Translation pattern to meeting place trunk.

Hey Guys,
Have a sip trunk from Service Provider.. When people from PSTN dial a number it should land onto our meeting place bridge. I am able to land it to any dn on call manager thru translation pattern. But how can i land the call to meeting place. ?
Thanks
VR

So if that's the only thing they offer, you're going to need a transcoder or get them to offer G.711ulaw/alaw. Another option would be getting the MeetingPlace server to advertise G.729. It looks like that might be possible if you put it in High Quality audio mode- http://docwiki.cisco.com/wiki/Cisco_Unified_MeetingPlace_Release_8.5_--_Planning_the_Capacity_of_your_Cisco_Unified_MeetingPlace_System

Similar Messages

Meeting Place and Webex Call Back Failing

Hi
Meeting place 8.5 with WebEx scheduling, SIP trunk to CUCM 8.5.
Calling in using the specified number configured in call manager completes over the SIP trunk however call back from webex fails.
Error:
WebEx call back failed; no answer.
Problem is I never recieve a call. I have enabled Webex call backs in webex.
Seems like there is an issue on either MP or CUCM.
Just looking for some ideas as I am stumped.

This turned out to be a few things:
1) I needed to configure a SIP proxy to CUCM from meeting place. This was the CUCM IP address and TCP port number.
2) The SIP trunk security profile needed to be reconfigured for this trunk.
3) E164 was being used so I created a translation pattern to strip the + for the CSS associated with the RP.
A few things learned along the way.
1) The "activity" command ran from a SSH prompt with allow you to simulate a call from meeting place. Option 4, enter the number to dial and "f" for port.
2) In meeting place - Services -Logs - System Information Capture - Loading the index file and under the CCA SIP LOG is the error being returned by CUCM. In my case it was Warning: 399 CSLCCM1 "Unable to find a device handler for the request received on port
3) Command "cptrace" again from the SSH prompt will allow you to view the digits being sent from webex to meeting place.
I hope this information helps someone...

Avaya - CUCM -Meeting Place

Hi,
The current integration is between Avaya and CUCM using SIP Trunk, calling from Avaya to Cisco is working fine, however when the calls originate from the Avaya to Meeting place which uses the SIP trunk between Avaya and CUCM the call does not connect. From the logs I can see that 503 Service unavailable which normally means that the DN is not registered to CUCM.
Is there any specific configuration for integration of Avaya with Meetingplace using CUCM.
Thanks,
Vinay

First verify that the Avaya system is passing the MP number via the SIP trunk to CUCM. Then checked what number the Avaya system is passing and if that number is routed to MP. The Avaya system may be translating the MP phone number so it is not being reconized on the CUCM system.

Translation pattern not matching

Hello All
I am configuring a cucm 4.2 (yes i know its obsolete) integration with Lync 2010 and am having issues with a translation pattern.
The Lync server is sending me 86xxxxxxxxxx for calls within china and will send 61xxxxxxxx for australia (strips the +)
I have configured a [^86]! which should match any international numbers (other than China) and be prefixed and sent to the gateway. Here is the wiered thing I can dial +44xxxxxxxxxx using my lync client which proves that this is matching (when i delete the translation the call will fail).
But when i dial a number like +61xxxxxxxx it doesnt get through and i get
Cisco CallManagerDigit analysis: match(pi="1",fqcn="", cn="removedbymyself", plv="5", pss="LYNC:PT_Reception", TodFilteredPss="LYNC:PT_Reception", dd="61xxxxxxxx ",dac="0")
Cisco CallManagerDigit analysis: potentialMatches=NoPotentialMatchesExist" on the traces.
The LYNC partition has the translation rules. and the CSS assigned to the sip trunk has access to it. the CSS configured in the translation rules is the also the one assigned to the sip trunk.
Anyone see this sort of thing before? how can i check if there is another transformation taking place?
Only way i get round is to put a translation patter for " ! " and it works for all international calls.
Thanks,

Hi,
Have you tried testing the call with Dialed Number Analyzer? I find that's a fantastic and often-overlooked tool for this kind of issue. If DNA shows the call will not route, it's probably a CSS issue for the Stafford gateway. If DNA shows the call will route, then it's probably a dial-peer issue on the Stafford gateway.
-Jameson

Need Urgent Help on Meeting Place Integration with CUCM 7.1 and AS5400 PSTN Gateway

Hi,
This is first time I am on this forum.
I have already tried going through a lot of docs on docwiki.cisco.com but couldn't find complete configuration help.
I have to integrate Meeting Place 8.X with an existing CUCM and an E1 gateway (PSTN Gateway) AS5400.
The CUCM is already part of a Telepresence Environment. I need to create a SIP trunk between AS5400 and CUCM 7.1 and then create a Trunk between AS5400 and Cisco Unified MP 8.X and then between CUMP and CUCM.
I need help on AS5400 SIP Configs as well as parameters I need to cover on CUCM (Though I have done some basic dial-peer configs but they haven't been of much help).
Then I also need help on AS5400 SIP configs with CUMP 8.0
Any docs on Integration between CUMP and TP3000 will be of great help too.
Rgds,
Asim

I can get the Ricoh to register as sip endpoint, it answeres then imediatly disconnects. Doing a monitor with Wireshark looks like it attempts to negotiate t38 but fails. Any idea why this fails?
|160.260684000|         INVITE SDP (g711U)            |                   |SIP From:
|         |(5060)   ------------------> (5060)   |                   |
|160.338806000|         INVITE SDP (t38)              |                   |SIP Request
|         |(5060)   <------------------ (63435) |                   |
|160.339545000|         491 Request Pending           |                   |SIP Status
|         |(5060)   ------------------> (5060)   |                   |
|160.547894000|         406 Not Acceptable            |                   |SIP Status
|         |(5060)   <------------------ (63435) |                   |

Need help Creating a translation pattern that adds dial out digits to incoming calls

I came across an article yesterday and it showed the steps how to fix Missed Call/Received Call numbers so that you can dial them from the menu correctly (auto-add a 9, etc.)?
I tried it this morning and came up with this translation pattern:
voice translation-rule 6
rule 1 /^201\(.*\)/ /8\1/
rule 2 /\(..........\)/ /81\1/
voice translation-profile filter_Incoming
translate calling 6
This translation pattern rule 1 adds the dial out character 8 and strips 201 for local calls. Rule 2 adds dial out character 8 and adds 1 for long distance. The purpose of this translation rule is when the ephone receives the phone call the characters 8 and 1 are added so when you quickly need to redial you do not have to edit the number and add 8 for each call.
I tested the translation-rule:
ROUTER-2911#test voice translation-rule 6 9082121231
Matched with rule 2
Original number: 9082121231     Translated number: 819082121231
Original number type: none      Translated number type: none
Original number plan: none      Translated number plan: none
ROUTER-2911#test voice translation-rule 6 2019121231
Matched with rule 1
Original number: 2019121231     Translated number: 89121231
Original number type: none      Translated number type: none
Original number plan: none      Translated number plan: none
ROUTER-2911#
Issue is I am not sure with my inbound call leg if it can even work. We dial out through the SIP Trunk and the incoming is translated to the AutoAttendant on Cisco Unity Express.
voice translation-rule 1
rule 1 /2015552100/ /2003/
voice translation-profile CUE_Voicemail/AutoAttendant
translate called 1
dial-peer voice 9 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming CUE_Voicemail/AutoAttendant
call-block translation-profile incoming BLOCKED-INCOMING
call-block disconnect-cause incoming call-reject
session protocol sipv2
session target dns:nd01-04.fs.SIPPROVIDER.net
incoming called-number .%
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
Can what I am trying to do be done with my current setup?

Hi patldmart012,
The dial-peer 9 that you have attached will not be affected by following config
voice translation-rule 6
rule 1 /^201\(.*\)/ /8\1/
rule 2 /\(..........\)/ /81\1/
voice translation-profile filter_Incoming
translate calling 6
Because you have not applied the translation profile "filter_incoming" on the dial-peer.
Could you please provide the exact call flow?
Along with that, If you are facing issue with calls on SIP Trunk, please collect following debugs in a logging buffer and attach the file. I will analyse it and will get back to you.
debug voip ccapi inout
debug ccsip message
debug voice translation
Debug h225 asn1 (If H323 involved)
Debug h245 asn1 (If H323 involved)
Debug MGCP Packets (If MGCP involved)
Also provide the running config of the GW.
These are verbose debugs, so please collect them in the following manner:
Router(config)# service sequence
Router(config)# service timestamps debug datetime msec
Router(config)# logging buffered 30000000 7
Router(config)# no logging con
Router(config)# no logging mon
Router# Clear log
Router# term no mon
<Enable debugs, then wait for issue to occur.>
Router# term len 0
<Enable session capture to txt file in terminal program.>
Router# Undebug all
Router# sh log
Once i have the logs, i will analyse it and will get back to you.
Regards,
Mudit Mathur

Meeting Place Configuration Doc

Hi
I am configuring Meeting Place 8.5. Can anyone please provide me Meeting Place 8.5 basic configuration guide. I have the cisco guide with me. I have created sip trunk in cucm and done basic configuration in Meeting Place server as per cisco guide. But when I test calling meeting place number, the call gets disconnected. Something somewhere I have missed. I have successfully imported the users to the meeting place server. Thank u.

Hi,
Are you using EMS or HMS? Can you make sure you have correct configuration on SIP trunk (CUCM side) and make sure on MP you have configured the correct CUCM configuration under SIP config.
Login to your MP as root and see check below:
1. cd /var/mp
2. ls -l
and see if you see cca directory or not? If not then looks like you configured the MP for HMS but you don't have hardware MCU or possibly MCU is not online or not configure properly.
If you see above directory do below:
cd /var/mp/cca
tail -f tvsip* | grep -e '--'
make a test call and you will see sometihng on your screen, capture the whole stuff and post here will look tell you what's going on but if you don't see any calls coming that means call is not even landing to MP then check your CUCM for possibly PT, CSS, RP, RP, Trunk etc.
Please let me know if you need any more info.
If you need immediate assistance kindly open a TAC case.
HTH
Arun

Meeting Place Queries

Hi,
   I am having CCM(8.5) in cluster and meeting place,i have 2 Voice Gateways for my company,Currently Meeting Place runs on 1 Gateway which is used for calling and the other Voice Gateway is also usedfor calling but for other Office in same company,
   question is :-The Other Gateway which is used by OTHER OFFICE OF SAME COMPANY CAN IT BE USED FOR CONFIGURING OTHER NUMBER OF MEETING PLACE(as a back up MEEITNG PLACE) number??
         Any Help will be appreciated

Hi Yogesh,
MP does not perform digit analyisis. In other words, it will honour an INVITE sent to it regardless the number.
Be aware that MP is expected to be deployed on SIP trunk with CUCM.
hth

Recommended CM System Speed Dials: Translations Patterns vs. DNs

Using Call Manager 8.6, I've successfully tested System Speed Dialing through using both Translation Patterns and as Directory Numbers (using the Call Forward Destination field). Since using Translation Patterns DOES NOT allow for a alpha display name to the user dialing the speed dial, I tested the DN approach. This offered about 1-2 seconds of a display on the phone so the user could get a confirmation that they dialed the speed dial they desired. Is there any reason to not use DNs as system speed dials? Do they use up resources in some way that Translation Patterns do not? I'm just wondering if there is a compelling reason to not use the DN approach. Otherwise, I think this will be customer's avenue to using system speed dials as it provides that 1-2 second of alpha display of the speed dials name. Thank you. Dan

No, they do not use any more resources. The only reason I am hesitant of using them is that unassigned DNs can be accidentally purged out by unaware administrator, as one can simply find unassigned DNs thinking those are old discarded ones and click to delete them. If you have a process in place that prevents this there is absolutely no down side.
HTH,
Chris

How to change host name and domain name in installed Cisco Meeting Place Express 2.0?

Hi
I have Cisco Meeting Place Express 2.0 installed on server. Meting Place uses Linux (Red Hat) as base operation system. Please tell me how I can change host name and domain name for Meeting Place Express application to integrate it into corporate domain infrastructure.
Thanks in advance!!!
Sincerely yours
Sergey Bondarenko

Hi
Thanks for your attention for my humble problem.
I have performed Meeting Place Express configuration through the “net” command.
But now unfortunately I can't connect to meeting place through web browser. I just get blank page.
Though I can reach the server with help of ssh. I think my DNS works properly because I can resolve the ip address of cumpxvoice .uclab.com and cumpxweb. uclab.com with help of dig command.
Have you any idea?
A lot of thanks in advance!!
Sincerely yours Sergey Bondarenko
Here I have printed the net configuration output of Cisco meeting Place Express:
[root@cumpxvoice root]# net
This is the MeetingPlace Express network configuration utility.
Up to the point where you save changes, it is safe to hit ^C to get
out at any time, in which case nothing will have changed.
1) List current configuration
2) Configure Ethernet port 1
3) Configure Ethernet port 2
4) Configure service bindings
5) Set host names
6) Set domain name
7) Configure DNS service
8) Configure NTP service
9) Configure routing
10) Done
Select: 1
Primary/audio host name: cumpxvoice
HTTP host name:          cumpxvoice
RTMP host name:          cumpxweb
Domain name:             uclab.com
Service bindings:
Audio:                   port 1 (eth0)
Web browsing (HTTP):     port 1 (eth0)
Web conferencing (RTMP): port 2 (eth1)
Port 1 (device eth0):
IP Address:      10.62.60.116
Network mask:    255.255.255.0
Default gateway: 10.62.60.1
Link setting:    autoneg on
Port 2 (device eth1):
IP Address:      10.62.60.117
Network mask:    255.255.255.0
Default gateway: 10.62.60.1
Link setting:    autoneg on
Static routes:
Destination      Gateway          Network mask     Port
0.0.0.0          10.62.60.1       0.0.0.0          1
DNS Servers:
10.62.60.115
10.0.1.4

Time issue while scheduling meeting in meeting place 8.0

Dear Team,
I am very new to Meeting Place system. Here we have version 8.0 and i am facing below issue.
While scheduling meeting when i have mentioned start time as 13.30 and end after 30 min.
but the issue is user can join the meeting before 13.30 and thats is security issue
i would like to know how to restict this. my requiremnt is user can only join before 10 min.
Regards
Hari

Hi Hari,
Is you MeetingPlace 8.0 integrated with WebEx, or it is an Audio Only solution?
If you are scheduling meetings in Audio Only solution, you can specify "Early meeting start (minutes)" variable in System Configuration > Meeting Configuration.
If you have WebEx integration and WebEx scheduling, you would need to specify a default "Join before host", but every host would have to configure that for their own profile.
-Dejan

Time issue iwhile scheduling meeting in meeting place 8.0

Dear Team,
I am very new to Meeting Place system. Here we have version 8.0 and i am facing below issue.
While scheduling meeting when i have mentioned start time as 13.30 and end after 30 min.
but the issue is user can join the meeting before 13.30 and thats is security issue
i would like to know how to restict this. my requiremnt is user can only join before 10 min.
Please help

Hi Hari,
Is you MeetingPlace 8.0 integrated with WebEx, or it is an Audio Only solution?
If you are scheduling meetings in Audio Only solution, you can specify "Early meeting start (minutes)" variable in System Configuration > Meeting Configuration.
If you have WebEx integration and WebEx scheduling, you would need to specify a default "Join before host", but every host would have to configure that for their own profile.
-Dejan

Meeting Place 8.0 Audio Recording play back

Hi,
We are currently using Meeting place 8.5, it is currently installed as Express Media server (Single server solution) as audio only conferencing solution.
I am able to record a meeting using meetingplace conference manager, but there is no option to find/play the recording, when i select the meeting and check the meeting details, I can find that the meeting is recorded but Attachment is displayed as No.
I found links which says to find the meeting using Meeting place web scheduler and find meetings but is if no avail as there is not web server installed in my setup.
Would like to if there is any way to find/play the recording.
Thanks,
Vinay

Hi Vinay,
Accessing or playback of recordings requires a Web Scheduling server to manage and convert the recording from the Application server. This is not possible with just an applicaiton server alone.
Thanks,
Derek Johnson
Conferencing TAC

Meeting Place Express 2.1.1.2 Web Access Issue

Hello all,
I'am having an issue with a Meeting Place Express 2.1.1.2 server. There was a power failure and now we can still access the server via dial-in but the web acess is not working. From the CLI i run the mpx_sys status command and it says webservices is active. The server has been restarted. When I check the errorlog I see "unable to connect to webservices, trying again" Anyone know what the issue might be? Thanks
Jason

Hello Hasnain,
Have you tried the recovery procedures below:
http://docwiki.cisco.com/wiki/Cisco_Unified_MeetingPlace_Express%2C_Release_2.x_--_Troubleshooting_System_Issues#Recovering_the_Password_for_the_root_Account
If that doesnt work, you will need to open a TAC case and TAC will send you a password recovery .iso file to burn on a CD/DVD.
Regards,
Victor Dang.

Meeting Place 8.5.5.14 to 8.6.1.2 License

Hello,
Do I need a new license after I've upgrade my meeting Place from 8.5.5.14 to 8.6.1.2 ?
In the upgrade document, there is no mention that we need the license (not yet).
Regards,

Hi,
There is no license change between MeetingPlace 8.5 and 8.6 versions. The same license stays in place.
Kind regards,
-Dejan

Translation pattern to meeting place trunk.

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