Jabber cannot Call - Integration with CUCM/AD
Dear all,
I installed CUCM 9.1.1 and CUPS 9.1.1 with Jabber client 9.0.1.
when I run Jabber on a joined domain PC (login with AD username and password), there are no problem with the calling ability.
But when I try to use a non-join domain PC (I am using VPN to connect to office network and using my personal PC), only the chatting feature that are available. I cannot loggin to phone accounts and cannot make any calls with my Jabber client.
Should I use UDS integration with AD?
Thanks,
Hasan
Hi Hasan,
Can you take a look on this thread:
https://supportforums.cisco.com/message/3914353#3914353
If you still have problem connecting, can you try with newer version of Jabber?
Regards,
Srdjan
Similar Messages
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Jabber for Windows - without LDAP integrated with CUCM Jabber UDS - NO PEOPLE CAN SEARCH
Hi all Jabber Experts,
I have the CUCM, which is the versin 8.6 and the Presence Server, which is the version 8.6, that is not integrated the LDAP, but I want to deploy the Jabber for Windows.
So I would use the UDS to deploy the Jabber for Windows (modified the XML and uploaded to the CUCM TFTP server).
Finally, that can login the users, which is manually added from CUCM.
But I cannot search other users from the Bubby List. Any idea for that?First of all, either you use CUCM 8.6 with CUPS 8.6, or you use CUCM 9.1 with IM&P 9.1, what you're mentioning is just impossible as they're not compatible and that's not supported.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk -
Cisco TS 320 integration with CUCM
Dear Expert,
Is there any way we can integrate the TS 320 with CUCM without any conductor.
Means by SIP trunk or as Conference bridge.
BRI have TS on Media 320, so its means we can not use it as locally managed.
The TelePresence Server on Media 310/320 and Cisco TelePresence Server on Virtual Machine do not support locally managed mode; on these platforms, the
TelePresence Server must be managed by a system like Cisco TelePresence Conductor or Cisco TelePresence Exchange System.
BR -
Softphone Registration Error - Jabber 9.2.5 with CUCM/CUPS 8.6
All,
I'm having trouble getting my softphone feature inside Jabber 9.2.5 to work with my CUCM/CUPS 8.6 deployment. I have the CCMIP profile setup and user associations in place. The show connection status indicates a status of Not Connected with an error of "Connection error. Ensure the server information in the Phone Services tab on the Options window is correct. Contact you system administrator for assistance. The CCMCIP address is correct.
Any ideas?
I verified the service is running in CUCM. The Jabber client appears to never have attempted to register to CUCM. Status of the endpoint in CUCM is unknown, unknown.
Thanks!
JasonSorry for the delay in getting back to you. You can determine what CUCM server "name" the server is trying to use by doing a TFTP GET of the device config file from a workstation with access to the TFTP server.
An example from a windows box is
tftp -i GET .cnf.xml
This will pull a copy of the device's config file to the pc. You can then open it with any text editor and look for the CUCM name. It is most likely the hostname of the subscriber in question. If you can't resolve the name by itself (no fqdn) then the reg will fail. At least it did in my case.
You can change the server name to IP by browsing to System > Server in CUCM. Change the device in question name to the correct IP.
Alternatively you could correct the name resolution issue.
HTH,
Jason -
Need Urgent Help on Meeting Place Integration with CUCM 7.1 and AS5400 PSTN Gateway
Hi,
This is first time I am on this forum.
I have already tried going through a lot of docs on docwiki.cisco.com but couldn't find complete configuration help.
I have to integrate Meeting Place 8.X with an existing CUCM and an E1 gateway (PSTN Gateway) AS5400.
The CUCM is already part of a Telepresence Environment. I need to create a SIP trunk between AS5400 and CUCM 7.1 and then create a Trunk between AS5400 and Cisco Unified MP 8.X and then between CUMP and CUCM.
I need help on AS5400 SIP Configs as well as parameters I need to cover on CUCM (Though I have done some basic dial-peer configs but they haven't been of much help).
Then I also need help on AS5400 SIP configs with CUMP 8.0
Any docs on Integration between CUMP and TP3000 will be of great help too.
Rgds,
AsimI can get the Ricoh to register as sip endpoint, it answeres then imediatly disconnects. Doing a monitor with Wireshark looks like it attempts to negotiate t38 but fails. Any idea why this fails?
|160.260684000| INVITE SDP (g711U) | |SIP From:
| |(5060) ------------------> (5060) | |
|160.338806000| INVITE SDP (t38) | |SIP Request
| |(5060) <------------------ (63435) | |
|160.339545000| 491 Request Pending | |SIP Status
| |(5060) ------------------> (5060) | |
|160.547894000| 406 Not Acceptable | |SIP Status
| |(5060) <------------------ (63435) | | -
LDAP Integration with CUCM 9.0
We would like to use LDAP to sync all of our users from Active Directory. All of our current CM Users are local, the problem is that they have the same user names as our Active Directory users. From what I understand this is going to be a problem because:
"If accounts from LDAP match an existing Unified CM account that is not marked as an LDAP synchronized account, then these accounts are ignored."
Does that mean we will have to delete all our existing CM users in order to sync the LDAP users correctly? Is there a best practice for this? Once we syncronize the LDAP users how to I ensure that the user gets associated with the proper phone? Or do I have to visit each user individually?I just did a quick test for this, my lab CUCM 9 is already LDAP integrated, but I created a local user, then I created that same local user in my LDAP OU, and performed a full sync.
The user is no longer showing as a local active user, but as an active LDAP synchronized user.
Which was my thought, there's only one conversion, from LDAP to local.
The behavior is just as with any previous release, local users who match an LDAP user after you enable it, are just updated, and kept with all their configurations.
I checked the option to turn it back again into a local user, did a full sync, and it's again an active LDAP user.
HTH
java
if this helps, please rate
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Cannot call functions with this[]
Hello,
I'm in trouble because I need to call functions from MCs with
a variable
name.
I tried to use this[] that should do the trick, but it
doesn't.
Here's what I did in code to try to debug :
- trace what's inside the this[]
- the actual this[]
- a cut and paste of the above trace() with the function call
Of course, both "hardcoded" calls work..; what am I doing
wrong ?
Here's the code :
trace(_parent + "." + buttonName);
trace("_parent." + buttonName);
this[_parent + "." + buttonName].setBtnState("test1");
this["_parent." + buttonName].setBtnState("test2");
_parent.mcBtn1.setBtnState("test3");
_level0.mcInt.mcBtn1.setBtnState("test4");
What am I supposed to feed this[] with ? Path as a string ?
Thanks in advance.
PJWhat you need to feed the array operator is a string or a
variable that can be resolved to a string. And that string needs to
be the sole name of an instance or property of something that can
be found in that object.
So the reason the ones with the "_parent.mcMe" didn't work is
because there is no object with the name "_parent.mcMe" inside of
this. There is a _parent object and inside the parent object there
is an mcMe, but that isn't the same thing.So you need to pick out
the path one item at a time.
this["_parent"]["mcMe"]["testMe"]();
Should also work. And any of those string litterals could be
replaced by a variable which held a string as well. Also notice
that the function at the end can also be referenced since you are
looking inside the mcMe object for some object with a name of
testMe. But the parens which call the function need to be outside
the array access because they aren't part of the name of the
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Contact Center Ent. Integration with CUCM
I have CUCM 8.0 and CCX 8.0 and i am trying to setup the integration between them.
I have created a user on the CUCM and CTI Port and done the required configuration on the CCX.
Under CTI Port Status & IP Address it is giving unknown.
I have attached the configuration on CUCM and CCX.
Any help will be highly appreciated.this is wrong , they have to be created on the UCCX , the UCCX will automaticly creat them on the CUCM.
check the link below :
http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_8_0/configuration/guide/uccx801ag.pdf
Amer -
Cisco BE6k with CUCM 10.X integration with Siemens with Hipath 3350 and 3550
Dear Experts;
We have one requirment from customer.
They have 4 branches with Siemens Hipath 3350 and 3550 PBX and they want to integrate with cisco CUCM 10.x.
The Head Quater with complete CUCM as IPT and branches with with Hipath 3350 and 3550 PBX.
CIsco TAC says interoperability doc is not available and also I can see tehre is no doc.
TAC Reply As below.
We do not have any documents about CUCM integration with Hipath 3350 and 3550 on our interoperability portal. This means that their integration with CUCM was never properly tested. So we can not confirm that all features will be working. But from CUCM perspective this is third-party device that can be integrated using SIP or ISDN (with Voice Gateway between them).
Link for interoperability portal:
http://www.cisco.com/c/en/us/solutions/enterprise/interoperability-portal/networking_solutions_products_genericcontent0900aecd805b561d.html#callmgr10
Regards
DebashisIf the Siemens supports SIP or H323 you can attempt to simply create a trunk between the systems and route calls via it. If IP integration is not an option then as pointed by TAC integrate via PRI circuit (preferably QSIG). From CUCM side it will be just another GW, similarly from Siemens side. You'll need to select which side is going to be ISDN network (I always use Cisco side) and then make the other side user.
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Oracle BI Publisher 10x/11x Integration with Siebel CRM 8.2 Release
Hello Gurus,
We are in the process of implementing Siebel (8.2) Public Sector for one of our client and in term s of reporting we have decided to use latest version of OBIEE and BI Publisher and thus decided to go ahead with implementing BI Apps 7.9.6.3 combined package of (OBIEE 11.1.1.6, BI Publisher, Informatica 9.x, DAC 10.1.3.4.1 with hot fix).
There is My Oracle Support (MOS) article # 1172844.1 (titled ‘Supporting Documentation for Siebel Reports by Product Version’) which explains that Oracle BI Publisher 10.1.3.4.2 version supports Siebel CRM release 8.2.
Due to the limitations of the Oracle BI Publisher usage as per the Oracle support documentation, we need to make use of OBIEE 10g version for BI Publisher for integrating with Siebel 8.2 release and we need to also use OBIEE 11g for the main reporting and henceforth we end up maintaining 2 servers for reporting in our environment.
Does Oracle BI Publisher 11.1.1.5 or 11.1.1.6 version cannot be integrated with Siebel 8.2? Is this not supported version with Seibel 8.2 version?
Oracle BI Publisher architecture in 10g is different when compared to OBIEE 11g version and does it make any impact if we go ahead with 2 different versions pointing to pull the data from the OLTP source ( Siebel) ?
I am looking out for exact approach to be followed for reporting requirements for this kind of scenario ?
Help me with your inputs and suggestions on this.
Thanks
Praveenhi Praveen,
Did you got any answer for this?
Cheers,
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Hp 35s integrating with complex numbers/functions
Are there any examples of how to integrate a complex function for
the HP 35s..the manual suggests its possible..but, I can not find one
example...anywhere... I am just interested in testing.....I do
not want to break things into parts etc...I just want to try
e^z for example, where z is complex..any help?
I have tried xiy, rTheta, x+iy...syntax errors or non existent..funnyHi,
The manual states that complex numbers can be used with EXP(x), but it does not mention anything about using complex numbers for integration. Edit: I missed some text in the manual, see below.
As far as I can tell it cannot do integration with complex numbers. (The 50g can do this.)
However, you may get a more definitive answer if you ask your question here:
http://www.hpmuseum.org/cgi-sys/cgiwrap/hpmuseum/forum.cgi
EDIT: that forum has moved to a new version:
http://www.hpmuseum.org/forum/forum-4.html
Note: I do not work for HP, I just like playing with calculators :-) -
CWMS v.2 - how to configure CWMS to authenticate user with CUCM
Hi,
I have a CUCM with no LDAP or AD integration. I already configured the directory integration with CUCM and it synchronized the user accounts to CWMS. When trying to login with end user account, password configured in CUCM doesn't work. What is the process to configure CWMS to authenticate with CUCM user database? Thanks.
-AlanHi Alan,
CUCM and LDAP integration is a prerequisite for using Directory Integration on CWMS.
http://www.cisco.com/c/en/us/td/docs/collaboration/CWMS/1_5/Administration_Guide/Administration_Guide_chapter_01011.html#task_DB0D271D6EB1459EB4DA269461E93B36
Before You Begin
You must configure AXL and LDAP directory service on CUCM before you can use the directory integration feature. CUCM is required to import users into your Cisco WebEx Meetings Server system. Use CUCM to do the following:
Enable Cisco AXL Web Service
Enable Cisco directory synchronization
Configure LDAP integration
Configure LDAP authentication
-Dejan -
Jabber cannot do voice calls to PSTN or to other jabber user
Hi,
I'm trying to troubleshoot why jabber for windows 9.7.2 is not able to establish calls to PSTN dialed by DN and to other jabber user called by URI, not DN.
We have CUCM 10 with IM and Presence 10. IM and Presence is newly installed. I have created 2 x Cisco Unified Client Services Framework devices for 2 users. Users are able to connect through jabber for windows, IM, do direct DN-to-DN calls (including video), but not able to call outside to PSTN and by URI to other users.
Cases:
- jabber for windows - call DN to DN between internal users - working with video
- jabber for windows - call DN to PSTN through SIP-to-SIP connection (SIP trunk to GW, SIP trunk from GW to ISP) - not working CUCM returning me Q.850 cause 65 to jabber immediately when other side pick up phone, even jabber hear ringing with music from ISP side. - Cause: 65(0x41)[Bearer capability not implemented]
- jabber for windows - in search field I search for second user, I can see presence, when I click call button, call is immediately dropped - don't know reason still not troubleshooted.
communication between GW 2801 and ISP is OK, they are acknowledge codec and everything is fine.
EDIT: I tried to connect jabber for windows directly to CUCM TFTP servers without IM and Presense server. I have same behavior, when outgoing call is picked up it is immediately dropped by CUCM and send Q.850 cause 65 to jabber.
EDIT2: any other SIP device configured same way as jabber CSF is working. just jabber not.
EDIT3: when I selected "Media Termination Point required" on jabber device in CUCM, then voice is working normally. But if possible I do not want this and make voice without MTP. How?
THanksI found it, i have enabled "Allow Presentation Sharing using BFCP" on SIP profile assigned to trunk pointing to voice gateway. This causes that SDP from gateway didn't come.
-
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 -
Hello All,
I can Call from any Branch office to CUCM through Gatekeeper. But I cannot call from CUCM through Gatekeeper to any Branch.
My CUCM version is 9.1.2
Regards
BahlulDo you see anything shw up on the debugs on the branch router?
Check debug h225 asn1 to see if there is a setup coming from the CUCM. If not, that means the CUCM is not receiving the IP address of the branch gateway from the GK. This could mean a config issue on the GK with respect to the branch gateways, or on the UCM.
Take debug gatekeeper main 10 from the gatekeeper. These debugs will show what's happening on the GK while processing the incoming ARQ from CUCM.
Also please upload the config of the GK, branch gateway here.
Hantale
Sree
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How do you set up a yahoo email account on my iMac ?
How do I set up a yahoo email account on my imac?