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- About Zekeria Sammantar
04-14-2019
Zekeria Sammantar
Member since
09-02-2012
Participant
Personal Web Page : http://sa.linkedin.com/pub/zekeria-sammantar/62/685/aa4/
Created by
Zekeria Sammantar
in IP Telephony and Phones
in IP Telephony and Phones
11-21-2018
10:20 PM
11-21-2018
10:20 PM
it works, I'm looking for ghosts.
Thanks
... View more
Created by
Zekeria Sammantar
in IP Telephony and Phones
in IP Telephony and Phones
11-21-2018
10:00 AM
11-21-2018
10:00 AM
TP PT: UCCX_Agent_xTranslate
Device CSS: None (For all ip-phones)
... View more
Created by
Zekeria Sammantar
in IP Telephony and Phones
in IP Telephony and Phones
11-21-2018
09:55 AM
11-21-2018
09:55 AM
TP rule: Pattern: 4XXX CSS: UCCX_Agent Urgent Priority: CHECKED
IP-Phone-1: 4012
IP-Phone-2: 4017
Line PT: UCCX_Agent
Line CSS: UCCX_Agent
UCCX_Agent :(CSS) UCCX_Agent UCCX_Agent_xTranslate
///////////////////////////////////////////////
IP-Phone-3: 4020
Line PT: Internal_PT
Line CSS: Mobile_CSS
Mobile_CSS:
Internal_PT
Mobile_CSS
UCCX_Agent
... View more
Created by
Zekeria Sammantar
in IP Telephony and Phones
in IP Telephony and Phones
11-21-2018
07:57 AM
11-21-2018
07:57 AM
even when its not marked urgent it matches, i also tried the 4XXX TP with and without the urgent priority and it was still matching TP which has 1000 options instead of the 4017 DN which is only 1 Option. CUCM uses closest-match logic for call routing, and as far as I remember the if a DN is created register or otherwise it be considered a potential match the same as TP or RP.
... View more
Created by
Zekeria Sammantar
in IP Telephony and Phones
in IP Telephony and Phones
11-21-2018
07:30 AM
11-21-2018
07:30 AM
I'm using the calling party transformation mask set to: 5000
0! TP, matches with outgoing RP with the access code 0.
I'm confused as there is no DN or TP/RP being with "401"
"CUCM won’t accept user input after matching a TP."
I cant find what is matching, as it doesn't have access to Partition with access to DN in the 410X Range.
... View more
Created by
Zekeria Sammantar
in IP Telephony and Phones
in IP Telephony and Phones
11-21-2018
04:58 AM
11-21-2018
04:58 AM
I use urgent priority because it needs to match RP without a T302 timer. 401X - IP-Phone agents 40[2-9]X - regular IP-Phone Agents when dialing regular phones need to numbers transformed to 5000 and show original DN when dialing other agents. I have resolved it by changing the TP to XXXX instead. I just want to know why it's not find a match with or without the urgent priority.
... View more
Created by
Zekeria Sammantar
in IP Telephony and Phones
in IP Telephony and Phones
11-21-2018
03:19 AM
11-21-2018
03:19 AM
I already did it matches and the call work also i enter the number full before i dial it works, this problem happens when dialing number by number.
En block dialing on other phones works.
... View more
Created by
Zekeria Sammantar
in IP Telephony and Phones
in IP Telephony and Phones
11-21-2018
02:46 AM
11-21-2018
02:46 AM
I'm curious to find out why the below call fails, this is whats configured:
TP rule: Pattern: ! CSS: Mobile_Test Urgent Priority: CHECKED
IP-Phone-1: 4012 (UCCX_Agent) (UCCX_Agent_xTranslate) IP-Phone-2: 4017 (UCCX_Agent) (UCCX_Agent_xTranslate)
CSS-Agent: UCCX_Agent UCCX_Agent_xTranslate
IP-Phone-1 dials IP-Phone-2 "401" before he finishes, he gets a reorder tone.
After some checking I found the only reason he can dial 401 to being with is that TP CSS has access to other DN's with starting DN "40" but not to 4017.
Urgent Priority is some to only avoid T302 when there is a mutliple matches and not to force routing. Also when Urgent Priority is checked on the 4017 DN, same thing happens.
Currently I avoid matching by using 0! instead of !.
Can someone tell me why this is?
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200477-Urgent-Priority-Configuration-on-Directo.html
Logs are attached.
... View more
Created by
Zekeria Sammantar
in IP Telephony and Phones
in IP Telephony and Phones
11-14-2018
02:27 AM
11-14-2018
02:27 AM
CUCM 11.5 with CUBE.
User B: Call Forward All to 1-800-273-8255
User A: (1000) calls User B:(2000)
Redirecting Diversion Header Delivery - Outbound ( Checked)
Calling Party Selection: Originator
Calling Number is: User-A(DID MASK).
Everything is fine until then.
External PSTN User: (1-800-273-8255) calls User B:(2000)
Call fails, from header sent to ITSP is my mobile number.
I understand this is normal scenario and I would like to avoid the CUCM changes.
FROM CUCM:
Received: INVITE sip:18002738255@GATEWAY-IP:5060 SIP/2.0 Via: SIP/2.0/TCP CUCM-IP:5060;branch=z9hG4bK3a04667120917 From: <sip:2025550191(MY_MOBILE)@CUCM-IP>;tag=2359521~e4075e8e-2Db3-4f16-b7d0-8e1642c64577-3340579279 To: <sip:18002738255(CFA:#)@GATEWAY-IP> Date: Wed, 14 Nov 2018 10:01:51 GMT Call-ID: 4d4a5700-beb1f28f-330d8-b15a050a@CUCM-IP Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM11.5 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Allow-Events: presence Supported: X-cisco-srtp-fallback Supported: Geolocation Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED Session-ID: 222222222222222;remote=00000000000000000000000000000000 Cisco-Guid: 129647175dd68-00000f65536-0000022955-2975466762 Session-Expires: 1800 Diversion: <sip:2000@CUCM-IP>;reason=unconditional;privacy=off;screen=yes P-Asserted-Identity: <sip:2025550191@CUCM-IP> Remote-Party-ID: <sip:2025550191@CUCM-IP>;party=calling;screen=yes;privacy=off Contact: <sip:2025550191@CUCM-IP:5060;transport=tcp> Max-Forwards: 67 Content-Length: 0
ITSP doesn't support Diversion header so I have to remove it on the outgoing dial-peer to ITSP.
Options(at least I think)
If SIP INVITE from CUCM contains Diversion header then:
Its doesn't match any extension that are only 4 Digits in the from field From: <sip:XXXX>
Copy the diversion header and replace the From header:
Diversion: <sip:2000@CUCM-IP> replace the From:<sip:2025550191(MY_MOBILE)@CUCM-IP> with From: <sip:701200@Gateway-IP>
Remove Diversion Header
If this can be done in the cucm with lua script, I'm ready to try it.
Thanks.
Edit:
I half way there, I can now Copy the diversion header and place into the From header with a mask.
voice class sip-profiles 1 request INVITE sip-header Diversion copy "<sip:(.*)@.*" u01 request INVITE sip-header From copy ".*<sip:(.*)@.*" u02 request INVITE sip-header From modify "<sip:@" "<sip:\u02@" request INVITE sip-header From modify "(.*)<sip:.*@(.*)" "\1<sip:686\u01@\2" request ANY sip-header Diversion remove
Output:
Received: INVITE sip:18002738255@GATEWAY-IP:5060 SIP/2.0 Via: SIP/2.0/TCP CUCM-IP:5060;branch=z9hG4bK3a04667120917 From: <sip:6862000@CUCM-IP>;tag=2359521~e4075e8e-2Db3-4f16-b7d0-8e1642c64577-3340579279 To: <sip:18002738255(CFA:#)@GATEWAY-IP> Date: Wed, 14 Nov 2018 10:01:51 GMT Call-ID: 4d4a5700-beb1f28f-330d8-b15a050a@CUCM-IP Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM11.5 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Allow-Events: presence Supported: X-cisco-srtp-fallback Supported: Geolocation Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED Session-ID: 222222222222222;remote=00000000000000000000000000000000 Cisco-Guid: 129647175dd68-00000f65536-0000022955-2975466762 Session-Expires: 1800 P-Asserted-Identity: <sip:2025550191@CUCM-IP> Remote-Party-ID: <sip:2025550191@CUCM-IP>;party=calling;screen=yes;privacy=off Contact: <sip:2025550191@CUCM-IP:5060;transport=tcp> Max-Forwards: 67 Content-Length: 0
I only need it to match Invite Messages with the From header is not 4 Digits or match and replace if its more than 4 Digits.
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Labels:
10-14-2018
06:09 AM
CUCM Publisher was restored using vmware snapshots to an earlier period which leads to phones being removed from CUCM everytime publisher is connected to the network. Subscriber was untouched and has the latest added phones.
I'm hoping to avoid using to reinstall the Publisher and sync it off the Sub.
Is there a way for the Publisher to sync of the Subscriber?
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Labels:
12-04-2017
10:26 AM
I have a over 200 of SIP Phones registered in Branch with
C3945E-CME-SRST/K9 ISR G2
If i dont configure SIP SRST and only SCCP SRST, will the phones convert to SCCP when the CUCM is unreachable.
This is because the ISR G2 doesnt support CUBE with SRST SIP.
If they convert to SCCP, I will not add SCCP SRST as the nightmare of rejections from CUCM when the WAN link is restored is just to much.
Thanks for the help.
... View more
Labels:
Created by
Zekeria Sammantar
in IP Telephony and Phones
in IP Telephony and Phones
12-04-2017
10:22 AM
12-04-2017
10:22 AM
I was almost going to implement this on C3945E-CME-SRST/K9 ISR G2 but I'm awaiting confirmation from this Cisco AM as it already December and I cant find it listed as a new feature on the latest SRST version. Thanks for helping me dodge the bullet.
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01-10-2017
05:56 AM
Kemal
This is not possible as because jabber client use the email@domain supplied by the user to do a dns srv lookup, so you have to publish the domain your users are using to login to jabber.
Hope this helps.
... View more
07-23-2016
11:25 PM
Polycom ---> CUCM ---> vTPS.
Problem:
Polycom to UCM:
From: <sip:IT%20Group%20Room@Polycom-DMA>;tag=????
UCM to Conductor:
From: <sip:IT%20Group%20Room@CUCM-IP>;tag=?????
To: <sip:Conference-Alias@vTPS>
vTPs:
Found "To" = "<sip:IT%20Group%20Room@CUCM-IP>;tag=????"
The Polycom System is adding %20 to every space that is empty, I need help removing it from the Inboud Invite:
M = {}
function M.outbound_INVITE(msg)
local FROM= msg:getHeader("From")
if string.matc("%20")
then
-- Removing %20 and replace with space?.
else
Thanks for the help
... View more
- Tags:
- #lua #cucmscript
Labels:
Created by
Zekeria Sammantar
in IP Telephony and Phones
in IP Telephony and Phones
08-31-2015
12:24 AM
08-31-2015
12:24 AM
Is there a device pack for cucm 5.1 or not. Many thanks for the quick support.
... View more
Created by
Zekeria Sammantar
in TelePresence and Video Infrastructure
02-11-2019 01:59 AM
02-11-2019 01:59 AM
Conductor XC4.2 TMS 15.1 TMPSE 1.9 Everytime I login to Smart Scheduler
using a AD account(aztest@ex...
Created by
Zekeria Sammantar
in IP Telephony and Phones
11-21-2018 10:20 PM
11-21-2018 10:20 PM
it works, I'm looking for ghosts. Thanks
Created by
Zekeria Sammantar
in IP Telephony and Phones
11-21-2018 10:00 AM
11-21-2018 10:00 AM
TP PT: UCCX_Agent_xTranslate Device CSS: None (For all ip-phones)
Created by
Zekeria Sammantar
in IP Telephony and Phones
11-21-2018 09:55 AM
11-21-2018 09:55 AM
TP rule:Pattern: 4XXXCSS: UCCX_AgentUrgent Priority: CHECKED IP-Phone-1:
4012 IP-Phone-2: 4017 Line ...
Created by
Zekeria Sammantar
in IP Telephony and Phones
11-21-2018 07:57 AM
11-21-2018 07:57 AM
even when its not marked urgent it matches, i also tried the 4XXX TP
with and without the urgent pri...
Created by
Zekeria Sammantar
in IP Telephony and Phones
11-21-2018 07:30 AM
11-21-2018 07:30 AM
I'm using the calling party transformation mask set to: 5000 0! TP,
matches with outgoing RP with th...
Created by
Zekeria Sammantar
in IP Telephony and Phones
11-21-2018 04:58 AM
11-21-2018 04:58 AM
I use urgent priority because it needs to match RP without a T302
timer.401X - IP-Phone agents40[2-9...
Created by
Zekeria Sammantar
in IP Telephony and Phones
11-21-2018 03:19 AM
11-21-2018 03:19 AM
I already did it matches and the call work also i enter the number full
before i dial it works, this...
Created by
Zekeria Sammantar
in IP Telephony and Phones
11-21-2018 02:46 AM
11-21-2018 02:46 AM
I'm curious to find out why the below call fails, this is whats
configured: TP rule:Pattern: !CSS: M...
Created by
Zekeria Sammantar
in IP Telephony and Phones
11-14-2018 02:27 AM
11-14-2018 02:27 AM
CUCM 11.5 with CUBE. User B: Call Forward All to 1-800-273-8255 User A:
(1000) calls User B:(2000) R...
Created by
Zekeria Sammantar
in TelePresence and Video Infrastructure
10-24-2018 12:18 PM
10-24-2018 12:18 PM
Can I use a wildcard certificate? CN: tms.example.com SANL:
*.example.com
Created by
Zekeria Sammantar
in TelePresence and Video Infrastructure
10-24-2018 01:57 AM
10-24-2018 01:57 AM
I need confirmation if the CTI-410V will support vTPS 46 vCPU.
Created by
Zekeria Sammantar
in TelePresence and Video Infrastructure
10-24-2018 01:56 AM
10-24-2018 01:56 AM
I'm deploying TMS 15.4 with NLB The certificate part does't make sense
to me, its says in the doc: "...
Created by
Zekeria Sammantar
in TelePresence and Video Infrastructure
10-24-2018 01:55 AM
10-24-2018 01:55 AM
I'm deploying TMS 15.4 with NLB The certificate part does't make sense
to me, its says in the doc: "...
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