was advertising g722 codec, disabled that function. Now receiving cannot complete conference error message, rebooted IP voice streaming media services, reset conference bridges, both are registered, anybody got any other ideas on why this is not fuctioning properly??
Thanks ahead of time...

No Available Conference Bridge
Symptom
The following message displays: No Conference Bridge Available.
Possible Cause
This could indicate either a software or a hardware problem.
Recommended Action
1. Check to see whether you have any available software or hardware conference bridge resources that are registered with Cisco Unified CallManager.
2. Use the Cisco Unified CallManager Real-Time Monitoring Tool to check the number of Unicast AvailableConferences.
The Cisco IP Voice Media Streaming application performs the conference bridge function. One software installation of Cisco IP Voice Media Streaming will support 16 Unicast Available Conferences (three people/conference), as shown in the following trace.
To make a call between two devices that are using a low bit rate code (such as G.729 and G.723) that do not support the same codec, you need a transcoder resource.
Assume Cisco Unified CallManager has been configured such that the codec between Region1 and Region2 is G.729. The following scenarios apply:
?If caller on Phone A initiates a call, Cisco Unified CallManager realizes it is a Cisco Unified IP Phone model 7960, which supports G.729. After the digits are collected, the Cisco Unified CallManager determines that the call is destined for User D who is in Region2. Because the destination device also supports G.729, the call gets set up, and the audio flows directly between Phone A and Phone D.
?If a caller on Phone B, who has a Cisco Unified IP Phone model 12SP+, initiates a call to Phone D, this time the Cisco Unified CallManager would realize that the originating phone only supports G.723 or G.711. Cisco Unified CallManager would need to allocate a transcoding resource so audio would flow as G.711 between Phone B and the transcoder but as G.729 between the transcoder and Phone D. If no transcoder were available, Phone D would ring, but as soon as the call was answered, the call would disconnect.
?If a user on Phone B calls Phone F, which is a Cisco Unified IP Phone model 12SP+, the two phones would actually use G.723, even though G.729 is configured as the codec to use between the regions. G.723 gets used because both endpoints support it, and it uses less bandwidth than G.729.

IOS Conference Bridge Vs. IOS Transcoder

Hello all,
I was wondering what the primary advantage is to setting up both IOS Hardware Transcoding resources and an IOS conference bridge for DSP farm use.
The reason I ask is that I know that an IOS transcoder can actually be used for conference purposes, is the IOS conference bridge a more efficient way to do this vs. just using the transcoder?
The reason I ask is because from my research our end users need to use the transcoding resource far more often than conference bridging for g.729-g.711 conversions. If I wanted to utilize the DSP farm resources for a router and knew that transcoding would be used frequently and conference bridging only occasionally wouldn't it make sense to configure that resource as as a transcoder and let it also handle the conference bridging, even though it will not be as efficient at it as a dedicated IOS resource?

Hi,
I believe DSP chips can be set as a transcoder or conf bridge, but once you allocate a DSP chip as a Transcoder, it remains as transcoder and cannot be used as a Conf Bridge. Of course a Transcoder can also act as an MTP.
Secondly, you definitely need DSP based HW Transcoders whenever you need anything which does not involve G.711 (e.g. G.729 to G.729ab or G.726 etc) aka Universal Transcoding. Software Transcoders can only do transcoding between G.711 and any other Codec.
It really depends on if your network needs a transcoder or not (in my case, so far I have never needed a transcoder) however there have been plenty of Conf Bridge requirement (e.g. Meet Me Conf Bridges or Ad-Hoc Conf bridges).
A transcoding can be really avoided whole together if either your system is completely G.711 or both the parties can / do support the other alternate codec e.g. a remote site to local site can be configured to use G.729 codec, however if both the parties can use G.729, then there is no need for a Codec at all.
HTH

No conference bridge

Hi.
I have CUCM 7 , and suddenly conference function stop worked. I can call to one person, hit conference button, call to another person, but when hitting conference button again, it says - no conference bridge.
In media recources i have configured software conference bridge which is assigned to device pool and have registred status. Cisco IP Voice Media Streaming App is started.
I try to add it to MRG and MRGL, it does not help, but before worked without adding CFB to MRGL.
I found some solution for single device - in phone configuration - add manually MRGL which by default is none, but i have more then 100 phones and this is will be not proper way to make conference function work on all phones.
Any suggestions about why it stop worked and how to bring it back?
Codec used by default is g711. CUCM has been rebooted 2 times, and CFB has been readded , but it does not help.

Make sure all the region settings are set to g711. If the conference bridge is allocated to a MRG then it will not be available by default to other phones which do not have a MRGL defined with this resource in the MRG. For all the IP phones to have access to it by default, you need to remove it from all MRG's. However, assigning the MRGL's to devices with proper MRG's is the recommended approach.
HTH
Manish

Meetme No Conference Bridge

I have had my conference bridge fail, when attempting either a meetme or adhoc conference brige I get a No conference Bridge message on the phone. I am running a software conference Bridge and have made no changes to any of my bridges. I have CUCM 8.6., any advice is appreciated.

What type of codec are using the IP phones and the conference bridge?
what is the status of the conference bridge shows at call manager.?
Br,
Nadeem
Please rate all useful post.

How to trigger Linksys SPA 942 to send the info configured in Conference Bridge URL?

According to the Adminstration Guide for Linksys SPA 942, Conference Bridge URL field under EXT tab should be used to join into a conference call. However, when I press on CONF or confLx, I do not see the Conference Bridge URL gets sent in the INVITE message. Is there anything special I need to do to trigger the Conference Bridge URL to be sent?

Conference Bridge URL is the URL used to join into a conference call, generally in the form of the word “conference” or “user@IPaddressort”. By default it is blank so you need to fill the specific user and IP address with the appropriate port for the said settings to work. I further look into this matter and maybe you can be guided with this via this link:
https://www.myciscocommunity.com/thread/1457
Other than that I suggest contacting Cisco Tech support to further look into your concern. I believe this unit belongs to the business series devices that Cisco is now supporting. Try to go to this link for the other business series devices and the site where you can get hold of Cisco for support:
http://www.cisco.com/web/products/linksys/index.html

Dial In Conference Bridge

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