Media Termination Point

Hello There,
We have a CCM 4.1(3). In the Gateway configuration page do I need to enable the Media Termination Point? What exactely does the MPT?
Thanks,

Bahman,
You can read all about MTPs here.
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00801ec5be.html

Similar Messages

MTP - Media Termination Point and SBC - Session Border Controller

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A client of mine wants to build up a MTP - Media Termination Point and SBC - Session Border Controller to use it to connect to the internet or leased lines to their VoIP provider at USA and route the calls into their current IPCC and also use it with their auto dialer to outbound the international calls.
I NEED TO KNOW WHAT ARE THE REQUIREMENTS FOR THIS REQUEST AND HOW CAN IT BE IMPLEMENTED??!!

try this for SBC in XR12000.
http://www.cisco.com/application/pdf/en/us/guest/products/ps6342/c1244/cdccont_0900aecd80391b66.pdf

Media Termination points and media paths with CUCM

Hi all,
I could really use some help regarding this topic. Please allow me to explain the situation.
Ok we have some endpoints that are not Cisco but cisco have developed cop files for CUCM and they are seen as a cisco device and register the same as any other phone. This isn't anything to do with the problem, but I want to make this clear. These are actually dealerboards for trading and register cisco lines the same as any other cisco telephone.
Ok so here is the story;
We have a CUCM cluster in London and a local gateway for local PSTN breakout. There are endpoints that are in Dubai and the register to the CUCM cluster in London. Dubai have their own MGCP gateway for local breakout to PSTN.
Here is the problem;
When a Dubai endpoint makes a call to a PSTN number in dubain there RTP is send from the endpoint, back to the London CUCM, the London CUCM then points the RTP at the gateway in Dubai. This is using a lot of bandwidth and I am not clear on how to change this so the endpoints send media directly to their MGCP gatway.
In my lab I have sent up 2 CUCM's connected by E1 via MGCP gateways, this is to simulate the PSTN network. I can see that on CUCM 7 this is not the case and media is sent directly to the MGCP gateway, on CUCM 8 media is sent via the CUCM to the MGCP gateway. I have since then added in a software MTP on the MGCP gateway for CUCM 8 and now everything hits the gateway as I have specified MTP is required on the phones profile. This was good but even when I make an internal call, everything hits the MTP instead of using IP-IP direct medial.
Please can someone advise what I need to do or if this type of behaviour is expected.
Many thanks,
Joe

Hi,
You can use this link to learn how to collect cum traces..
https://supportforums.cisco.com/document/126666/collecting-cucm-traces-cucm-862-tac-sr
We can help you look at the logs and explain whats going on..
Please include the calling and called number and time of call..Check the traces to ensure that the call is there first before sending it over here.
What type of phones are you using? What dtmf method have you specified in your mgcp configuration

Changing the Media Start point

So I just had a hard drive crash on me and I did not have back-ups of my footage. It's not to bad because I have back-ups of the project file I was working on. So all I was going to do is a Batch Capture to reconnect all of my clips. Only problem is the person I had capture the footage did not leave enough pre-roll when they did a capture now. So now when it tries to re-capture it says "Cannot locate specified timecode"
This is because there is not enough pro-roll on the clip to allow it to batch capture. When I look at the Media Start in the browser window for the clip it says it starts at 00:00:02:31. I need to change this to a later start so it does not try to roll the tape so far back. So.....
How do I change the Media Start Point???
Thanks

Ok, I am following you, I just need to make sure the clips in the edited timeline reconnect with the old master clip.
I could keep the old offline master clip and just make it reconnect to the new captured clip. But will this make all of my already edited clips in my sequence that are looking at the old master clip off by however many seconds I had to reset the in-point to get the tape re-captured?
Thanks for the tip on dragging clips to the L&C window, I did not know you could do that.

How to determine termination point of call

We're using UCCE 7.2 and are moving to 9.0 in a few weeks. One of the questions that we have had is how to determine the termination point of a call. I'm looking for a way to report through either the HDS or call manager whether the agent or the caller disconnected the call.
Does anyone know if there is a way to identify this in any of the reporting databases available?
Thanks,
Randy Black

I should probably provide a bit more detail about what I'm trying to determine. We have two different situations:
1: Customer reports that a call was disconnected. Agent reports that they did not hang up on the customer and we need to determine if the call was terminated by the agent or the customer.
2: We offer a post-call survey using a custom CVP application. Customers opt in to the survey at the beginning of the CVP call flow, the customer makes their IVR selections and are routed to an agent. Once the agent disconnects, the customer is routed back to CVP to take the post-call survey. We have quite a few customers that opt in for the survey, but never complete the survey. We know that some number of these are not taken because the customer forgets or changes their mind, but suspect that number of these are not completed because the agent does not disconnect the call so the customer is never routed to the survey. We have a policy in place that requires the agent to disconnect the call after the call is completed, but want to determine what agents are not disconnecting the calls.
I'm looking to determine agent vs. customer termination on all calls so we can confirm who disconnected.
I've looked at the records in the Termination Call Detail on the HDS, but have been told that this is not a great way to identify this information so I'm looking for any alternatives that are more accurate for this reporting.
Thanks,
Randy

CER_TELEPHONY | Cannot register media terminal for port : 7000000

CER_TELEPHONY | Cannot register media terminal for port : 7000000
This error is coming up over and over again in my Cisco Emergency Responder logs. The media port correlates back to CTI ports on call manager with the same identifier. So I went to CUCM and issued a reset on the CTI port that matches but I am still getting the same error. My 911 calls are working just fine btw. Any ideas on how to clear this up or if it could potentially cause issues?
RESOLVED:
I hit the correct button by accident. The fix to this was to restart CTI Manager on CUCM that was connecting to CER.

The partition they have on the CTI port wouldn't matter would it? CER would communicate to the Application user account which would then register up one of those ports to initiate the call to the destination number.
Since nothing is calling the CTI port it really wouldn't matter what partition they are in as long as their CSS contained the partition of the destination number right?
About the RTMT trace.. how would I know what the calling number is? I had the thought that the calling number would be the CTI ports DN. Is my train of thought correct? I can try a trace on the called number and see if I can track it that way though. Thanks for the suggestions!

When Batch Capturing, Final Cut doesnt notice Media Out points

I have logged about 20 clips with all of the in and out points set and correct. I have put them in a logging bin and choose batch capture and it begins the process just like usual. However now it wont stop when it comes to the out point of each clip the camera will just keep running until i cancel the batch capture. What could be wrong? any help would be great. thanks

You need more than one hard drive. Capturing to your system disk is not a good idea and will bite you.
The in/out ignoring issue could be a problematic QT install or a problem with the firewire protocol.
1. Make sure you are using the correct Firewire (Basic or std) for your camera/VTR first.
2. Make sure there is no interference between external hard drives on the firewire bus and your camera
3. Use the Apple QT7 reinstaller
Cheers,
x

Jabber cannot do voice calls to PSTN or to other jabber user

Hi,
I'm trying to troubleshoot why jabber for windows 9.7.2 is not able to establish calls to PSTN dialed by DN and to other jabber user called by URI, not DN.
We have CUCM 10 with IM and Presence 10. IM and Presence is newly installed. I have created 2 x Cisco Unified Client Services Framework devices for 2 users. Users are able to connect through jabber for windows, IM, do direct DN-to-DN calls (including video), but not able to call outside to PSTN and by URI to other users.
Cases:
- jabber for windows - call DN to DN between internal users - working with video
- jabber for windows - call DN to PSTN through SIP-to-SIP connection (SIP trunk to GW, SIP trunk from GW to ISP) - not working CUCM returning me Q.850 cause 65 to jabber immediately when other side pick up phone, even jabber hear ringing with music from ISP side. - Cause: 65(0x41)[Bearer capability not implemented]
- jabber for windows - in search field I search for second user, I can see presence, when I click call button, call is immediately dropped - don't know reason still not troubleshooted.
communication between GW 2801 and ISP is OK, they are acknowledge codec and everything is fine.
EDIT: I tried to connect jabber for windows directly to CUCM TFTP servers without IM and Presense server. I have same behavior, when outgoing call is picked up it is immediately dropped by CUCM and send Q.850 cause 65 to jabber.
EDIT2: any other SIP device configured same way as jabber CSF is working. just jabber not.
EDIT3: when I selected "Media Termination Point required" on jabber device in CUCM, then voice is working normally. But if possible I do not want this and make voice without MTP. How?
THanks

I found it, i have enabled "Allow Presentation Sharing using BFCP" on SIP profile assigned to trunk pointing to voice gateway. This causes that SDP from gateway didn't come.

Can CUBE register with two CUCM clusters?

We have two CUCM clusters - one is in US and one is in Australia. Currently CUBE is registered with US Cluster with the settings below -
sccp local GigabitEthernet0/0
sccp ccm 10.10.1.21 identifier 2 priority 2 version 7.0
sccp ccm 10.10.1.20 identifier 1 priority 1 version 7.0
sccp
Now we need CUBE to communicate with Australia CUCM. Should we set sccp up for Australia CUCM cluster (version 6.0)?
Thanks,
Jessica Wang

I think you are refering to registering media termination points / transcoders on the same router to two different CUCM clusters, correct?
If yes, we can do it by creating separate sccp ccm groups.
Example :
sccp ccm identifier 1 version 7.0
sccp ccm identifier 2 version 7.0
sccp ccm identifier 3 version 7.0
sccp ccm identifier 4 version 7.0
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register Transcoder1
sccp ccm group 2
associate ccm 3 priority 1
associate ccm 4 priority 2
associate profile 2 register Transcoder2
dspfarm profile 1 transcode
codec g729r8
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 5
associate application SCCP
dspfarm profile 2 transcode
codec g729r8
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 5
associate application SCCP
Arun

Using CUCM as a proxy like service

Here is the situation: I have a "BOYD" wireless SSID for all employees to use, I would like to move our mobile jabber clients off the current clinical device SSID and onto the new one. As this would be simple enough normally, but our security engineers are refusing to open up ACL's for all of the voice vlans throughout the enterprise. I have opened up the ACL's to the UC servers only at this point which allows the phones to register, get voice mail and send participate in chats. As it is configured now the calls are placed with one way audio( no surprise here ) the ACL's are blocking the RTP streams from connecting phone to phone.
What I would like to accomplish: Is to keep CUCM in the middle as the termination point of both call legs, so it would work like a proxy service for the calls but only for the jabber client calls. I know that ip media service on CUCM is capable of terminating a call but I'm not exactly sure how to approach this.
Will an ios media termination point accomplish this man in the middle feat? Is there something i could do in my media resources list/group to create the desired behavior?
Any ideas would be greatly appreciated.

This was actually an easier problem to solve then I had anticipated. All that was really required was a new MGRL with CUCM software MTP MRG associated. Then build a phone template for the Dual mode Jabber for iPhone and Android, change the use trusted relay box to on and save.
The call manager cluster is now standing in the middle of all the calls made from the mobile jabber clients. This fixes the audio issue.
Now I have to find a way ( if even possible ) to allow for video to pass through.

SCCP Transcoders and Conference doesn't register.

Dear Friends,
I having a strange issue, I have configured my 3945 GW with SCCP for Conferencing and Transcoding. Also on the CUCM Side, i have used the Cisco IOS Enhanced media termination Point for this setup but I see the transcoders are not getting registered. I had reset the items from both the sides but No luck.
In the "Debug SCCP all" i do see error 257 in socket.
The configs:
sccp local GigabitEthernet0/2
sccp ccm 19.106.182.15 identifier 2 version 7.0
sccp ccm 19.106.214.15 identifier 1 version 7.0
sccp
sccp ccm group 1
bind interface GigabitEthernet0/2
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 2 register CFB-12345678901
associate profile 1 register XCD-12345678901
dspfarm profile 1 transcode
description #XCODE dsp farm - G711/G729r8 only#
codec g711ulaw
codec g711alaw
codec g729r8
maximum sessions 30
associate application SCCP
dspfarm profile 2 conference
description Ad-hoc conference - G711 only
codec g711ulaw
codec g711alaw
maximum sessions 8
associate application SCCP
DEBUG sccp all:
Jan 14 15:20:13: sccp_connect_to_ccm_on_priority_basis: Trying connecting to CCM on priority basis prof_id 2, appl_type 3
Jan 14 15:20:13: sccp_connect_to_ccm_on_priority_basis: Trying CCM with ipaddr 19.106.214.15, priority 1, port 2000
Jan 14 15:20:13: sccp_appl_service_start_timer: Start timer type 1 with time_out 30000ms
Jan 14 15:20:13: sccp_tcp_socket_connect: Trying tcp soc connect for appl_type 3, prof_id 2, to ipaddr 19.106.214.15
Jan 14 15:20:13: sccp_get_ccm_intf_vrf_id: ccm sccp local interface vrfid=0appl type is 3, appl profile 2
Jan 14 15:20:13: sccp_tcp_open_and_set_option: Socket 1 opened and binded to addr 19.106.214.15 - for appl_type 3, prof_id 2
Jan 14 15:20:13: sccp_socket_connect_to_ccm :: connecting to port 2000 sccp_socket_connect_to_ccm: soc conn to 19.106.214.15 port 2000 in progressfor appl_type 3, state 1, soc_fd 1 appl 144C622C
Jan 14 15:20:13: sym_xapp_xlate_ccapi_eve: Unsupported ccapi eve 2
Jan 14 15:20:13: sym_xapp_process_ccapi_events: Unsupported ccapi eve_id 2
Jan 14 15:20:13: sccpapp_process_socket_events: appl_type 2, soc_fd 0, soc 0, swb_soc -1
Jan 14 15:20:13: sccpapp_process_socket_events: TCP_SOCKET_READ: appl_type 2, eve 4, state 1
Jan 14 15:20:13: sccp_appl_service_stop_timer: Stop 2064E60 timer
Jan 14 15:20:13: sccp_process_socket_connect_result: appl 2064E08, soc_recv err 257, len -1, appl_type 2
Jan 14 15:20:13: sccp_process_socket_connect_result: appl_type 2, eve 1, soc_id -1, state 1

fingw001-vg3945#sh running-config
Building configuration...
Current configuration : 25223 bytes
! Last configuration change at 13:06:36 UTC Tue Jan 14 2014 by kmohsin
! NVRAM config last updated at 12:22:27 UTC Tue Jan 14 2014 by skarth43
version 15.1
no service pad
service timestamps debug datetime localtime
service timestamps log datetime localtime
service password-encryption
hostname fingw001-vg3945
boot-start-marker
boot system flash:c3900-universalk9-mz.SPA.151-3.T2.bin
boot-end-marker
logging buffered 16384
no logging console
enable secret 5 $1$OFF8$8Z8jdwPVNpsGaUg4DYXR11
aaa new-model
aaa authentication login default group tacacs+ enable
aaa authentication enable default group tacacs+ enable
aaa authorization config-commands
aaa authorization commands 0 default group tacacs+ if-authenticated
aaa authorization commands 15 default group tacacs+ if-authenticated
aaa accounting update newinfo
aaa accounting commands 15 default
action-type start-stop
group tacacs+
aaa session-id common
errdisable recovery cause udld
errdisable recovery cause rootguard
errdisable recovery cause pagp-flap
errdisable recovery cause dtp-flap
errdisable recovery cause link-flap
errdisable recovery interval 60
no ipv6 cef
no ip source-route
ip cef
ip multicast-routing
ip domain list nls.ford.com
ip name-server 19.155.192.72
ip name-server 19.175.128.73
multilink bundle-name authenticated
crypto pki token default removal timeout 0
voice-card 0
dspfarm
dsp services dspfarm
voice call send-alert
voice call convert-discpi-to-prog
voice rtp send-recv
voice dsp crash-dump file-limit 3
voice dsp crash-dump destination flash:dsp-crash
voice service voip
ip address trusted list
ipv4 19.106.182.0 255.255.255.0
ipv4 19.106.214.0 255.255.255.0
address-hiding
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
header-passing
error-passthru
asserted-id pai
early-offer forced
midcall-signaling passthru
privacy-policy passthru
sip-profiles 1
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice class sip-profiles 1
request INVITE sip-header Supported remove
request INVITE sip-header Min-SE remove
request INVITE sip-header Session-Expires remove
request INVITE sip-header Unsupported modify "Unsupported:" "timer"
request ANY sip-header Allow-Header modify ", UPDATE" ""
response ANY sip-header Allow-Header modify ", UPDATE" ""
voice translation-rule 1
rule 1 /$^...$$/ /7319\1/
voice translation-rule 2
rule 1 /^103447$...$$/ /7319\1/
voice translation-rule 3
rule 1 /^7319$...$$/ /103447\1/
voice translation-rule 4
rule 1 /^9$.*$/ /\1/
voice translation-profile Internal-to-PhoneDN
translate called 1
voice translation-profile PSTN-to-PhoneDN
translate called 2
voice translation-profile PhoneDN-to-PSTN
translate calling 3
translate called 4
application
package callfeature
param med-inact-det enable
param med-inact-disc-cause 44
param med-inact-action disconnect
global
service alternate DEFAULT
license udi pid C3900-SPE150/K9 sn FOC174198Z0
license accept end user agreement
hw-module pvdm 0/0
vtp domain fingw001-vg3945
vtp mode transparent
redundancy
ip tcp path-mtu-discovery age-timer 30
interface GigabitEthernet0/0
no ip address
shutdown
duplex auto
speed auto
interface GigabitEthernet0/1
description rhost="finfw001" rint="G1/0"
ip address 19.155.95.206 255.255.255.252
duplex auto
speed auto
service-policy input EdgeMark2
interface GigabitEthernet0/2
description rhost="finrs001" rint="G3/22"
ip address 19.155.95.198 255.255.255.252
no ip redirects
duplex auto
speed auto
ip forward-protocol nd
no ip pim dm-fallback
ip pim autorp listener
no ip http server
no ip http secure-server
ip flow-cache timeout active 1
ip flow-export source GigabitEthernet0/2
ip flow-export version 5
ip flow-export destination 19.97.9.164 9995
ip route 0.0.0.0 0.0.0.0 19.155.95.197
ip route 10.80.144.0 255.255.255.0 19.155.95.205
ip route 192.168.13.232 255.255.255.252 19.155.95.205
ip tacacs source-interface GigabitEthernet0/2
ip access-list extended Virus_LAN
deny   53 any any
deny   55 any any
deny   77 any any
deny   pim any any
deny   tcp any any eq 5554
deny   tcp any any eq 8594
deny   tcp any any eq 8563
permit ip any any
ip access-list extended Virus_LAN_PIM
deny   53 any any
deny   55 any any
deny   77 any any
deny   tcp any any eq 5554
deny   tcp any any eq 8594
deny   tcp any any eq 8563
permit ip any any
ip access-list extended Virus_Server
deny   53 any any
deny   55 any any
deny   77 any any
deny   pim any any
deny   tcp any any eq 5554
deny   tcp any any eq 8594
deny   tcp any any eq 8563
permit ip any any
ip access-list extended Virus_Server_PIM
deny   53 any any
deny   55 any any
deny   77 any any
deny   tcp any any eq 5554
deny   tcp any any eq 8594
deny   tcp any any eq 8563
permit ip any any
ip access-list extended Virus_WAN
deny   53 any any
deny   55 any any
deny   77 any any
deny   pim any any
deny   udp any any eq 1434
deny   tcp any any eq 5554
deny   tcp any any eq 8594
deny   tcp any any eq 8563
permit ip any any
ip access-list extended Virus_WAN_PIM
deny   53 any any
deny   55 any any
deny   77 any any
deny   udp any any eq 1434
deny   tcp any any eq 5554
deny   tcp any any eq 8594
deny   tcp any any eq 8563
permit ip any any
ip sla responder
logging trap debugging
logging source-interface GigabitEthernet0/2
logging 19.97.9.50
logging 19.97.9.80
access-list 4 remark - SNMP access for NMS subnet
access-list 4 permit 19.97.9.0 0.0.0.255
access-list 9 remark - SNMP access for HP DDMA subnets
access-list 9 permit 19.106.113.16 0.0.0.15
access-list 9 permit 19.106.113.32 0.0.0.15
access-list 9 permit 19.106.117.16 0.0.0.15
access-list 9 permit 19.106.117.32 0.0.0.15
access-list 9 permit 19.110.113.16 0.0.0.15
access-list 9 permit 19.110.113.32 0.0.0.15
access-list 9 permit 19.110.117.16 0.0.0.15
access-list 9 permit 19.110.117.32 0.0.0.15
access-list 141 remark * Iron class - Interactive Queue (IP Precedence 1)
access-list 141 remark - generic TCP traffic, excludes Internet web & C3P Metaphase Mux
access-list 141 remark - Public Internet proxy HTTP/HTTPS excluded (drops to default queue)
access-list 141 deny   tcp any any eq 83
access-list 141 deny   tcp any eq 83 any
access-list 141 remark - C3P bulk traffic port number
access-list 141 deny   tcp any any eq 4544
access-list 141 deny   tcp any eq 4544 any
access-list 141 deny   tcp any any eq 16016
access-list 141 deny   tcp any eq 16016 any
access-list 141 remark - MX2 Handhelds (intentionally one way)
access-list 141 deny   tcp any range 4000 4050 any
access-list 141 remark - The following permit catches any non-excluded TCP
access-list 141 permit tcp any any
access-list 141 remark - SNMP traffic from NetOps Mgt subnet with CS1 *
access-list 141 permit udp 19.97.9.0 0.0.0.255 any eq snmp
access-list 141 permit udp any eq snmp 19.97.9.0 0.0.0.255
access-list 141 permit udp any 19.97.9.0 0.0.0.255 eq snmptrap
access-list 142 remark * Bronze class - Interactive Queue (IP Precedence 2)
access-list 142 remark - DNS query/response, Pinnacle
access-list 142 permit udp any any eq domain
access-list 142 permit udp any eq domain any
access-list 142 remark - HTTP traffic (within Ford only)
access-list 142 permit tcp any any eq www
access-list 142 permit tcp any eq www any
access-list 142 remark - SSL Web traffic
access-list 142 permit tcp any any eq 443
access-list 142 permit tcp any eq 443 any
access-list 142 deny   tcp any any lt 1024
access-list 142 deny   tcp any lt 1024 any
access-list 142 permit tcp any any eq 8058
access-list 142 permit tcp any eq 8058 any
access-list 143 remark * Silver class - Video Queue (IP Precedence 3)
access-list 143 remark - Exclude low ports to reduce false triggers on
access-list 143 remark client random ephemeral ports
access-list 143 deny   udp any any lt 1024
access-list 143 deny   udp any lt 1024 any
access-list 143 deny   tcp any any lt 1024
access-list 143 deny   tcp any lt 1024 any
access-list 143 remark - Devices allowed into the video queue are trusted
access-list 143 remark at source and mark IP Prec 3.   Adding "precedence flash"
access-list 143 remark to permits excludes unplanned usage of queue.
access-list 143 remark - H.225 RAS
access-list 143 permit udp any any eq 1719 precedence flash
access-list 143 permit udp any eq 1719 any precedence flash
access-list 143 remark - H.323 control traffic
access-list 143 permit tcp any any eq 1720 precedence flash
access-list 143 permit tcp any eq 1720 any precedence flash
access-list 143 remark - H.245 control traffic
access-list 143 permit tcp any any range 5555 5560 precedence flash
access-list 143 permit tcp any range 5555 5560 any precedence flash
access-list 143 permit tcp any any range 11000 11999 precedence flash
access-list 143 permit tcp any range 11000 11999 any precedence flash
access-list 143 remark - SCCP (skinny) traffic
access-list 143 permit tcp any any range 2000 2002 precedence flash
access-list 143 remark - Video & Audio RTCP traffics, MGCP (UDP:2427)
access-list 143 permit tcp any any range 2326 2331 precedence flash
access-list 143 permit tcp any range 2326 2331 any precedence flash
access-list 143 permit udp any any range 2327 2499 precedence flash
access-list 143 permit udp any range 2327 2499 any precedence flash
access-list 143 remark - Secure SCCP (skinny) traffic
access-list 143 permit tcp any any eq 2443 precedence flash
access-list 143 remark - Secure and non-secure SIP traffic
access-list 143 permit tcp any any range 5060 5061 precedence flash
access-list 143 permit tcp any range 5060 5061 any precedence flash
access-list 143 permit udp any any range 5060 5061 precedence flash
access-list 143 permit udp any range 5060 5061 any precedence flash
access-list 143 remark - Avaya IP telephony control signaling
access-list 143 permit tcp any any range 5010 5012 precedence flash
access-list 143 permit tcp any range 5010 5012 any precedence flash
access-list 144 remark * Gold class - Interactive Queue (IP Precedence 4)
access-list 144 remark - SQL*Net
access-list 144 permit tcp any any eq 66
access-list 144 permit tcp any eq 66 any
access-list 144 remark - Telnet
access-list 144 permit tcp any eq telnet any
access-list 144 permit tcp any any eq telnet
access-list 144 remark - Altel TN3270
access-list 144 permit tcp any any eq 1023
access-list 144 permit tcp any eq 1023 any
access-list 144 deny   tcp any any lt 1024
access-list 144 deny   tcp any lt 1024 any
access-list 144 remark - DLSw
access-list 144 permit tcp any eq 2065 any
access-list 144 permit tcp any any eq 2065
access-list 144 remark - SQL
access-list 144 permit tcp any any eq 1521
access-list 144 permit tcp any eq 1521 any
access-list 144 remark - Ford TN3270
access-list 144 permit tcp any any eq 5031
access-list 144 permit tcp any eq 5031 any
access-list 144 remark - Deny multicast and mark lightweight wireless CAPWAP control traffic
access-list 144 deny   udp any 224.0.0.0 15.255.255.255
access-list 144 permit udp any any eq 5246
access-list 144 permit udp any eq 5246 any
access-list 145 remark * Platinum class - Voice Queue (IP Precedence 5)
access-list 145 remark - Exclude low ports to reduce false triggers on client random ephemeral ports
access-list 145 deny   udp any any lt 1024
access-list 145 deny   udp any lt 1024 any
access-list 145 remark - voice RTP packets
access-list 145 permit udp any any range 16384 32767 dscp ef
access-list 145 permit udp any range 16384 32767 any dscp ef
access-list 146 remark * Internet class - Routing Queue (Trust Marking)
access-list 146 remark - Routing protocols: OSPF, EIGRP, BGP
access-list 146 permit ospf any any
access-list 146 permit eigrp any any
access-list 146 permit tcp any any eq bgp
access-list 146 permit tcp any eq bgp any
access-list 146 remark - Lightweight wireless CAPWAP data traffic
access-list 146 permit udp any any eq 5247
access-list 146 permit udp any eq 5247 any
access-list 148 remark * Mcast class - Interactive Queue (IP Precedence 1)
access-list 148 permit udp any 239.80.0.0 0.0.0.127
access-list 149 remark * IP_Video class - Video Queue (IP Precedence 4)
access-list 149 remark - Exclude low ports to reduce false triggers on client random ephemeral ports
access-list 149 deny   udp any any lt 1024
access-list 149 deny   udp any lt 1024 any
access-list 149 permit udp any any range 2326 2498 dscp af41
access-list 149 permit udp any range 2326 2498 any dscp af41
nls resp-timeout 1
cpd cr-id 1
snmp-server community B1K7PD3L RO 9
snmp-server ifindex persist
snmp-server trap-source GigabitEthernet0/2
snmp-server location addr Bldg Abbr"FCEFFI" (7007) Bldg name "Ford Finland & FORSO Address: Plaza Business Park Building Halo Ayritie 24 City: Vantaa Finland 01510
snmp-server contact name ="Jere Vastaki" Fordnet"7319152" Phone"358-9-35170152" CDS"jvastama"
snmp-server enable traps ospf state-change
snmp-server enable traps ospf errors
snmp-server enable traps ospf retransmit
snmp-server enable traps ospf lsa
snmp-server enable traps ospf cisco-specific state-change nssa-trans-change
snmp-server enable traps ospf cisco-specific state-change shamlink neighbor
snmp-server enable traps ospf cisco-specific errors
snmp-server enable traps ospf cisco-specific retransmit
snmp-server enable traps ospf cisco-specific lsa
snmp-server enable traps envmon
snmp-server enable traps bgp
snmp-server enable traps entity
snmp-server enable traps fru-ctrl
snmp-server enable traps hsrp
snmp-server host 19.97.9.134 fordsnmp
snmp-server host 19.97.9.135 fordsnmp
tacacs-server host 19.97.9.193
tacacs-server host 19.97.9.202
tacacs-server timeout 10
tacacs-server key 7 1335051718180D2E2223212734364315
control-plane
voice-port 0/0/0
voice-port 0/0/1
voice-port 0/0/2
voice-port 0/0/3
mgcp profile default
sccp local GigabitEthernet0/2
sccp ccm 19.106.182.15 identifier 2 version 7.0
sccp ccm 19.106.214.15 identifier 1 version 7.0
sccp
sccp ccm group 1
bind interface GigabitEthernet0/2
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 2 register CFB-FINGW001EU2
associate profile 1 register XCD-FINGW001EU2
dspfarm profile 1 transcode
description #XCODE dsp farm - G711/G729r8 only#
codec g711ulaw
codec g711alaw
codec g729r8
maximum sessions 30
associate application SCCP
dspfarm profile 2 conference
description Ad-hoc conference - G711 only
codec g711ulaw
codec g711alaw
maximum sessions 8
associate application SCCP
dial-peer voice 3501 voip
tone ringback alert-no-PI
description Incoming from PSTN/SIP provider
translation-profile incoming PSTN-to-PhoneDN
session protocol sipv2
incoming called-number .T
voice-class codec 1
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte digit-drop
ip qos dscp cs3 signaling
no vad
dial-peer voice 3601 voip
tone ringback alert-no-PI
description incoming from CUCM-SIP Trunk
session protocol sipv2
incoming called-number 9T
voice-class codec 1
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte sip-notify
ip qos dscp cs3 signaling
no vad
dial-peer voice 4501 voip
description Outgoing PSTN/SIP provider
translation-profile outgoing PhoneDN-to-PSTN
destination-pattern 9T
session protocol sipv2
session target ipv4:10.80.144.10
voice-class codec 1
voice-class sip early-offer forced
voice-class sip profiles 1
voice-class sip options-keepalive
dtmf-relay rtp-nte digit-drop
ip qos dscp cs3 signaling
no vad
dial-peer voice 4601 voip
description Outgoing CUCM-SIP Trunk to CUCM Subscriber 1
preference 1
destination-pattern 7319...$
session protocol sipv2
session target ipv4:19.106.214.15
voice-class codec 1
voice-class sip early-offer forced
voice-class sip profiles 1
voice-class sip options-keepalive
dtmf-relay sip-notify rtp-nte
ip qos dscp cs3 signaling
no vad
dial-peer voice 4602 voip
description Outgoing CUCM-SIP Trunk to CUCM Subscriber 2
preference 2
destination-pattern 7319...$
session protocol sipv2
session target ipv4:19.106.182.15
voice-class codec 1
voice-class sip early-offer forced
voice-class sip profiles 1
voice-class sip options-keepalive
dtmf-relay sip-notify rtp-nte
ip qos dscp cs3 signaling
no vad
dial-peer hunt 1
gateway
media-inactivity-criteria all
timer media-inactive 5
timer receive-rtp 1200
sip-ua
set pstn-cause 1 sip-status 503
set pstn-cause 3 sip-status 503
retry invite 2
retry bye 2
retry cancel 2
timers trying 550
g729-annexb override
gatekeeper
shutdown
banner motd ^C
WARNING! THIS IS A PRIVATE COMPUTER SYSTEM. USAGE MAY BE
MONITORED AND UNAUTHORIZED ACCESS OR USE MAY RESULT IN
CRIMINAL OR CIVIL PROSECUTION.
Except for some privacy rights granted by applicable law,
by signing on to the system you acknowledge:
You do not have any expectation of privacy in your use of the system.
You are familiar with, understand, accept, and will comply with the
provisions of Company Directive B-109.
^C
line con 0
exec-timeout 20 0
password 7 045307225A7364775E4A03
transport preferred none
line aux 0
exec-timeout 20 0
password 7 045307225A7364775E4A03
line vty 0 4
exec-timeout 20 0
password 7 060E0305191C2120524414
transport preferred none
transport input telnet
scheduler allocate 20000 1000
ntp server 19.211.4.1
ntp server 19.211.5.1
end
fingw001-vg3945#
fingw001-vg3945#

Proper amount of DSP resources in router?

Let's say we have a small site with 25 IP phones, 4 analog devices using a FXS module in the router and an ISDN PRI circuit using a T1 module in the router. I assume a PVDM3 module with 64 DSPs would be more than enough to handle that site? Does the FXS module have its own DSP resources and we don't need to figure those in? I figure the ISDN PRI would use up to 23 DSP resources and then what's left out of the 64 on that PVDM3 could be used for conferencing, media termination point and so on. Am I thinking about this correctly? Sound reasonable (64 DSPs for a site like that)?

I didn't know there was one....but I just found it. Thanks

MTP check box when cme integrates with ccm

I am reading a document abour cme and ccm integration.The following step indicates that we need to check MTP.
Step 3 Ensure that the CiscoCallManager network uses a media termination point (MTP). The MTP is required to provide DSP resources for transcoding and for sending and receiving G.729 calls to the Cisco CME 3.1 system. All media streams between Cisco CallManager and Cisco CME 3.1 must pass through the MTP because Cisco CME 3.1 does not support transcoding.
Questions:
1. So it means that if I use g729 on the wan between cme and ccm, I must check this one?
2. If I don't need transcode, i can uncheck this box right?
3. I notice as soon as check the box, call transfer button not work on ccm phone. Is this a bug or supposed to be?
Thanks

Restrictions on CME
* Cisco CallManager must use a media termination point (MTP), intercluster trunk (ICT) mode, and slow start.
* Codecs on all the VoIP dial peers of the H.450 tandem gateway must be the same.
* Only one codec type is supported in the VoIP network at a time, and there are only two codec choices: G.711 (A-law or
mu-law) or G.729.
* Transcoding is not supported.
* Codec renegotiation is not supported. For example, if an H.323 call that uses a G.729 codec is received by a Cisco CME system and is forwarded to a voice-mail system that requires a G.711 codec, the codec cannot be renegotiated from G.729 to G.711.

Add new type phones CUCM 4.2

Hello, i have work to buy new phones in organization. But 7900 not selling now, how i see. How i can add new types phones to my CUCM.
Thanks!

Hi Dmitry,
The "G" in the suffix of these models (like 7941G) stands for Global. The
"GE" in the 7941G-GE stands for "Gigabit ethernet" as it is a 10/100/1000
model.
Keep in mind that there are 7900 models like the 7945G/65G and
7942G/62G that are not End of Sale or End of Life and can be added to
CCM 4.2(3) with the installation of a Device Pack like the one shown below;
Firmware Versions
Cisco Unified CallManager Device Package 4.2.3(60.0) includes the following firmware versions:
Firmware
4.2(3)Devpack   60
Analog Access   WS-X6624
A00204000013   / A0034322
Digital Access   WS-X6608
D00404000032   / D0054322
Conference Bridge WS-X6608
C00104000003   / C002E031
Media Termination   Point WS-X6608
M00104000006   / M002E031
ATA18X
ATA030204SCCP090202A
7915-12 Button   Extension Module
B015-1-0-3
7915-24 Button   Extension Module
B015-1-0-3
7916-12 Button   Extension Module
B016-1-0-3
7916-24 Button   Extension Module
B016-1-0-3
IP Phone 7902g
CP7902080002SCCP060817A
IP Phone 7905g
CP7905080003SCCP070409A
IP Phone 7906G/11
SCCP11.8-4-4S
IP Phone 7912g
CP7912080003SCCP070409A
IP Phone 7920
cmterm_7920.4.0-03-02
IP Phone 7921g
CP7921G-1.3.2
P Phone 7925g
CP7925G-1.3.2
IP Phone 7935
P00503021900
IP Phone 7936
cmterm_7936.3-3-20-0
IP Phone 7937
apps37sccp.1-3-3-0
IP Phone 7941/7961
SCCP41.8-4-4S
IP Phone   7941G/7961G
SCCP41.8-4-4S
IP Phone 7942/7962
SCCP42.8-4-4S
IP Phone 7945/7965
SCCP45.8-4-4S
IP Phone 7970/7971
SCCP70.8-4-4S
IP Phone 7975
SCCP75.8-4-4S
IP Phone 7985
cmterm_7985.4-1-7-0
IP Phone 7960
P00308010100
IP Phone 7940
P00308010100
7914 14-Button Line   Expansion Module
S00105000400
http://www.cisco.com/web/software/282074299/26745/ciscocm.4-2-3-DevPack-60_readme.html
Available via this path;
Downloads Home
Products
Voice and Unified Communications
IP Telephony
Unified Communications Platform
Cisco Unified Communications Manager (CallManager)
Cisco Unified CallManager Version 4.2
Unified Communications Manager/CallManager Device Packages-4.2(3.60)
Cheers!
Rob
"Clocks go slow in a place of work
Minutes drag and the hours jerk"
-The Clash

Conferencing problems...please help

I am using CCM 4.0 and i was having problems conferencing two calls. My setup is as such that i am using G.711 for local extension and G.723 for external extensions. For this i have created two regions for local and external gateway. What happens is that if i make a call from my cisco soft phone and then try to conference with another external call i am unable to do. These external calls are over the internet so using G.711 is not feasible. If i conference internal calls i am able to successfully do this. I found out that i will need to use hardware transcoders to fix this problems since the software conference bridge only allows G.711. Can anybody verify this or if there are alternative solution for this? Also are there any software conference bridges available which i can use to solve this problem.
Thanks

Maybe if i wrote down the configurations you could figure out what i am missing out.
Region setting for R_Conference:
Default --> G.711
International --> G.711
R_Conference(within this region) --> G.711
Region setting for International:
Default --> G.711
International(within this region) --> G.723
R_Conference --> G.711
Device pool settings:
Device pool name --> DP_Conference
Region --> R_Conference
Media Resource Group List --> MRGL
Conference bridge settings:
Conference Bridge Name --> CFB_CCM
Device pool --> DP_Conference
Media Resource Group Configuration:
Media Resource Group Name --> Intl_MRG
Selected Media Resource --> CFB_CCM(CFB)
--> MTP_CCM(MTP)
Media Resource Group List Configuration:
Media Resource Group List Name --> MRGL
Selected Media Resource Groups --> Intl_MRG
Media Termination Point Configuration:
Media Termination Point Name --> MTP_CCM
Device Pool --> DP_Conference
Phone Settings:
Device Name --> Test_Phone
Device Pool --> DP_Conference
Phone call steps:
1. I make a call to a 18004633339
2. I press the conference button
3. I dial another number 1800xxxxxx
4. I press the start conference button
5. Both the call disconnect

Media Termination Point

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