SRST maximum on CUCM

I cannot find anything that tells me if there is a maximum number of gateways allowed to hang off CUCM, does anyone know if there is a maximum? We currently have 4 sites that are SRST enabled and we may have 2-3 more in the near future and I want to make sure we are not reaching a capacity issue or crash our CUCM, we are currently running 6.1.2 and plan to migrate to 6.1.3 then 7.1.3 soon.
Thanks.
John

So, your question is about is there a limit to the number of gateways you can register to CUCM? OR is your question more along the lines of is there a limit to the number of SRST-enabled users across the cumulative count of gateways in my environment?
For the first, I don’t know of a set limit of gateways. I have built CUCM clusters for school systems where every school has a single or even two gateways in addition to centralized gateways - so well over a 100 gateways configured without issue. If there is a hard limit, I am not aware of one.
As for the second, SRST maximum is not cumulative across devices. It is per device and is based on hardware, SRST version, and what you're licensed for. For example, a 3845 can support more SRST users than a 2801 but the 2 are independent of each other because once the devices are in SRST mode, they lose connection to CUCM (as you already know) and would have no direct IP / call setup path between them. They operate independently until the CUCM comes back online.
Hailey
Please rate helpful posts!

Similar Messages

COR in SRST

Dears,
I have 2 question please answer
voice register pool 1
cor list incoming International 1 2001,2022,2403,2504,2705,2807,2999,2367
I cannot add more than 8 DN in one COR list as shown above and we have only 4 no's of COR list option so how i can accommodate my corporate DN in SRST mode. Once it falls to SRST mode those DN which are not listed in the COR list they get full privilege to call so it is hammering me.
SRST Fallback to CUCM takes around 5 min (when CM comes up)
Thanks

A quick suggestion regarding your second query. There is a setting called "Connection Monitor duration" on the device pool of the IP phone on CUCM.
This setting defines the time that the Cisco Unified IP Phone monitors its connection to Cisco Unified Communications Manager before it unregisters from SRST and reregisters to Cisco Unified Communications Manager.
To use the configuration for the enterprise parameter, you can enter -1 or leave the field blank. The default value for the enterprise parameter equals 120 seconds.
Change this setting if you need to disable the connection monitor or if you want to extend the connection monitor time. The maximum number of seconds that you can enter in the field equals 2592000.
HTH
Manish

Ask the Expert: Upgrading Cisco Unified Communications Manager (CUCM) to Version 9.1 (Drive to 9)

Welcome to the Cisco Support Community Ask the Expert conversation. Learn from experts Vijay Rao and Amit Singh about simplified upgrade process and focused support from Cisco to migrate to version 9.1.
This is a continuation of the live Webcast
Drive to 9 is a comprehensive and holistic program designed to help you upgrade the current Cisco® Unified Communications Manager installed base to version 9.1 or higher. This upgrade will enable customers to have next-generation collaboration experiences.
During the live event, Cisco subject matter experts Vijay Rao and Amit Singh focussed on the simplified upgrade process and focused support from Cisco to migrate to version 9.1. They also talked about the changes made to the licensing model of User Connect Licensing and Cisco Unified Workspace Licensing.
Vijay Rao is a Network Consulting Engineer and is currently a unified communications (UC) consultant for Bank of America. He has been providing consulting assistance to the bank for the past 6 years. He helps design complex UC networks for large enterprise customers. He was previously part of Cisco IT in the Asia Pacific, Japan, and China (APJC) region and was instrumental in designing and implementing the Bangalore campus. He has been working with Cisco for 9 years and has 12 years of UC experience. He has a Cisco CCVP® certification.
Amit Singh is a customer support engineer at the Cisco Technical Assistance Center in Bangalore, India. He has 7 years of experience in his areas of expertise: wireless, Cisco Unified Communications Manager, multiservices, Cisco Unity®, and Cisco Unified Contact Center Express. He has been involved in various escalation requests from India, Singapore, and Australia and is currently working as a technical lead for the Voice team in Bangalore, India. He is a computer science graduate.
Remember to use the rating system to let Vijay and Amit know if you have received an adequate response.
Vijay and Amit might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation on the Collaboration, Voice and Video sub-community forum shortly after the event. This event lasts through July 19, 2013. Visit this forum often to view responses to your questions and the questions of other community members.
Webcast related links:
Webcast Video
FAQ from the live webcast
Slides from the live webcast

Hello Robert,
Apologies for a delayed response, some days get very hectic.
In CallManager, we only define the SRST reference, and CUCM version and SRST version are independent of each other.
The only thing, which is related and will change with CUCM upgrade is Phone F/w version.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/compat/ccmcompmatr1.pdf
You may just want to check your, phone f/w compatibility with the SRST version running on your ISR G1 Gateways:
http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_device_support_tables_list.html
For Example: SRST version 7.1
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps2169/data_sheet_c78-520521.html
You may want to do some lab testing with CUCM 9.1 and an SRST supported f/w on your phones.
If you decide to run the old Phone/F/w to support the SRST version, you may not be able to take advantage of new features.
Also, you can try and upgrade your phones(Wih CUCM 9.1) and test them with your SRST version.
It should work fine, but from a troubleshooting perspective, TAC may request you to come into a Cisco Supported combination.
Please, let me know if this clarifies your doubt or we can have a quick phone call.
Regards
Amit Singh

SRST Incoming digits

Hi,
I have an issue where i see the incoming digits show up little different than normal. My incoming calls from the duct to the IP phone registered to SRST gateway does not come properly. But when we turn off the srst the calls work normal.
I am sharing the Debud isdn q931 of the same. Please let me know if some one came across the similar issue.
The Debug when in SRST ON:
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0036
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18382
                Preferred, Channel 2
        Facility i = 0x9FAA068001008201008B0100A1100201010201553008820301B850860101
        Facility i = 0x9FAA068001008201008B0100A11802010202010080102020566F696365205365727669636573
        Progress Ind i = 0x8183 - Origination address is non-ISDN
        Calling Party Number i = 0x0181, '8026'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0x89, '30123'
                Plan:Private, Type:Unknown
        User-User i = 0x00FE, 'U', 0x0100, 'Y', 0x0100B00C060981, '8026', 0x0F1282818E, ' Voice Services'
        Shift to Codeset 4
        Codeset 4 IE 0x31 i = 0x80
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8036
        Channel ID i = 0xA98382
                Exclusive, Channel 2
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x8036
        Cause i = 0x8081 - Unallocated/unassigned number
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x0036
        Cause i = 0x8181 - Unallocated/unassigned number
        User-User i = 0x00FEB0
Aug 20 03:29:56: ISDN Se0/2/0:15 Q931: RX <- SETUP pd = 8 callref = 0x3F6A
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA1838D
                Preferred, Channel 13
        Facility i = 0x9FAA068001008201008B0100A1100201010201553008820301B850860101
        Facility i = 0x9FAA068001008201008B0100A113020102020100800B5A6F72696E612054657374
        Progress Ind i = 0x8183 - Origination address is non-ISDN
        Calling Party Number i = 0x0181, '68152'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0x89, '30123'
                Plan:Private, Type:Unknown
        User-User i = 0x00FE, 'U', 0x0100, 'Y', 0x0100B00C070981, '68152', 0x0F0D828684, 'ZorinaTest'
        Shift to Codeset 4
        Codeset 4 IE 0x31 i = 0x80
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: TX -> SETUP_ACK pd = 8 callref = 0xBF6A
        Channel ID i = 0xA9838D
                Exclusive, Channel 13
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: RX <- INFORMATION pd = 8 callref = 0x3F6A
        Called Party Number i = 0x89, '0'
                Plan:Private, Type:Unknown
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: RX <- INFORMATION pd = 8 callref = 0x3F6A
        Called Party Number i = 0x89, '1'
                Plan:Private, Type:Unknown
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0xBF6A
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: TX -> ALERTING pd = 8 callref = 0xBF6A
        Facility i = 0x9FAA06800100820100A11D020101060528EC2C000180115A6F72616E2053746566616E6F76736B69
        Progress Ind i = 0x8088 - In-band info or appropriate now available
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: RX <- FACILITY pd = 8 callref = 0x3F6A
        Facility i = 0x9FAA06800100820100A406020101810101
Aug 20 03:29:59: ISDN Se0/2/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x3F6A
        Cause i = 0x8190 - Normal call clearing
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0036
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18382
                Preferred, Channel 2
        Facility i = 0x9FAA068001008201008B0100A1100201010201553008820301B850860101
        Facility i = 0x9FAA068001008201008B0100A11802010202010080102020566F696365205365727669636573
        Progress Ind i = 0x8183 - Origination address is non-ISDN
        Calling Party Number i = 0x0181, '8026'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0x89, '30123'
                Plan:Private, Type:Unknown
        User-User i = 0x00FE, 'U', 0x0100, 'Y', 0x0100B00C060981, '8026', 0x0F1282818E, ' Voice Services'
        Shift to Codeset 4
        Codeset 4 IE 0x31 i = 0x80
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8036
        Channel ID i = 0xA98382
                Exclusive, Channel 2
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x8036
        Cause i = 0x8081 - Unallocated/unassigned number
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x0036
        Cause i = 0x8181 - Unallocated/unassigned number
        User-User i = 0x00FEB0
The below is when the SRST is OFF :
Aug 20 03:29:56: ISDN Se0/2/0:15 Q931: RX <- SETUP pd = 8 callref = 0x3F6A
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA1838D
                Preferred, Channel 13
        Facility i = 0x9FAA068001008201008B0100A1100201010201553008820301B850860101
        Facility i = 0x9FAA068001008201008B0100A113020102020100800B5A6F72696E612054657374
        Progress Ind i = 0x8183 - Origination address is non-ISDN
        Calling Party Number i = 0x0181, '68152'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0x89, '30123'
                Plan:Private, Type:Unknown
        User-User i = 0x00FE, 'U', 0x0100, 'Y', 0x0100B00C070981, '68152', 0x0F0D828684, 'ZorinaTest'
        Shift to Codeset 4
        Codeset 4 IE 0x31 i = 0x80
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: TX -> SETUP_ACK pd = 8 callref = 0xBF6A
        Channel ID i = 0xA9838D
                Exclusive, Channel 13
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: RX <- INFORMATION pd = 8 callref = 0x3F6A
        Called Party Number i = 0x89, '0'
                Plan:Private, Type:Unknown
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: RX <- INFORMATION pd = 8 callref = 0x3F6A
        Called Party Number i = 0x89, '1'
                Plan:Private, Type:Unknown
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0xBF6A
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: TX -> ALERTING pd = 8 callref = 0xBF6A
        Facility i = 0x9FAA06800100820100A11D020101060528EC2C000180115A6F72616E2053746566616E6F76736B69
        Progress Ind i = 0x8088 - In-band info or appropriate now available
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: RX <- FACILITY pd = 8 callref = 0x3F6A
        Facility i = 0x9FAA06800100820100A406020101810101
Aug 20 03:29:59: ISDN Se0/2/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x3F6A
        Cause i = 0x8190 - Normal call clearing

Hi,
This is an MGCP gateway. We are testing the SRST configs. When we apply access list to block the route to the ccm, then we are treating as SRST ON and when we remove the block commands to the CUCM in Gateway we assume that it is SRST OFF.
Now when the SRST is OFF, CUCM is waiting untill all the digits are received. However in SRST we are only receiving the first 5 digits and not the complete digits.
Not sure why we are receiving the other two digits seperately. when the gateway is registering with CCM. When the gateway is in SRST, it does not get the other two digits. In our example it is 0,1 that we are getting seperately when GW registered with CCM.

SRST licenses

Hi,
I am in process to design a VOIP solution. I am confuse on one point.
Customer has single site with 50 VOIP users and 15 analog users. I am quoting 2921 Voice Gateway with SRST. For analog devices, we are proposing single VG224. How many total SRST licenses are suppose to include in router configuration?
IP Phones + Analog Phones = 65 SRST licenses since analog phones will also register with CUCM
OR
IP Phones + VG224 = 50 +1 =51 SRST licenses since CUCM sees only one VG224 connected to it
I will appreciate quick reply, thanks in advance.

Hello
Please refer to this, This might help you.
https://supportforums.cisco.com/discussion/11562256/srst-license
https://supportforums.cisco.com/discussion/10226346/srst-licensing
Br,
nadeem

SRST How to Documentation CUCM Version 7.1.5.34900-7

Hello:
I am looking for some really good documentation on how to setup SRST and how to enable it when a internet/network circuit is down but the router is there on site.
Thanks
Cathy

Hi Cathy,
Please see the link below:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesrst.html
It is one of the chapter of
Cisco Unified Communications Manager Express System Administrator Guide
Hope this helps.

CUCM 8.6.2 maximum rings

Hi all.
We currently have an alarm system at one of our sites that is triggered by calling an analog port that the alarm is plugged into, and then the alarm rings only as long as the call is ringing. Since there is a maximum ringing time on the system, it eventually times out. The business is asking for an alternative that would allow it to ring indefinitely. I can't raise the global Max Ring Time, so can anyone suggest an alternative?
This hardware solution they're using is obviously quite inelegant, but we're trying to accomodate.

Surprisingly, it has made it all the way to 10.5(x) with the same info and the same error...
I did found a method to change it via root access, and you might not require root access, but I can't tell for sure as I would need to look at exactly what the contents of the file that TAC changes, but apparently it's just the platformConfig.xml that they need to change and reboot.
If that's the case, using the utils import config using pretty much all the same info, except the country, would end up with the same outcome.
Again, not 100% sure but theory says that should do the trick, you can run that thru TAC if you open the case and see what they think about it.

Issues with Multicast MOH with CUCM SRST Sites

Hi there,
I am having an issue streaming multicast music on hold at remote sites by sourcing the music on hold from the local flash of the Cisco 2951.
I am running CUCM 9.1(2) SU1, I have 1 x PUB and 1 x SUB. I have enabled the PUB to stream unicast MOH and have enabled multicast MOH on the SUB. Base IP: 239.1.1.1 and Port 16384.
I have set the MOH Region to all remote sites regions to G.711 and have put the SUB into an MRG and have enabled MOH at the MRG and Audio Source Level as per documentation. The MRGL for remote sites contain the local sites hardware resources such as MTP, CFB and XCODE. Second to that MRG, I have set the MOH_MRG containing the SUB enabled for multicast.
When a remote site user put the PTSN caller on hold, I can see an active multicast music on hold session active on the local router but the PSTN caller hears silence !!! The number of In/Out packet on the multicast session does not increase at all.
Could you please help me guys? I have tried all options and still I am getting silence all the times.
Below is the configuration on the local gateway at remote sites:
XX-XXX-VG01#sh run | s call-m
call-manager-fallback
max-conferences 4 gain -6
transfer-system full-consult
ip source-address 10.114.80.1 port 2000
max-ephones 58
max-dn 120
transfer-pattern 031580....
keepalive 10
call-forward pattern 031580....
moh "music-on-hold.au"
multicast moh 239.1.1.1 port 16384 route 10.114.80.1 1.1.1.1
time-zone 29
time-format 24
date-format dd-mm-yy
Please note that when the PSTN caller is on hold, when I issue the # show ccm-manager music-on-hold, I get the following output all the times:
XX-XXX-VG01#sh voice call status
CallID     CID ccVdb      Port        Slot/DSP:Ch Called #   Codec    MLPP Dial-peers
0x17740    22AA 0x3D1D09A4 0/0/0:15.4       0/1:2   4312      g711ulaw 200/202
1 active calls found
XX-XX-VG01#sh ccm-manager music-on-hold
Current active multicast sessions : 1
Multicast       RTP port   Packets       Call   Codec    Incoming
Address         number     in/out        id              Interface
===================================================================
239.1.1.1         16384   0/0              96064 g711ulaw
XX-XXX-VG01#sh ephone summary
hairpin_block:
Max 58, Registered 0, Unregistered 0, Deceased 0 High Water Mark 59, Sockets 0
ephone_send_packet process switched 0
Max Conferences 32 with 0 active (4 allowed)
Skinny Music On Hold Status - group 0
Active MOH clients 0 (max 210), Media Clients 0, B-ACD Clients 0
File music-on-hold.au (not cached) type AU Media_Payload_G711Ulaw64k 160 bytes
Moh multicast 239.1.1.1 port 16384 route 10.114.80.1 1.1.1.1
I can confirm that the flash of the local router has indeed the music-on-hold.au file
XX-XXX-VG01#dir flash:
Directory of flash0:/
247 -rw-      496521 Sep 20 2014 10:30:06 +02:00 music-on-hold.au
263 -rw-    90063932 Sep 22 2014 11:24:06 +02:00 c2951-universalk9-mz.SPA.152-4.M6a.bin
256503808 bytes total (141819904 bytes free)

Hi Rachel,
The IP address CUCM selects is based on the codec and the audio stream.
See the following URL. http://voiceonbits.com/2010/06/29/moh-issues-and-resolution/
I borrowed this table from that URL and cleaned up the port numbers. Hopefully this helps.
Inc. Multicast on IP Address
Inc. Multicast on Port Number
Audio Stream
Codec
Dst. IP Address
Dst. Port
Dst. IP Address
Dst. Port
1
G.711 ulaw
239.1.1.1
16384
239.1.1.1
16384
1
G.711 Alaw
239.1.1.2
16384
239.1.1.1
16386
1
G.729
239.1.1.3
16384
239.1.1.1
16388
1
Wideband
239.1.1.4
16384
239.1.1.1
16390
2
G.711 ulaw
239.1.1.5
16384
239.1.1.1
16392
2
G.711 Alaw
239.1.1.6
16384
239.1.1.1
16394
2
G.729
239.1.1.7
16384
239.1.1.1
16396
2
Wideband
239.1.1.8
16384
239.1.1.1
16398

What licenses do I need for CUCM and CME

Hi Experts,
I am a newbie to Cisco IP telephony and request your guidance.
My boss has asked me to order some 7942 phones for a CUCM based site and some 7942 phones for a CME based site.
For CUCM 9, I understand that we need an enhanced user licenses on CUCM side.
For the IP phone, I see 2 part numbers and am confused.
CP-7942G - Cisco Unified IP Phone 7942G
CP-7942G-CH1 - Cisco Unified IP Phone 7942G, for Channels, with one station user license
What is the station user license mentioned in the part number CP-7942G-CH1 ? Is it the same as the enhanced user licenses? Assuming i am ordering enhanced user licenses, what should I order CP-7942G or CP-7942G-CH1?
I also see a part number SW-CCM-UL-7942= ? What is this part? is it needed?
And what do I need for CME?
Is FL-CME-SRST-10 enough for the CME?
What is the part number for the phones? Is it CP-7942G or CP-7942G-CCME?
I also see another part number "SW-CCME-UL-7942="? What is it? And is it really needed?
I heard that Cisco simplified Licensing from version 9. At least it doesn't look simple to me :(
Can you please help me, Experts?
Thanks,
Pete

Hi without knowing your business goals, current network, etc here is some feedback.
Depending on your design, if the sites are connected via a reliable WAN you could run the branch site as SRST and not CME, this way you get a rich feature set of the centralised call manager and simplify licenses, etc.
Anyway, you need to size your CUCME (or SRST) correctly as each Cisco router hardware platform has a maximum supported handsets. If you have a SIP provider new purchased ISRs now come with 10 CUBE licenses other wise you will need something else such as a VIC for dial tone.
For instance the following is a router that comes with 25 CME
C2901-CME-SRST/K9
2901 Voice Bundle w/PVDM3-16,FL-CME-SRST-25,UC Lic,FL-CUBE10
You can add CP-7942G-CCME which is the physical phone and license for CME
Use CP-7942G= for the Call Manager Deployment as you will get UCL Enh or CUWL for the user side
If you have existing routers you will need to obtain upgrade licenses for voice/CME add DSPs if needed for conferencing, media termination, etc, CUBE or voice cards and you can add the phones.
Router handset capacities see Table 7
http://www.cisco.com/c/en/us/products/collateral/routers/2900-series-integrated-services-routers-isr/data_sheet_c78_553896.html
Hope this helps

What is the recommended Delay/Latency between Call manager and The SRST setup

Hi ,
Need to understand the readability between the CUCM and SRST.
What is the recommended Delay/Latency, Bandwidh between Call manager and The SRST gateway setup.
Regards,
Velu S

Hi Manish,
I've been struggling to get this information and what I could understand from the SRND is that this 80ms is just related to the Intra-Cluster communications (Between Servers UCS), there is no relation with SRST Gateways:
"The maximum one-way delay between any two Unified CM servers should not exceed 40 ms, or 80 ms round-trip time."
From that I assume that the RTD between the SRST and the Cluster should be based on the affirmation below:
"If a voice service is hosted across a WAN where the one-way latency is 200 ms, for example, users might experience issues such as delay-to-dialtone or increased media cut-through delays. For other services such as presence, there might be no problem with a 200 ms latency."

Jabber and SRST???

Cisco,
I'd like to see if proper SRST support for Jabber is coming to CUCM and IM&P, or if the community is going to continue to be limited in this area? In its current iteration, Jabber is only SRST capable if an active connection to IM&P remains in place. In most cases, this is a catch 22, as increasingly remote environments and users, if losing connectivity to CUCM, lose connectivity to the colocated IM&P server as well. That limitation makes absolutely no sense at all.
Many customers these days are looking to cut costs, and with the advances in application capabilities and feature sets, many are going softphone only and doing away with desk phones altogether. I have one such customer in the early stages of their BE deployment right now. They wish to use Jabber only, but require SRST at their 10 remote site locations. Because of the, frankly, ridiculous caveat in Jabber, this is proving to be a serious issue for the deal.
Our only option for softphone failover is IP Communicator. Why would an aging platform that is no longer the standard or direction that Cisco is pushing be the only option for softphone failover? If Jabber is the new standard, its feature set should at least be comparable to what was, and provide for phone mode only SRST capability. At this point, I now have to configure and maintain inert CIPC devices in CUCM for use on the off chance a remote site circuit goes down. Add to that the overhead for management, user turnover, and additional training on a platform that users will infrequently use, and it adds up to a lot of lost dollars in time.
This has been an issue since Jabber's release, and a hot topic on the forums since the early versions, and still goes unresolved. I believe that this is a very worthy feature for Cisco to address (and more useful and silly things like popup chat windows and toasts) and I'm disappointed that the community's pleas have gone unanswered.
Can we please please get this on the roadmap?! Thanks!
Jimmy

I haven't gotten any info pointing toward a full resolution on the roadmap. What I have learned from a Cisco contact is that Jabber will continue to function if its running when the connection to IMP and CUCM are lost, however, if Jabber is closed, any subsequent login attempt after that, while CUCM and IMP are down, will fail. Of course, this isn't a very viable solution for a customer. Here's what we're doing for failover with Jabber only environments.
We're are installing redundant WAN circuits at each customer location that Jabber can then use for MRA registration back to CUCM and IMP at the two locations where the severs reside. That gives complete redundancy across the sites from the two HQ server locations all the way out to the branch office.
Thanks
Jimmy

CTSMan 1.9.1 and CUCM 8.6.2 sync problems Error 502407

Hi,
I ran into recent problem synchronizing CUCM 8.6.2(b) into CTSman 1.9.1
CTSMan is getting an Error message: 502407
Details
ID:
502407
Severity:
critical
Module:
DiscoveryMgr
SubModule:
CTIAdapter
Summary:
Communication to Unified CM failure
Recommendation:
Verify CCM App User credentials. Verify CTIManager service is activated on the primary application server.
Message:
Unable to create CTI adapter to Unified CM 'CUCM-xxx' because Provider is null.
I researched this earlier in CTSman admin guides, and this error points to CUCM CTI Manager service to be restarted. I did restart this as well as other services, including CUCM hardware itself and, but in vain.
Among other things, i followed the CUCM/CTSman guides when bringing CUCM into CTSman to: created an application user account within CUCM and allocate a proper permissions/user groups etc... it was done.
Among other things there are other servers like Exchange being synchronized with it ( had problems with it earlier that dealt with certificates, it was corrected.).
Can anyone help with this issue.
Responses will be greatly appreciated.
Thank you.

Thanks Manish again for this.
I did get few MediaReosurceListExhausted events from RTMT alerts but was not paying much attention to them. Location bandwidth settings have been checked and increased a bit for sites which were repeatedly getting these events. I will monitor to see if that reduces these alerts. Could you please advise how I can verify it is hitting the maximum limit?
I have also raised this with Cisco TAC and some cores were found on the CUC subscriber server. We seemed to have run into a bug according to Cisco TAC as in here https://tools.cisco.com/bugsearch/bug/CSCug61581/?reffering_site=dumpcr and both CUC servers have been upgraded to 8_6_2_24901_1. It has fixed cores and we are now monitoring the voicemail service for few days to see if it will reoccur.
Regards,

CUCM 8.6.2 LDAP User Delete Pending LDAP Sync Status Inactive

BE6K ver 8.6.2
Client has a user who recently got married. They changed her account information in Active Directtory to reflect her new last name. At that point CUCM shows her as
Delete Pending
LDAP Sync Status Inactive
CUC shows
LDAP User has been deleted.
The user still exists in both CUC and CUCM and is actively takign and receiving calls. User has VM access.
Shorrt of deleting the user in AD and recreating her, is there a way to force this to re-sync?
Thanks
Matt

Then that's expected to happen, for all purposes to CUCM/CUC eyes, msmith no longer exists and will be deleted, and a new user mjones now will be imported.
Depending on when the change was done and when CUCM detected this, it might take up to 48 hours maximum to delete the user
You'll need to associate everything to the new user, and also add that new user into CUC.
Or switch back her userID to the old one, and just change the surname for directory purposes.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk

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

No ringbacktone for inbound calls with cucm 8.6

Hi,
we have this problem from many days...
we have two branches with cucm cluster(Publisher and Subscriber) at Head office and cisco untiy.The branches are connected to Head office through MPLS vpn and all the ip phones are registred to publisher located at headoffice.
our setup is like below
HO and BR2 having SIP lines and BR1 has PSTN Lines.
we implement greetings for head office and 2 branches at Headoffice Unity.
when any call comes to headoffice gateway the greetings will be played and call will be diverted to the appropriate extension.everything is fine.
But the problem is when the call comes to Branch gateway and the greetings will be played and the call gets diverted to the IP phone to which the caller dialed the extension. but the caller is not hearing the ringback tone while the extension is ringing. and the caller cannot know whether the extension is ringing or the call got disconnected.
i tried to change the " Send h225 User Information Message" in service parameters from "Use ANN for Ring Back" to H225 Info for call Progress Tone"
whenever i am changing to "H225 Info for call Progress Tone" then the branches problem getting solved but Headoffice getting the same problem.
please can anyone help............................

Hi Carlo,
Thankyou for the Response...
here is the Runn config for BR1 Connected to PSTN lines....
voice-card 0
dspfarm
dsp services dspfarm
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
voice class h323 1
h225 timeout tcp establish 3
interface Tunnel100
description " Tunnel JED-RYD "
bandwidth 2048
ip address 10.10.0.1 255.255.255.252
tunnel source 172.31.217.202
tunnel destination 172.31.3.18
interface FastEthernet0/0
description DAMMAM Local LAN
no ip address
duplex auto
speed auto
interface FastEthernet0/0.20
description JEDDAH Local LAN
encapsulation dot1Q 20
ip address 192.168.20.5 255.255.255.0
interface FastEthernet0/0.21
description JEDDAH VOICE VLAN
encapsulation dot1Q 21
ip address 192.168.21.5 255.255.255.0
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.21.5
interface FastEthernet0/1
ip address 172.31.217.202 255.255.255.252
duplex auto
speed auto
router eigrp 200
network 10.10.0.0 0.0.0.3
network 192.168.20.0
network 192.168.21.0
no auto-summary
router bgp 65412
no synchronization
bgp log-neighbor-changes
neighbor 172.31.217.201 remote-as 65000
no auto-summary
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.20.1
ip route 192.168.20.50 255.255.255.255 192.168.20.1
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
voice-port 0/0/0
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/1
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/2
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/3
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/2/0
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
connection plar 2022
shutdown
impedance complex2
description STC
voice-port 0/2/1
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
shutdown
impedance complex2
description STC
voice-port 0/3/0
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/3/1
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/3/2
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/3/3
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
sccp local FastEthernet0/0.21
sccp ccm 192.168.12.190 identifier 1 priority 1 version 5.0.1
sccp ccm 192.168.12.189 identifier 2 priority 2 version 5.0.1
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register CONFJEDRAW
associate profile 2 register TRNJED
associate profile 3 register MTPJED
switchover method immediate
switchback method immediate
switchback interval 15
dspfarm profile 2 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 2
associate application SCCP
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
shutdown
dspfarm profile 3 mtp
codec g729r8
maximum sessions software 250
associate application SCCP
shutdown
dial-peer voice 1 pots
dial-peer voice 1000 voip
description To CallManager - SBWPMPUB
destination-pattern [1-5]...
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.190
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 9001 pots
description ** 02-6140294(outgoing) **
destination-pattern [^2].T
port 0/0/1
dial-peer voice 9002 pots
description ** 02-6140295(outgoing) **
destination-pattern [^2].T
port 0/0/2
dial-peer voice 9003 pots
description ** 02-6140296(outgoing) **
destination-pattern [^2].T
port 0/0/3
dial-peer voice 9004 pots
description ** 02-6140293(outgoing) **
destination-pattern [^2].T
port 0/0/0
dial-peer voice 290 pots
incoming called-number .
direct-inward-dial
dial-peer voice 9006 pots
description ** 02-6529323(local) **
destination-pattern [^0].T
port 0/3/0
dial-peer voice 9010 pots
description ** 02-6578249(local) **
destination-pattern [^0].T
port 0/3/1
dial-peer voice 9011 pots
description "to pstn service"
shutdown
destination-pattern 0.T
port 0/3/3
dial-peer voice 9009 pots
description "to pstn service"
shutdown
destination-pattern [^0].T
port 0/3/2
dial-peer voice 9005 pots
destination-pattern .T
dial-peer voice 1001 voip
description To CallManager - Subscriber
destination-pattern [1-5]...
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.189
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 1002 voip
description " TO Unity Greetings"
destination-pattern 2050
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.190
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 1003 voip
description " TO Unity Greetings"
destination-pattern 2050
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.189
dtmf-relay h245-alphanumeric
no vad

SRST maximum on CUCM

Similar Messages

Maybe you are looking for