3527 PRI GW + GK + CUCM

Issue is the following:
Inbound video from a remote H.320 Tandberg platform fails when it calls to a SCCP 7985.
Call Flow:
H.320 Tanberg places a call to a number that lives on a 7985. This call hits a video PRI terminated on a 3527 H.320/H.323 gateway. 3527 does a lookup for destination and the default path is a trunk that is between the GK and a CUCM5.1 platform to which the 7985 is registered. CUCM sees the call ingress and places it to the 7985 where the call rings.
Problem:
TCS is not negotiating. If the call is just an audio call from my cell phone, the call works fine. However, when the call is placed as a video call, the call initially connects at 64k and tries to negotiate up to video, but fails and dies after 10 seconds (timeout because it would seem the remote end Tandberg is not sending its TCS).
We have ruled out the remote end by placing inbound calls from other remote video conferencing stations and have had the same issue, so, from our perspective, the issue lay somewhere between the 3527 and the Callmanager in negotiating the TCS.
Important Note:
Callmanger trunk with GK is a GK controller H.225 trunk with "Wait for Far End H.245 Terminal Capability Set" unchecked (as per SRND). Tested it out with the toggle checked and still failed.
GK configuration is as follows:
voice service voip
h323
no call service stop
gatekeeper
zone local TEMP mydomain.com 10.10.10.1
gw-type-prefix 1#* default-technology
no use-proxy TEMP default inbound-to terminal
no use-proxy TEMP default outbound-from terminal
no shutdown
Any help would be most appreciated. My guess is that a an H.245 pass-through configuration may be needed on the GK, but that is a grasp at straws.

There is a bug CSCee16667 filed for this. Later releases seems to fix it.You can also try to reset tandberg.

Similar Messages

Re-Registering the Address of a PRI Router in CUCM

Hello All,
We are moving one of our old PRI Routers to our new building we just moved into and in doing so we need to change its IP Address.
Once we change the address of the router, how do we "re-register" the device to the CallManager? And is there anything we should be aware of/look out for, when doing so?
Thanks in Advance,
Matt

Hey Guys, thanks for the reply.
There are two H.323 Gateways. So we've already changed the IP Address on the Gateways. I then went into CallManager Device > Gateways, found the Gateways that now have a Registration of "Unknown". Changed the IP Address to the correct one, Saved it then Applied the Config.
But it still says Registration "Unknown". Do I need to reboot the Gateway or anything like that?
Thanks Again,
Matt

Telco Messages via PRI (VG) connected to CUCM 9.X via SIP Trunk??

Hello
Questions:
Should I hear tel-co message on an IP Phone if a call that is meant to be long distance is sent to the gateway without a one (1). Currently these calls simply ring until disconnect, but work properly if the user dials a 1, the user expects to get a message from the provider and I am wondering if the SIP trunk between GW and CUCM is not allowing it?
Scenario:
IP Phone --> CUCM (SIP Trunk) --> ISR 2901(PRI) --> PSTN
In CUCM we have a local pattern 9.[2-9]XX[2-9]XXXXXX
If the IP Phone dials a number that is actually a long distance number, which would require a 1, but they dial it as a 10 digit local number, the call gets to the gateway and IP Phone hears ringing but it, rings until it eventually disconnects. At this point I believe the gateway is sending a cause code back to CUCM and I would expect error message back from the telco informing the caller of the issue, such as "Please dial 1 before a long distance number". Is there a specific setting on the SIP trunk and or Gateway to achieve this?
See Q931 Debug
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9838F
Exclusive, Channel 15
Display i = 0xB1, 'London Hydro'
Calling Party Number
UCS5-GW-02# i = 0x2181, '5193334444'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '5191112222'
Plan:ISDN, Type:National
*Jul 23 15:33:28.363: ISDN Se0/0/1:23 Q931: RX <- STATUS pd = 8 callref = 0xC3FA
Cause i = 0x80AB28 - Access information discarded
Call State i = 0x01
*Jul 23 15:33:28.411: ISDN Se0/0/1:23 Q931: RX <- CALL_PROC pd = 8 callref = 0xC3FA
Channel ID i = 0xA9838F
Exclusive, Channel 15
*Jul 23 15:33:28.415: ISDN Se0/0/1:23 Q931: RX <- PROGRESS pd = 8 callref = 0xC3FA
Cause i = 0x80FF - Interworking error; unspecified
Progress Ind i = 0x8088 - In-band info or appropriate now available
Thanks
Richard

Yes, Early media needs to be turned on the SIP Profile that is associated to the SIP trunk. The setting is SIP Rel 1xx options. It needs to be set for Send PRACK for all 1XX messages. If you have CUCM 7.x , this setting is a service parameter.

Not receiving the 486 message from CUCM to Genesys via SIP trunk.

I have setup where Genesys is used along with CUCM 9.1
Below is the snapshot how it will look for call flow.
PRI----V.G----CUCM---SIP trunk (created in CUCM)-----Geneys server.
Query here is for outbound call from SIP softphone to PSTN, where if the PSTN user cancel the call.
the SIP phone is still assuming the call is continuing and after 40 sec its getting disconnected.
after looking in to the sip traces... it looks like that SIP trunk from cucm is not sending the user busy message 486....
(checked in V.G and its giving user busy)...but in the CUCM its not getting sent to the genesys...
After some time in genesys server itself send the 480 Temporarily Not Available message...
I assume I should get the 486 message from CUCM to genesys when the PSTN party disconnect the call without answering.
Please assist.

From logs what i can see is after one min call legs stops transmitting and receiving packets.
You certainly need to check this Genesys support for this behavior , as far as I know there is no problem either with the CUCM or with VG.
1477 : 1515 22820290ms.1 +0 pid:0 Originate connecting
dur 00:01:15 tx:3765/602400 rx:3387/541760
IP 10.129.0.45:32596 SRTP: off rtt:0ms pl:67720/140ms lost:0/1/0 delay:55/55/65ms g711ulaw TextRelay: off
1477 : 1514 22820290ms.2 +0 pid:0 Originate active
dur 00:01:16 tx:3387/568856 rx:3838/614080
Tele 0/3/0:15 (1514) [0/3/0.31] tx:76760/76760/0ms g711ulaw noise:-68 acom:3 i/0:-64/-62 dBm
1477 : 1515 22820290ms.1 +0 pid:0 Originate connecting
dur 00:01:26 tx:4330/692800 rx:3387/541760
IP 10.129.0.45:32596 SRTP: off rtt:0ms pl:67720/140ms lost:0/1/0 delay:55/55/65ms g711ulaw TextRelay: off
1477 : 1514 22820290ms.2 +0 pid:0 Originate active
dur 00:01:30 tx:3387/568856 rx:4515/722400
Tele 0/3/0:15 (1514) [0/3/0.31] tx:90290/90290/0ms g711ulaw noise:-68 acom:3 i/0:-67/-61 dBm
Rate all the helpful post.
Thanks
Manish

Outbound FAX is not working

Outbound FAX is not working. Below is the Call Flow
FAX ServeràH.323àCUCMàMGCPàGatewayàE1 PRIàPSTN
The CUCM version is 8.0.3 while the gateway IOS version is 12.4(24)T5.
Collected the following
Packet Captures between FAX server and CUCM
CUCM traces
Debugs in PSTN Gateway
Below are the observations
H.225 Setup message is seen in packet captures and CUCM traces. See snippet below
16:53:00.684 |In Message -- H225SetupMsg -- Protocol= H225Protocol|*^*^*
16:53:00.684 |Ie - H225BearerCapabilityIe -- IEData= 04 03 90 90 A5 |*^*^*
16:53:00.685 |Ie - Q931CalledPartyIe -- IEData= 70 09 A1 39 36 33 32 33 39 39 38 |*^*^*
16:53:00.685 |Ie - H225UserUserIe -- IEData= 7E 00 86 05 30 88 06 00 08 91 4A 00 05 22 80 B5 00 00 30 13 44 69 61 6C 6F 67 69 63 20 43 6F 72 70 6F 72 61 74 69 6F 6E 00 AC 15 98 05 06 B8 00 14 15 79 1B C9 3E 02 1F 3F 03 80 B9 FE 5F AC 75 00 D5 0D 98 00 07 00 AC 15 98 14 07 FB 11 00 25 13 79 1B C9 3E 02 1F 3F 03 80 B9 FE 5F AC 75 01 00 01 00 01 00 01 00 01 00 01 40 40 B5 00 00 30 14 44 69 61 6C 6F 67 69 63 20 43 6F 72 70 6F 72 61 74 69 6F 6E 10 80 01 00 |*^*^*
16:53:00.685 |MMan_Id= 0. (iep= 0 dsl= 0 sapi= 0 ces= 0 IpAddr=149815ac IpPort=2043)|*^*^*
16:53:00.685 |IsdnMsgData1= 08 02 5B CA 05 04 03 90 90 A5 70 09 A1 39 36 33 32 33 39 39 38 7E 00 86 05 30 88 06 00 08 91 4A 00 05 22 80 B5 00 00 30 13 44 69 61 6C 6F 67 69 63 20 43 6F 72 70 6F 72 61 74 69 6F 6E 00 AC 15 98 05 06 B8 00 14 15 79 1B C9 3E 02 1F 3F 03 80 B9 FE 5F AC 75 00 D5 0D 98 00 07 00 AC 15 98 14 07 FB 11 00 25 13 79 1B C9 3E 02 1F 3F 03 80 B9 FE 5F AC 75 01 00 01 00 01 00 01 00 01 00 01 40 40 B5 00 00 30 14 44 69 61 6C 6F 67 69 63 20 43 6F 72 70 6F 72 61 74 69 6F 6E 10 80 01 00 |*^*^*
16:53:00.685 |value H323-UserInformation ::= |*^*^*
16:53:00.685 |SPROCRas - {
h323-uu-pdu
    h323-message-body setup :
        protocolIdentifier { 0 0 8 2250 0 5 },
        sourceInfo
          vendor
            vendor
              t35CountryCode 181,
              t35Extension 0,
              manufacturerCode 48
            productId '4469616C6F67696320436F72706F726174 ...'H
          terminal
          mc FALSE,
          undefinedNode FALSE
        destCallSignalAddress ipAddress :
            ip 'AC159805'H,
            port 1720
        activeMC FALSE,
        conferenceID '1415791BC93E021F3F0380B9FE5FAC75'H,
        conferenceGoal create : NULL,
        callType pointToPoint : NULL,
        sourceCallSignalAddress ipAddress :
            ip 'AC159814'H,
            port 2043
        callIdentifier
          guid '2513791BC93E021F3F0380B9FE5FAC75'H
        mediaWaitForConnect FALSE,|*^*^*
16:53:00.686 |
        canOverlapSend FALSE,
        multipleCalls FALSE,
        maintainConnection FALSE,
        presentationIndicator presentationAllowed : NULL,
        screeningIndicator userProvidedVerifiedAndFailed
    nonStandardData
      nonStandardIdentifier h221NonStandard :
          t35CountryCode 181,
          t35Extension 0,
          manufacturerCode 48
      data '4469616C6F67696320436F72706F726174 ...'H
    h245Tunneling FALSE
}|*^*^*
Based on this, the CUCM is doing correct digit analysis and routes the call to MGCP gateway’s endpoint S0/SU0/DS1-0/[email protected]
Gateway sends the ISDN q931 setup message to PSTN for which it does get the Call Proceed and Progress indicators
The gateway is dropping call in Cause Code “0x80AF - Resource unavailable, unspecified”. See Snippet below
016325: *May 2 17:12:47.661: MGCP Packet received from 172.21.152.5:2427--->
CRCX 5493 S0/SU0/DS1-0/[email protected] MGCP 0.1
C: D000000003d354dd000000F500000016
X: 1
L: p:20, a:PCMU, s:off, t:b8, fxr/fx:t38
M: recvonly
R: D/[0-9ABCD*#]
Q: process,loop
<---
016383: *May 2 17:12:47.669: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x0016
        Sending Complete
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98381
                Exclusive, Channel 1
        Calling Party Number i = 0x0083, N/A
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0xA0, '96323998'
                Plan:Unknown, Type:National
016384: *May 2 17:12:47.757: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x8016
        Channel ID i = 0xA98381
                Exclusive, Channel 1
016385: *May 2 17:12:47.761: ISDN Se0/0/0:15 Q931: RX <- PROGRESS pd = 8 callref = 0x8016
        Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info
016386: *May 2 09:12:48.193: %C3800_ENVM-3-MFAIL_OFF: There is more than one failure with the Power System 1 or this Power System h
as been turned off.
016387: *May 2 17:12:51.669: //494/2E810F7780E1/CCAPI/cc_handle_inter_digit_timer:
   Generate inter-digit timeout CC_EV_CALL_DIGIT_END event
016388: *May 2 17:12:59.773: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x0016
        Cause i = 0x80AF - Resource unavailable, unspecified
From the Fax Utility it shows No answer at Fax Number. Also it rings the remmote fax machine twice (Actually i can hear the ring as the macine is beside me)
Please help me to solve the issue. Except this other functions are working properly. Moreover I can send internal Fax from one user to other.
Thanks and Regards,
Ashfaque.

Just a quick update, as per Cisco TAC the CUCM version 8.0.3 does not support gateway IOS version is 12.4(24)T5 (Later I found its only for conference option). So we upgraded the IOS to 15.1(1)T3. But still having the sam issue.
Thanks,
Ashfaque

T38 rejected by ccm

Hi guys,
We did an upgrade from CUCM version: 7.1.5.34079-1 to CUCM version: 9.1.1.20000-5.
The problem now is that 95% of the incomming faxes are not coming into the fax server. (Some faxes are reveived)
Setup:
pri -> gateway (MGCP) -> CUCM -> ICT -> fax server
In de CUCM logs we see the next:
2 MDCX messages being sent to the gateway. The second one was being ignored.
This was because BOTH MDCX requests were in a=sendrecv mode. However, in the first MDCX, there is NO codec mentioned, while in the second there is.We believe this may be the reason for the MGCPInterface() not doing anything later in the call.
Did someone already had the same issue or knows a solution for the issue that i'm facing?
Any help would be much appreciated.
Regards,
Laurent

First of all, ICT is designed to communicate different CCM clusters, not to a Fax Server. Change the ICT to a H323 Trunk (Gateway):
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/trunks.html#wp1044685
Then you can further troubleshoot.
Jorge Armijo
Please remember to rate helpful responses and identify helpful or correct answers.

How to configure modem connection with GW (H323) and ATA 187

Hello Community,
i stock in configuration and need assistance.
My callflow: Telco – PRI – GW – H323– CUCM – SIP – ATA187 – Modem
Voicegateway (Version 15.3(2)T) + CUCM (Version= 8.6) + ATA187 (Version= 9.2.3.1)
The modem connection is still not working.
What is still to configure on the voicegateway? modem passthrough?
Regards Michael
ATA 187 Configuration:
Fax Mode= T.38 Fax Relay
Fax Error Correction Mode Override= Off
Maximum Fax Rate= 14000bps
Impedance= 900Ohms complex
Gateway Configuration:
voice service voip
ip address trusted list
ipv4 172.30.50.1
ipv4 172.30.50.2
ipv4 172.30.50.3
ipv4 172.30.50.4
ipv4 172.30.50.5
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h225-notify cid-update
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server expires max 600 min 60
dial-peer voice 1 pots
translation-profile incoming INCOMING_PSTN
incoming called-number .
direct-inward-dial
dial-peer voice 30 voip
description OUTGOING_CUCM
destination-pattern [1-9]..
session target ipv4:172.20.60.12
voice-class codec 1
dtmf-relay h245-alphanumeric
fax-relay ecm disable
fax-relay sg3-to-g3
fax nsf 000000
no vad

It is possible T38 isn't playing well with the PRI. You could try modem pass-through on the gateway and ATA187 if T38 isn't necessary.
Also, sometimes these commands are needed, but not always, so I would consider whether these fax commands under the dial-peer are necessary:
fax-relay ecm disable
fax-relay sg3-to-g3
fax nsf 000000

Audible alert that customer has disconnected?

Hello,
Our agents have asked if an audible alert can sound when a caller disconnects.
Their complaint is that if the customer is on hold and hangs up, they will be placed into the "Work" state and not know it until they return to their desk. This impacts their daily stats. This happens when they have left their desk to get help from a supervisor or work at a lab station (they use wireless headsets).
We are on UCCX 8.5xx and use 6921 phones for our agents.
Any input would be appreciated.
Thanks!

Short of the PC-based workflow that David mentioned the only way I could envisioning this happening would be with a custom TCL script on the voice gateway. It should be able to play a tone into the RTP stream upon receiving the disconnect event from the PRI before telling CUCM about the disconnect. There is no pre-built code for this to my knowledge so you'll need some developer skills.
Please remember to rate helpful responses and identify helpful or correct answers.

Standby PRI not working with voice Gateway Router & CUCM

Hi ALL ,
GOOD Day all of you .
I am facing a big problem i.e standby PRI not working with VG & CUCM , I have checked all the configuration parameter on VG & CUCM found ok but I am unable to make any call from standby link also incoming not come on the standby link . When I make a call on my Pilot no but getting busy tone .
I observer the some errors on VG like Cause i = 0x8286 - Channel unacceptable on my second PRI channel .
Please help me to reslove this proem .
Following are the PRI configuration Parameter on CUCM .
Product Specific Configuration Layout
Line Coding : HDB3
Framing : NON CRC4
Clock : External
Input Gain (-6..14 db) 0
Output Attenuation (-6..14 db) 0
Echo Cancellation Enable
Echo Cancellation Coverage (ms) 64
PRI configuration on VG
interface Serial0/0/1:15
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
BR,
SANDIPAN

Hi Craig ,
Thanks for your reply .
We are using the full 30 channel E1 PRI .
following are PRI Channel Statistics:
%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 1. Layer 3 output may not apply
ISDN Se0/0/0:15, Channel [1-31]
Configured Isdn Interface (dsl) 1
   Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart 5=Maint_Pend)
    Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
    State   : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 3 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
   Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend 9=OOSPend)
    Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
    State   : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 2 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
Please find following debug error
VG_RO_01#isdn test call interface serial 0/0/0:15 09665484798
Mar 7 03:00:54.570: ISDN Se0/0/0:15 Q921: User RX <- RRp sapi=0 tei=0 nr=5
Mar 7 03:00:54.570: ISDN Se0/0/0:15 Q921: User TX -> RRp sapi=0 tei=0 nr=4
Mar 7 03:00:54.570: ISDN Se0/0/0:15 Q921: User TX -> RRf sapi=0 tei=0 nr=4
Mar 7 03:00:54.574: ISDN Se0/0/0:15 Q921: User RX <- RRf sapi=0 tei=0 nr=5
CCIL_PUNE_DR_VG_RO_01#
Mar 7 03:00:55.994: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Called num 09665484798
Mar 7 03:00:55.994: ISDN Se0/0/0:15 Q921: User TX -> INFO sapi=0 tei=0, ns=5 nr=4
Mar 7 03:00:55.994: ISDN Se0/0/0:15 Q931: SETUP pd = 8 callref = 0x0084
Bearer Capability i = 0x8890
Standard = CCITT
Transfer Capability = Unrestricted Digital
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Called Party Number i = 0x81, '09665484798'
Plan:ISDN, Type:Unknown
Mar 7 03:00:56.006: ISDN Se0/0/0:15 Q921: User RX <- RR sapi=0 tei=0 nr=6
Mar 7 03:00:56.018: ISDN Se0/0/0:15 Q921: User RX <- INFO sapi=0 tei=0, ns=4 nr=6
Mar 7 03:00:56.018: ISDN Se0/0/0:15 Q931: RELEASE_COMP pd = 8 callref = 0x8084
Cause i = 0x8286 - Channel unacceptable
Mar 7 03:00:56.022: ISDN Se0/0/0:15 Q921: User TX -> RR sapi=0 tei=0 nr=5
Mar 7 03:00:59.995: ISDN Se0/0/0:15 Q921: User TX -> INFO sapi=0 tei=0, ns=6 nr=5
Mar 7 03:00:59.995: ISDN Se0/0/0:15 Q931: SETUP pd = 8 callref = 0x0084
Bearer Capability i = 0x8890
Standard = CCITT
Transfer Capability = Unrestricted Digital
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Called Party Number i = 0x81, '09665484798'
Plan:ISDN, Type:Unknown
Mar 7 03:01:00.007: ISDN Se0/0/0:15 Q921: User RX <- RR sapi=0 tei=0 nr=7
Mar 7 03:01:00.019: ISDN Se0/0/0:15 Q921: User RX <- INFO sapi=0 tei=0, ns=5 nr=7
Mar 7 03:01:00.019: ISDN Se0/0/0:15 Q931: RELEASE_COMP pd = 8 callref = 0x8084
Cause i = 0x8286 - Channel unacceptable
Mar 7 03:01:00.019: ISDN Se0/0/0:15 Q921: User TX -> RR sapi=0 tei=0 nr=6
Mar 7 03:01:03.995: ISDN Se0/0/0:15 Q921: User TX -> INFO sapi=0 tei=0, ns=7 nr=6
Mar 7 03:01:03.995: ISDN Se0/0/0:15 Q931: RELEASE_COMP pd = 8 callref = 0x0084
Cause i = 0x80E6 - Recovery on timer expiry
Mar 7 03:01:03.995: ISDN Se0/0/0:15 **ERROR**: CCPCC_CallOrigination: SETUP timed-out (2nd T303) to NETWORK. The SETUP failed.
BR ,
SANDIPAN

MGCP T1 PRI shows Unregistered on CUCM

CUCM: 8.6.2.21900-5
Call Manager page shows PRI as unregistered, however on gateway ISDN status is Multiple Frame Established
ISDN Serial0/0/1:23 interface
dsl 1, interface ISDN Switchtype = primary-qsig
**** Slave side configuration ****
L2 Protocol = Q.921 0x0000 L3 Protocol(s) = CCM MANAGER 0x0003
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 1 CCBs = 0
The Free Channel Mask: 0x807FFFFF
Number of L2 Discards = 0, L2 Session ID = 39
Total Allocated ISDN CCBs = 0
Additional Information:
1. Calls go through the PRI 0/0/1 even if the CCM shows it unregistered
Troubleshooting Done:
1. Reset the gateway from Call manager and MGCP gateway
Resetting the gateway didn't make any difference.

Hi Amit,
This might be an issue with RISDC on the cluster. This service is responsible for distributing updates about device statuses in the cluster from one server to another. How many servers do you have on the cluster?
Does the gateway appear unregistered to ALL servers in the cluster? Can you take a look at this from the other subscribers?
You could try giving RISDC a reset on all subscribers first, followed by resetting it on the publisher. You will find this under Serviceability -> Control Center - Network services.
Thanks

Require to configure CUCM and Gateway router PRI for 200 users

Hi all,
My current scenario is like this:
We have two CUCM servers(7.1) in reduandancy. One is publisher and other is subscriber. We have a gateway router 2921 with a single PRI card. We have 200 IP Phone users. Currently the problem is we have only 30 telephone lines. So with the help of DID configuration. We assign thirty ports to thirty users. Now these thirty users are able to to make an outbound call and recieve an Inbound call. Now I also want to create a pattern in which we dedicate 10 channaels for other 120 users to make an outbound call. And restrict 50 users to not make an outbound call. So we require such a scenario.
Total 200 Users:
1. First 30 users. Able to make an outbound call and also able to recieve an inbound call.
2. Next 120 Users. Able to make an outbound call. But not able to recieve an inbound call because of PRI restriction. Because we only have a thirty numbers. Dedicate 10 channels for them.
3. Last 50 users. Not able to make an outbound call. And also not able to recieve an Inbound call.
So I need a configurations of Gateway router and PRI. And also of CUCM in which we could define such a pattern.
Regards,
Ali Raza

Ali,
The outbound call restrictions can be implemented using the CUCM Calling Search Space configurations. A basic example to illustrate:
Partition 1: internal_pt (only internal patterns)
Partition 2: phones_pt (ip phones on your network)
Partition 3: pstn_pt (off net patterns/external)
CSS1: Offnet_css
internal_pt
phones_pt
pstn_pt
CSS2: Onnet_css
internal_pt
phones_pt
You place all patterns that can reach off net in "pstn_pt". Phones that are using CSS2 cannot reach off net. The above example is leveraging the device CSS for all call routing decisions.Using this approach, you would assign "onnet_css" to phones that can only dial internally. NOTE: I am just using a basic example here and not suggesting you use this PT/CSS config "as is".
There is another approach where you "allow" all patterns on a device and "restrict" on the line. For example:
Partition 1: internal_pt (only internal patterns)
Partition 2: phones_pt (ip phones on your network)
Partition 3: pstn_pt (off net patterns/external)
Partition 4: block-pstn_pt (blocking patterns for pstn)
CSS1: AllPhones_css
internal_pt
phones_pt
pstn_pt
CSS2: restrict-pstn_css
block-pstn_pt
Again, pstn_pt contains all pstn patterns. The block-pstn_pt would also contain off net patterns. The difference is that in the block-pstn_pt all patterns would have the "Block this pattern" flag enabled. All phones would have a Device Level CSS of AllPhones_css. Phone lines where you wanted to restrict off net dialing would have a Line Level CSS of restrict-pstn_css.
Just a quick refresher, when a phone LINE goes off hook to dial the CUCM is using the "Line Level CSS" + "Device Level CSS" to make the routing decision. So, assume Line 1 on Phone A has restrict-pstn_css and the phone Device Level CSS is AllPhones_CSS. The Effective Search Space is:
1. block-pstn_pt
2. internal_pt
3. phones_pt
4. pstn_pt
So, if we assume that block-pstn_pt contains patterns that override patterns in pstn_pt then you can effectively block off net access to Phone A, Line 1.
Why would this be a good approach? Well, what if you had a need to restrict Line 1 on Phone A but allow Line 2 on Phone A. Using line level restrictions is much more flexible. Especially if you have more than 2 options you need to consider.
As far as inbound restrictions. If a phone line doesn't have a DID then you have achieved your objective.
The part I can't answer is reserving 10 specific channels for one group of users. I do not believe this is possible.
HTH.
Regards,
Bill

No-CRC4 framing option for PRI Configuration on cisco 3527 gateway

Hi,
Cisco 3527 videoconfrencing support only below mentioned framing options:-
CRC-4
Extended CRC4
Double framing.
PRI service provider support only no-CRC4. Is there any option to get to PRI with no-CRC4 framing option teminated on 3527.
Please find attached screenshot of the same.
Regards,
Abhas Jain

Place an ISR router in between 3525 and PRI circuits. Beside teh CRC issues, that will allow you to use PRI for other things beside videoconferencing only.

How to set the caller tune on PRI-Line/CUCM 8.X

Hi Team,
If passible to set the caller tune songs or advertising in CUCM 8.x -( in place of ringing).
How to set the caller tune for incoming calls( advertising), including 4 digit extension. This feature is available or not (cucm 8.0.3) Kindly verify and confirm
regards,
Johnson

Dear Team,
Now i have configured the Call Queuing, Its working fine, But how to verify to incoming calls or Missed call are in queuing, how to showing.
Connected call are showing, but unable show missed call like who ever in queue.
Regards,
Johnson.

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 10 publisher & subscriber.

Dears,
I have a publisher and subscriber CUCM 10.5 with 2 voice gateway within a single site, instead of subscriber sleeping always i want to make it work also, so i have thought that i will split the 1000 phones on pub and sub with 2 no's cucm groups and phones registered with pub will use VG1 and VG2 (MGCP gateway) and phones registered with SUB will also use VG1 and VG2 (MGCP gateway).
Is it this is a correct thoughts for design and what obstacles i can face in this design

Dear Manish,
Thanks for the reply,
tftp, for example the phones on 2nd and 3rd flr will hit the option 150 with primary as subscriber and phones on 1st nd 2nd flr will hit the option 150 with primary as a publisher.
MGCP gateway: Gateway configuration will be same for both of them as they will have redundant publisher and redundant subscriber.
I have 2 no's E1 PRI so i can add them in 2 different route groups ???
for example as below.
RG-PUB
port1--router 1
port 2---router 2
RG-SUB
port2--router2
port1-router1
and these will be called in the RL with route pattern.
Thanks

3527 PRI GW + GK + CUCM

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