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- About maqsood ahmed
09-04-2019
Stats
Member since
12-21-2009
115
Posts
5
Helpful
4
Solutions
08-07-2019
05:03 AM
Actually i have HP , IBM and UCS B200M2 server blades , using any one of this i need to install PCCE 12.0 in LAB Mode. i have no good documents to follow,
... View more
08-06-2019
06:51 AM
Dears, if any one help to complete the installation of PCCE 12 in Lab mode with Spec Based servers or UCS 220M2 servers. spec based servers like Hp or IBM blades. there is no clear documentation for LAB mode installation ..
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- Tags:
- PCCE 12.0
Labels:
08-06-2019
05:42 AM
Can u support from any document to install PCCE 12.0 in Spec based servers , i have HP and UCS B series 220 M2 blades . installation pcce 12.0 was sucessfull in 220 M2 servers , but could not select the deployment type LAB MODE,,,it prompts errors ..so i had to abort.
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Created by
maqsood ahmed
in IP Telephony and Phones
in IP Telephony and Phones
03-24-2019
06:15 AM
03-24-2019
06:15 AM
hi shiraz , would u post the relevant config from VG , of 2.0.1.0 TCL script working for sip trunk, , mine it seems some problem,,
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Created by
maqsood ahmed
in IP Telephony and Phones
in IP Telephony and Phones
04-09-2018
02:49 AM
04-09-2018
02:49 AM
I have a issue with hunt pilot call treatment and call forwarding !
The requirement is we need to have a message repeated infinite time until the external user hang up if the hunt members does not answer the call.
1. An External user Call dial into 9249001 thru sip trunk ( SIP-ServiceProvider) , 9249001 is CFA to Voicemail .
2. 9249001 is a CallHandler number in unity connection , which answer with greeting message and transfer it to 4041 , with release to phone system selected
4. 4041 is HuntPilot with 3 members in HuntList ,,which never answer the calls.
5. Under Hunt Call Treatment Settings ,, Forward Unanswered Calls to is selected , and 9249001 is the number provided. with proper CSS selected, THE CALL NEVER GOES THERE,,,,i hear silence for 5-10 sec and hangs up ..
I want this call to go back to VM greetings to play message again ,,,and come back to HP ,,,try the members again ...
Note:If I change to any DN apart from 9249001 , in Forward Unanswered Calls the call forwards and it works even from external , and also internally inside the cluster,,it is working.
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Labels:
02-12-2017
09:13 PM
hi Manish !
Stuck with this error ,, could not go ahead , tried with IE ,firefox, chrome,etc...
cisco has not released any new update since dec 2015 ,,, it seems this product is seldom used , i believe cisco has no R&D on this product !
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Created by
maqsood ahmed
in Collaboration Documents
in Collaboration Documents
01-31-2017
01:46 AM
01-31-2017
01:46 AM
iam looking for latest SDK which support phone model 9951, 9971 , 8691 ... i think SDK 7.1.(3) is supported ,,,but from i can download ... pls share link !
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01-27-2017
10:10 AM
New Release 2.1.2 - Webbridge is back with version 1.9 - OLD interface
From Release Note:
Cannot join a call via the Web RTC app using Firefox
Fixed in 2.1.2, by replacing Web Bridge 2.0 with Web Bridge 1.9. -
Unable to change passcode on spaces created using Web Admin interface
Fixed in 2.1.2, by replacing Web Bridge 2.0 with Web Bridge 1.9.
... View more
wow ! look's good ,,,, let
in Other Collaboration Subjects
01-27-2017
09:55 AM
01-27-2017
09:55 AM
wow ! look's good ,,,, let me try and get back to u !
... View more
I do also having the same
in Other Collaboration Subjects
01-27-2017
05:16 AM
01-27-2017
05:16 AM
I do also having the same issue ! co-space owner cannot change password !
i have additional question related to this issue ! How can we change the passcode for ALL the spaces created to some default passcode like 1234 ...
i need to set Passcode to all spaces thru API setting ,,,does any body know !
... View more
Just replying into my own
in Other Collaboration Subjects
01-27-2017
04:34 AM
01-27-2017
04:34 AM
Just replying into my own answer , kind of update ! recently i updated the cms to 2.1.1 ,,,,and lo.... NO support for Telepresence devices 1300 , 3200 and others
cisco in the release document has no support for certain TP devices ,,,and additionally they have changed the webRTC user interface ! which very much confusing to end-users ,,, hard time explaining the interface and way to conduct the meeting ,,, to know u r call-id got to deep into visitors after clicking few icons ,,,
cisco need to improve this interface , looks cools but confusing ....
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01-17-2017
03:55 AM
hi adarsh !
the below profile changes what i need ,, but it also changes if the call is coming from cucm as well ,,, effectively all calls will have 13301234 as the calling number !
but ,,,i need it should look at @example.com ,,if it found this domain ,,then change the only the first part of uri ! user1@example.com ,,,,,change to 13301234@example.com
still working on it !
request INVITE sip-header From modify "(<.*:)(.*@)" "\13301234@"
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01-04-2017
07:57 AM
thanks adarsh !
iam still not decided where i should implement it in CUCM or VoiceGateway !
i think CUCM normalization script is good place ,but it needs to modified in multiple places.. like in from and contact headers. And i have never done it before !
currently iam trying to study the " A guide to SIP Normalization" ,,,and pickup something to write ...
meanwhile is the below script i tried to make is this good ... i need to change only the calling name into number
M = {}
function M.inbound_INVITE(msg)
local orig_to = msg:getHeader("From")
local new_to = orig_to:gsub("user1", "852741")
msg:modifyHeader("From", new_to)
end
return M
or u have better one !
regards/
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Created by
maqsood ahmed
in Other Collaboration Subjects
in Other Collaboration Subjects
01-03-2017
10:29 PM
01-03-2017
10:29 PM
Dear experts !
i have a situation in CMS 2.1 , outbound calls. i need to route the external calls from CMA / webClient to SIP Gateway.
i see a invite coming into Sip Gateway with calling name uri. as shown below ,,, the SIP gateway will NOT route it to SIP ISP if it is not from the known DID number
INVITE sip:1238402@96.1.163.10:5060 SIP/2.0
From: "user1" <sip:user1@12.163.1.10>;
there is nothing i can do in SIP trunk connection from cucm to cms2.1..there is no provision to change the calling name uri to number
format in cucm.
how can i change the ( calling name) <sip:user1@12.163.1.10 >; to something like <sip:852741@12.163.1.10> in CMS 2.1.
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Labels:
12-29-2016
09:03 AM
Question NO 1:
Well , now it shows connected ,,but i don't see any CDR coming into the folder
actually , our webserver was not supporting POST Http method ,,after some tweak in IIS, POST method was accepted and CMS connected to CDR receiver..
NO 3: this is also good ,,,i just restarted the core server ...dont know how it happened ,,,just like microsoft ,,,a restart was needed.
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08-07-2019
Actually i have HP , IBM and UCS B200M2 server blades , using any one of
this i need to install PCCE...
08-06-2019
Dears, if any one help to complete the installation of PCCE 12 in Lab
mode with Spec Based servers o...
08-06-2019
Can u support from any document to install PCCE 12.0 in Spec based
servers , i have HP and UCS B ser...
Created by
maqsood ahmed
in Unified Communications Infrastructure
06-10-2019
06-10-2019
Even though i opened the TAC Case, they closed after working with me for
a month stating 9.1.2 is en...
Created by
maqsood ahmed
in IP Telephony and Phones
03-24-2019
03-24-2019
hi shiraz , would u post the relevant config from VG , of 2.0.1.0 TCL
script working for sip trunk, ...
Created by
maqsood ahmed
in TelePresence and Video Infrastructure
02-20-2019
02-20-2019
just i did it today , i will share my experience about renewing the
database certificates , hope u p...
Created by
maqsood ahmed
in Collaboration Applications
01-29-2019
01-29-2019
Android device is using Chrome Browser , and still the problem persist .
Created by
maqsood ahmed
in Collaboration Applications
01-20-2019
01-20-2019
When the invite message is selected , it connects to meeting server and
ask the guest name ,,once th...
Created by
maqsood ahmed
in Collaboration Applications
12-23-2018
12-23-2018
How do i maximize the Video Window in Jabber for windows when the call
is answered. i need this feat...
Created by
maqsood ahmed
in TelePresence and Video Infrastructure
12-18-2018
12-18-2018
I am trying to register DX80 to VCS-Expressway , i received this below
error in vcs-E ,,, 8061102 is...
Created by
maqsood ahmed
in IP Telephony and Phones
04-09-2018
04-09-2018
I have a issue with hunt pilot call treatment and call forwarding ! The
requirement is we need to ha...
Created by
maqsood ahmed
in TelePresence and Video Infrastructure
12-17-2017
12-17-2017
upgrade to 2.2.9 resolve the issue !
Created by
maqsood ahmed
in TelePresence and Video Infrastructure
12-08-2017
12-08-2017
My deployment is showing Tcp_failure and TLS handshake error ,,and also
when the command " xmpp clus...
Created by
maqsood ahmed
in TelePresence and Video Infrastructure
12-03-2017
12-03-2017
just to update the question! xmpp cluster trust 3CBcerts.CRT xmpp
cluster trust 3XmPPCerts.crt .... ...
Created by
maqsood ahmed
in TelePresence and Video Infrastructure
12-02-2017
12-02-2017
Yes, indeed we did restart the whole cms servers , and also we did RESET
the two xmpp server ,except...
Public Statistics
| Date Registered | 12-21-2009 11:06 PM |
| Date Last Visited | 09-04-2019 01:41 AM |
| Total Messages Posted | 115 |
| Total Helpful Votes Received | 5 |
