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- About Kasiraman S
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Member since
01-23-2010
119
Posts
80
Helpful
13
Solutions
11-06-2015
03:22 AM
Its not incomplete entry. Static NAT translation in the Running-config shows like that in NAT translation table.
Example:
ip nat inside source static 10.10.10.4 172.16.6.14
router# show ip nat translation
Pro Inside global Inside local Outside local Outside global
--- 172.16.6.14 10.10.10.4 --- ---
Thanks,
Kasi
... View more
11-05-2015
02:36 AM
This is very much possible that the Voice Gateway which are facing internet get these kind of Toll Fraud calls and the built in Toll Fraud app is the best way to block those calls and other methods we can use are ACL which will drop those calls as soon those enter the Router which is really old method and they are not suggested these days.
Please take a look at the below URL to know about the Toll-Fraud Prevention Feature in IOS Release 15.1(2)T and later.
http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/112083-tollfraud-ios.html
Thanks,
Kasi
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11-05-2015
01:37 AM
Yes, you are correct and Switch will not push 30W.
Regarding the VLAN setting you can refer the below URL for 7962 and 42 Phones and you can also refer the Administration Guide for the model of the phone you are interested.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7962g_7942g/6_0/english/administration/guide/7962G-Admin-Book-Wrapper/7962net.html#wp1031028
Thanks,
Kasi
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11-05-2015
01:21 AM
Seems like Toll Fraud attempts to your GW. I see that call was blocked by TOLLFRAUD_APP in IOS and you do not have to worry :) *Nov 5 08:43:12.547 GMT: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search: Try with the demoted called number 900442476980898 *Nov 5 08:43:12.547 GMT: //2042/1536D1D3A2B3/CCAPI/ccCallSetContext: Context=0x21021E2C *Nov 5 08:43:12.547 GMT: //2042/1536D1D3A2B3/CCAPI/cc_process_call_setup_ind: >>>>CCAPI handed cid 2042 with tag 0 to app "_ManagedAppProcess_TOLLFRAUD_APP" *Nov 5 08:43:12.547 GMT: //2042/1536D1D3A2B3/CCAPI/ccCallDisconnect: Cause Value=21, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0) *Nov 5 08:43:12.547 GMT: //2042/1536D1D3A2B3/CCAPI/ccCallDisconnect: Cause Value=21, Call Entry(Responsed=TRUE, Cause Value=21)
Thanks,
Kasi
... View more
Created by
Kasiraman S
in IP Telephony and Phones
in IP Telephony and Phones
11-04-2015
07:45 AM
11-04-2015
07:45 AM
I do not see anything unusual with the config.
Execute "show ip address trusted list" and ensure it is Admin/Operationally UP and also check only the known IPs are in the trust list.
If you still find those calls, provide us the "debug voice ccapi inout" output.
Thanks,
Kasi
... View more
Created by
Kasiraman S
in IP Telephony and Phones
in IP Telephony and Phones
11-04-2015
02:20 AM
11-04-2015
02:20 AM
This does not work. Call park feature is intended to park the active call in a available park slot but the one you are trying to park is not answered. I tried the same before with different version but this did not work.
Workaround is to have Unity or UCCX answer the original call and then have them system transfer the call back to the park slot. You can confiure it in such a way you do not find any delay in parking the call and it works fine.
Thanks,
Kasi
... View more
Created by
Kasiraman S
in IP Telephony and Phones
in IP Telephony and Phones
11-03-2015
03:29 AM
11-03-2015
03:29 AM
Its good to consider enabling SIP OPTIONS PING under SIP Profile if not already, which helps not to route the call to the node which is considered to be down.
Use OPTIONS Ping to dynamically track the state of each destination IP address on each SIP trunk and the collective state of the trunk as a whole. If a destination address is unreachable, Unified CM will not extend calls to this device. When all destinations are unreachable, the SIP trunk is considered to be out-of-service.
Thanks,
Kasi
... View more
11-03-2015
01:30 AM
+5 Deji. Well explained. I believe your answer explains why only codec pass-through is configured.
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11-03-2015
01:04 AM
I believe we can consider taking a call which uses RSVP based CAC, which does require MTP for bandwidth reservation but it does not terminate/reoriginate the call.
And codec pass-through is mandatory for such software based MTP which act as a RSVP agent.
At present, I could not create this in a lab environment but I will see if I can try lab it out and get back.
Thanks,
Kasi
... View more
11-03-2015
12:14 AM
Hi Sanjay, Let me try to explain here. Lets say for an example you have configured this dspfarm profile for MTP which might be acting as a RSVP agent which is only needed to reserve the bandwidth but it does not modify the RTP stream in anyway, i.e does not terminate and re-originate the media, where RTP just work in a flow through mode and it does require codec pass-through command. Another example, we configure codec pass-through for the streams that has some immersive video calls which not only has audio and video but also has some Data Stream, BFCP stream, FECC, where BFCC is not supported for SIP trunks which involves MTP, RSVP, TRP or Transcoder and those streams are passed through without modifying the stream.
Thanks,
Kasi
... View more
Created by
Kasiraman S
in Collaboration, Voice and Video Videos
in Collaboration, Voice and Video Videos
06-19-2015
04:38 AM
06-19-2015
04:38 AM
Course Summary: This course provides audience an extensive knowledge on Cisco Unified Contact Center Express and we will cover from basics. We will go through the tasks typically performed during contact center deployment and this course consist of 60% of lab content and 40% theory, which will helps audience in planning, installation, configuration, scripting, and troubleshooting Unified CCX. This course is based on Unified Contact Center Express 9.0 Course Goals: Gain an understanding of Cisco Unified Contact Center Express(UCCX) product suite and all of its associated applications Gain knowledge on Unified CCX components, servers, deployment options, and clients. Describe how to install, integrate and configure Unified CCX with CUCM Learn how to use script editing tool (Unified CCX editor) and how to create IVR scripts Its not yet complete but will add remaining videos shortly. Course URL: www.youtube.com/playlist?list=PL6NMvQ2LNu2vh8ptMvQP49k4jnNDkc9Wf Thanks, Kasi
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Labels:
Created by
Kasiraman S
in IP Telephony and Phones
in IP Telephony and Phones
09-12-2013
07:03 AM
09-12-2013
07:03 AM
Btw can you let me what you configured, may be a simple call flow is enough. Sent from Cisco Technical Support iPhone App
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Created by
Kasiraman S
in IP Telephony and Phones
in IP Telephony and Phones
09-12-2013
07:01 AM
09-12-2013
07:01 AM
Check if the extension numbers are allowed in the restriction table in CUC . Also make sure that the CSS of voice mail ports can reach the user extensions. I mean make sure the CSS has the PT of reception's extension and other extensions. Sent from Cisco Technical Support iPhone App
... View more
09-09-2013
08:14 AM
Are you testing this out in lab or a production setup? What is the source and the detination IP address you used to test to see if the traffic are encrypted? Can you post the configuration of the other end as well. Are they in two different location and is it communicating with the Public IP on the other end. Becasue I see the RTR 1 has a private IP and is it behind any DSL modem?
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11-06-2015
Its not incomplete entry. Static NAT translation in the Running-config
shows like that in NAT transl...
11-05-2015
This is very much possible that the Voice Gateway which are facing
internet get these kind of Toll F...
11-05-2015
Yes, you are correct and Switch will not push 30W. Regarding the VLAN
setting you can refer the belo...
11-05-2015
Seems like Toll Fraud attempts to your GW. I see that call was blocked
by TOLLFRAUD_APP in IOS and y...
11-04-2015
I do not see anything unusual with the config. Execute "show ip address
trusted list" and ensure it ...
11-04-2015
This does not work. Call park feature is intended to park the active
call in a available park slot b...
11-03-2015
Its good to consider enabling SIP OPTIONS PING under SIP Profile if not
already, which helps not to ...
11-03-2015
+5 Deji. Well explained. I believe your answer explains why only codec
pass-through is configured.
11-03-2015
I believe we can consider taking a call which uses RSVP based CAC, which
does require MTP for bandwi...
11-03-2015
Hi Sanjay,Let me try to explain here. Lets say for an example you have
configured this dspfarm profi...
Created by
Kasiraman S
in Collaboration, Voice and Video Videos
06-19-2015
06-19-2015
LITHIUM.OoyalaPlayer.addVideo('https:\/\/player.ooyala.com\/static\/v4\/production\/',
'lia-vid-l1cG...
Created by
Kasiraman S
in Unified Communications Infrastructure
10-27-2014
10-27-2014
Thanks much Mohammed. Will give it a try and will update you!
Regards,Kasi
Created by
Kasiraman S
in Unified Communications Infrastructure
10-27-2014
10-27-2014
Hi Mohammed,Its running SNMP and from the Manage Credentials section, I
was able to test the credent...
Created by
Kasiraman S
in Unified Communications Infrastructure
10-27-2014
10-27-2014
Hi Mohammed,Both the services are running in both server. I tried
restart services/servers without l...
Public Statistics
| Date Registered | 01-23-2010 07:40 AM |
| Date Last Visited | 09-11-2017 02:30 AM |
| Total Messages Posted | 119 |
| Total Helpful Votes Received | 80 |
Helpful Votes Given To
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Helpful Votes From
| User | Helpful Count |
|---|---|
| 5 | |
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