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.

Similar Messages

CUCM Line Group 2 Not Ringing Phone

Good morning,
I have encountered a weird issue and was wonder if anyone here could provide some insight before I send it up to TAC.
I have a Hunt Pilot (HP)/ Hunt List (HL) and the HL has 3 Line Groups (LG) associated to it. Example LG configuration is below.
LG1: 1001, 1002, 1003
LG2: 1001, 1002, 1003, 1004
LG3: 1001, 1002, 1003, 1004, 1005
All LGs are set to RNA of 15s and broadcast.
Call comes in and hits LG1. 1001, and 1002 are on the phone so they do not receive a notification and 1003 just doesn't answer call even though 1003 is the only one ringing. Call rolls to LG2 and 1001, and 1002 are still on phone, but 1003 and 1004 don't answer even though those two phones are only ringing. Before the call rolls to LG3, 1001 hangs up the phone. Now when the call rolls to LG3, 1001 doesn't ring. The only phones that ring are now 1003, 1004, and 1005. 1002 is still on the phone from the beginning and this is expected of course but why doesn't 1001 ring even though the phone is on hook and was put on hook before the call went to LG3?
Phone Types: 6921, 7821, 7942, 7962
Phone FW: 6921 - SCCP 9.4.1.3, 7942 - SCCP 9.3.1ES13, 7962 - SCCP 9.3.1ES13, 7821 - 10.1.1ES6
CM Version: 9.1.2.10000-28
Thanks,

Anyone?

CUCM - P-Asserted-Identity

Topology: CUCM ---- SIP ---- CUBE ---- SIP ---- SP
Sometimes, when I place a call, it redirects it to an internal extension. The only SIP message that I can see being sent when that happens is a 180 ringing with a p-asserted-identity in it. It only happens about 50% of the time. example: I dial 9.8001234567 and end up talking to someone at extension 3041. The Dialed Number Analyzer shows everything correctly. This SIP message is below. HELP!! please
10/01/2013 08:59:39.274 CCM|//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to -CUBE--->10.3.20.240:[5060]:
SIP/2.0 180 Ringing
Date: Tue, 01 Oct 2013 13:59:39 GMT
Call-Info: <sip:-CUCM--->10.3.20.10:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: <sip:5141234567@--SP Edge device----->22.22.22.22>;tag=739821B4-19C2
Allow-Events: presence
P-Asserted-Identity: "Switchboard" <sip:[email protected]>
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Remote-Party-ID: "Switchboard" <sip:[email protected]>;party=called;screen=yes;privacy=off
Content-Length: 0
To: <sip:[email protected]>;tag=87d22a15-fd7d-492e-85ed-0dc8d67386d5-30912190
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
Via: SIP/2.0/UDP 10.3.20.240:5060;branch=z9hG4bK2D01637
CSeq: 101 INVITE

My guess is that the call got hairpinned back to CUCM by the provider or CUBE. We need to see the full trace from CUBE to be sure. PAI is really just attempting to indicate who the call is actually alerting because the From and To headers cannot change after the INVITE message.
The fact that CUCM is sending a 180 RINGING message implies that it is processing an *incoming* call. This is reinforced by the headers because the From header is the SP SBC, the via header is CUBE, and the To header is CUCM. Inbound call!
Compare the Call-ID of the SIP INVITE CUCM first sends to CUBE for your outbound call to the one shown in this message. Are they different?
Run these if you want us to look deeper. If you have multiple calls going be certain to point out the calling/called number and the IPs if any differ from what you have called out above.
show run | section dial-peerdebug ccsip messagesdebug voip dialpeer
Please remember to rate helpful responses and identify helpful or correct answers.

CUCM sends 180 followed by 180 SDP for early media.

Hi,
CUCM interacting with PBX from another vendor and facing some problem with early media. From the wireshark traces, i see that CUCM sends 180 followed by 180 with SDP which is causing an issue on other PBX. But my doubt is, i see the article says that CUCM never sends 180 SDP, only it handles 180 SDP if it receives from far-end.
PBX                 CUCM
    ---- INVITE ------->
   <--- 100 Trying ----
   <--- 180 ringing ----
   30 ms
    <---- 180 /SDP -----
Is it bug on CUCM or expected since it claims that CUCM does not send 180 SDP ? Or can we avoid this situation by changing some configuration ?
Regards,
Soman.

Hi,
Can you please send us the full CUCM trace. Please include calling and called number
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"

No Ring Back Tone after receiving 180/183(SDP) wo RTP

Hi,
I would like to ask how CUCM decides about which Ring Back Tone (Remote/Local/nothing) should be played?
There is a problem wisth 180/183 Message with SDP but without RTP. When CUCM receives that message, no Ring Back Tone is played. (CUCM 8.6 was used)
What's more:
1. When Ipphone receives 183(SDP) - remote device is sending RTP stream and later ipphone receives 180 (no SDP) -> Ipphone does not switch to play local RBT.
2. But when Ipphone receives 180(no SDP) and later Ipphone receives 183 (SDP) with RTP -> IPphone switches to play remote RBT.
Thanks for any help.
Marta

What is the PBX? Check if the PBX and version is on the Microsoft Interoperability Program (Supported or Qualified): http://technet.microsoft.com/en-us/office/dn788945 If listed,
there should be configuration notes or a deployment document.
If not listed, you might need to deploy a gateway between Lync and the PBX to resolve any interop issues.
Please mark posts as answers/helpful if it answers your question.
Blog
Lync Validator - Used to assist in the validation and documentation of Lync Server 2013.

Ringing Volume and Speaker Volume

Ok..so, I saw the fix but it is temporary.. do we have any news for a permament fix from Apple yet?
The volume suxs!

I'm experiencing the exact same problem!
When the maximum ringing volume is set, the phone will ring for about 3 seconds at a certain volume, and only then the volume will rise to the maximum. And it's set on "ringing", NOT "ascending"!
Ideas, anyone?

Is there additional ring tones? or How to make custom ring tones?

Hi,
I am thinking about adding additional ring tones.
Is there any site I can download the ringtones for cisco 7940, 7960, 7940G, 7960G?
or
Can I make custom ring tones by myself?
If there is an application to do the ring tones, pls anyone let me know?
thanks

Hi San,
Just to add a note to the great tips from Ingo and Bethany (+5 points each folks for your nice work here :)
Here we go :)
PCM File Requirements for Custom Ring Types
The PCM files for the rings must meet the following requirements for proper playback on Cisco Unified IP Phones:
â¢Raw PCM (no header)
â¢8000 samples per second
â¢8 bits per sample
â¢uLaw compression
â¢Maximum ring size-16080 samples
â¢Minimum ring size-240 samples
â¢Number of samples in the ring is evenly divisible by 240.
â¢Ring starts and ends at the zero crossing.
â¢To create PCM files for custom phone rings, you can use any standard audio editing packages that support these file format requirements.
Configuring a Custom Phone Ring
To create custom phone rings for the Cisco Unified IP Phone 7965G and 7945G, follow these steps:
Procedure
Step 1 Create a PCM file for each custom ring (one ring per file). Ensure the PCM files comply with the format guidelines that are listed in the "PCM File Requirements for Custom Ring Types" section.
Step 2 Upload the new PCM files that you created to the Cisco TFTP server for each Cisco Unified Communications Manager in your cluster. For more information, see the "Software Upgrades" chapter in Cisco Unified Communications Operating System Administration Guide.
Step 3 Use a text editor to edit the Ringlist.xml file. See the "Ringlist.xml File Format Requirements" section for information about how to format this file and for a sample Ringlist.xml file.
Step 4 Save your modifications and close the Ringlist.xml file.
Step 5 To cache the new Ringlist.xml file, stop and start the TFTP service by using Cisco Unified Serviceability or disable and re-enable the "Enable Caching of Constant and Bin Files at Startup" TFTP service parameter (located in the Advanced Service Parameters).
http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7965g_7945g/6_1/english/administration/guide/7965cst.html#wp1097651
There is a nice little Step by Step here;
http://www.globalknowledge.com/training/whitepaperdetail.asp?pageid=502&wpid=473&country=United+States
Or search sites like this one;
http://www.adman.net/cisco/ringtones/browse.asp
Hope this helps!
Rob
PS: Interesting tunes from your link Bethany, I'd love to hear "Scar Tissue" as a ringtone, pretty cool :)

TCS SIP URIs drop registration

Good Morning all,
Have an odd issue. I have a new TCS implementation that we had installed just about 2 weeks ago. The TCS will register H.323 and SIP URIs to my VCS without issue. However, I see that the SIP URIs disappear from the VCS after a while. I haven't been able to narrow down a time-frame or activity that causes it, but as soon as I reset the TCS services, it will re-register. The H.323 registration stays without problems. Any ideas why it does that??
VCS: X7.1
TCS: Cisco TelePresence Content Server v5.3 Build 3316

Hi Jaime,
Thanks for the reply..
Funnily enough I was thinking of something similar to your reply.. and it did occur to me how it would differentiate what resides in VC land and what is in CUCM land..
Having a different domain name for our CUCM registered DX80 endpoints is not particularly desirable, therefore does that mean we would need to define entry for each endpoint to tell it to route towards CUCM..
Ive made a bit of progress today but it's still messy to get it to work.. but think it touches on what your are saying..
So...
Dialling [email protected], hits the search rule on VCS which redirects it to the DX80 ( in CUCM land ) and our end point rings.
Dialling [email protected], does not seem to hit the search rule on VCS ( get the error, 404 Not found ) and therefore does not get routed to CUCM.. This is strange as it is basically the same as the rule Ive created for 1001.
** If it did manage to hit CUCM, I would expect it to resolve to the end point 1001, as Ive created a Directory URI associating the extension and SIP name..
As dialling [email protected] seemed to work, Ive created a new translation search rule on VCS.. So..
Dialling [email protected] now hits a search rule,which translates the entry to…. [email protected]
This then matches the other rule and correctly routes over to CUCM and our DX80 endpoint rings. !.
Again, this is odd as it does prove that a variation of the [email protected] search rule does work , so I don’t understand why a simple reroute statement doesn’t work and we're having to do a translation instead !!..
So kinda working but a very messy way of doing it…. And also we'd have to assign a static routing entry for each DX80 endpoint..
Any thoughts please ?

CUCM 8.6.1.2000 - Call History not logging, and no ring tone access.

Hello,
A little background:
Recently, I setup a UCS C-210 M2 with CUCM 8.6.1 and restored our Pub and Sub to it. Also migrated to from physical Unity to Virtual CUC, same with Presence.
I've also setup a 2nd CUCM Sub, as well as new CUC and CUP subs on a second UCS C210. I also upgraded CUP to 8.6.4. before adding the Sub CUP node.
I'm thinking my problems began when I added the 2nd Subscriber CUCM node on our 2nd C210. Here is a description of what's going on.
- I cannot change ring tones on an IP phone. I get "Ring List Unavailable" when I try to change a ring tone. I noticed that my phone had changed over to the CTU ring tone around the time I brought up the VM environment. The CTU ring tone was a custom one I had loaded at one time.
- I cannot access any call history on my IP phone. My Jabber client logs call history just fine.
Seems to be a TFTP issue, but I restarted the Cisco TFTP service yesterday, and it did not help. I also reset the Device Pool, which didn't seem to do anything.
Also, I'm still a bit unclear on which services I should be running on my new Subscriber node.
Any help is very much appreciated. Thanks again, guys!!
-Jonathan

I had this issue and resolved it. The problem was that we had changed DHCP servers over to a different platform and the new DHCP server didn't have Option 66 defined to point the phones to a TFTP server. Once we re-specified the TFTP server IP's for option 66 in DHCP and reboot the phones the issue was resolved.

Maximum file size in picture ring?

Hello folks!
I am planing to use a picture ring with a quite big amount of data needed.
My question: is there a maximum data size that i can embed in a picture ring (number of pictures or overall file sizes)?
Thanks!

If you have enough memory to keep all the images open simultaneously, then something like this might help. Put all your images in the same directory on disk and have no other files in that directory. Then use List Folder from the Advanced File palette to get an aaray of the filenames. Feed that array to a for loop where you open all the files and place the images into the pict ring. I have written a "slide show" program which does this. Never tried it with 400 images though.
If you do not have enough memory for all the images, then you need to manage the iamges much more carefully.
Lynn

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!

When I receive a call it won't allow me to hear the person on the other line unless I put my headphones in or use the speaker function. And when I turn the ringer volume up/down it is in maximum

when i recevie a call it won`t allow me to hear normally

There is nothing to input because your licensing system isn't functional. Reinstall the software properly using the full suite installer, not some obscure single app installers or whatever you mean by "backup".
Download CS3 products
Mylenium

Cisco CUCM 9.1 Call Queuing no Ring Back

I have a customer that has a hunt group that has 6 users in it, they are routing the calls by longest idle. The way the customer handles the inbound call is with the operator-> then they transfer the call to the hunt group. They want to be able to present MOH when they transfer to the hunt group. I enabled Call Queuing for this hunt group and set the MOH source to the customers recording , and work great as long as all agents are busy. When the callers are trasfered to the hunt and agents are available they hear the ringback tones as the call goes from agent to agnet until it gets answered. The customer is wanting to eliminate the ringback tone during the hunt cycle and just play MOH. I know this is possible with UCCX, but they are not willing to purchase UCCX. Is there any way i can silence the ringback tone and have MOH while the call is hunting?

Hi, there is a product intended exactly to do that:
http://www.imagicle.com/go/queuemanager
It extends the embedded call queuing capabilities and is already certified with 9.1, available in solution catalog and also providing advanced historical reporting.
Licensed per channels and not per operator/agent, cheap, dramatically EASY to deploy, config and manage.
Offer a free of charge supervisor app on iPAD and can be combined with the imagicle operator console (http://www.imagicle.com/go/bluesattendant) for a full, professional, out of the box customer service solution.
You can contact imagicle for more info or download the free 30 days eval.
Regards.
Christian Bongiovanni
coCEO and CTO
Imagicle SpA

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

CUCM Mobility/Single Number Reach is not working correctly

Hi,
I'm experiencing difficulties with the CUCM Mobility (Single Number Reach) function.
For a customer of mine, I'm busy setting up the Mobility/Single Number Reach which is designed as follows:
- Users have a 4 digit DN that is attached to their User Device Profile so that they can use Extension Mobility.
- Those same users also have a Remote Destination Profile with a Remote Destination (to a mobile number) attached to their DN.
All has been set but as I was testing a couple of DNs, I noticed that a some numbers could be called to both their DN and Remote Destinations while others could only reach their DN.
As an example I have configured DN 6380 with the correct CSS (which permits to call to mobile phones and national numbers) to a User Device Profile.
That same DN is also connected to a Remote Destination Profile with a configured Remote Destination, which also has the same CSS.
The End User that is needed to login into Extention Mobility is 6380.
The settings on the Remote Destination are to ring always and all the time.
All Remote Destinations have the "Line Association" active.
Each DN has a value of 3 in maximum number of lines field.
Their Remote Destination profile has a value of 2 in maximum number of lines.
With this particular user, I'm sure that I have the right mobile number.
I already found out that the numbers that are having this problem, do not make a call to the PSTN and mobile network when their DN are called.
So I think the problem is within CUCM.
Can somebody help me?
Many thanks in advance!
The version of call manager is 8.6.2.20000-2.

I did a debug isdn q931 on the voice router which is connected to the PRI circuit.
A couple of test calls later and this is the output I got:
Jun 28 08:34:15.737: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0E5A
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18382
                Preferred, Channel 2
        Calling Party Number i = 0x2183, '51365XXXX'
                Plan:ISDN, Type:National
        Called Party Number i = 0xA1, '88777XXXX'
                Plan:ISDN, Type:National
        Sending Complete
Jun 28 08:34:15.749: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8E5A
        Channel ID i = 0xA98382
                Exclusive, Channel 2
This company has 200+ Mobility users so I did a random check.
Strangely enough, the one I described in my first post is now reachable on both his DN and mobile phone.
This is 4 digit number was the only one though.
Jun 28 08:09:20.557: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x00A6
       Bearer Capability i = 0x8090A3
               Standard = CCITT
               Transfer Capability = Speech
               Transfer Mode = Circuit
               Transfer Rate = 64 kbit/s
       Channel ID i = 0xA18382
               Preferred, Channel 2
       Calling Party Number i = 0x2181, '51365XXXX'
               Plan:ISDN, Type:National
       Called Party Number i = 0xA1, '88777XXXX'
               Plan:ISDN, Type:National
       High Layer Compat i = 0x9181
       Sending Complete
Jun 28 08:09:20.569: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x80A6
       Channel ID i = 0xA98382
               Exclusive, Channel 2
Jun 28 08:09:20.573: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x2345
       Sending Complete
       Bearer Capability i = 0x8090A3
               Standard = CCITT
               Transfer Capability = Speech
               Transfer Mode = Circuit
               Transfer Rate = 64 kbit/s
       Channel ID i = 0xA9839C
               Exclusive, Channel 28
       Display i = 'Testnummer'
       Called Party Number i = 0x80, '51365XXXX'
               Plan:Unknown, Type:Unknown
        Redirecting Number i = 0x00008F, '088777XXXX'
               Plan:Unknown, Type:Unknown
I did a Remote Destination Profile export of all records with a Remote Destination attached to it and then re-imported them in CUCM.
The last output of the call to the mobile phone was not appearing last Thursday.
Apparently, the export and import of the profiles and destinations did change something within CUCM.
Could this be a bug in CUCM?

CUCM 8.6.2 maximum rings

Similar Messages

Maybe you are looking for