Welcome to the Cisco Proximity Community!

Welcome to the Cisco Proximity Support community. This community will be the main channel for providing feedback on the Cisco Proximity app. Here you can also create discussions and raise concerns and we will try to answer any question you might have.
Before asking a questions we recommend you to take a look in the Frequently Asked Questions document to see if your questions has already been answered. This document also contains a Troubleshooting section, which can be useful if you are seeing any of the most common issues.
Enjoy the application testing and we are waiting for your feedback!

Not sure about fixing it... I managed to find the SQL and PL/SQL FAQ (for some bizarre reason the new forum only likes to show one announcement message at a time at the top - the FAQ is there). Seems a lot of the old Jive tags like the noformat tags etc. seem to be defunct, so the FAQ is going to need some editing, and I don't know where to start in putting all the links right. (Sonya did warn us that links would be broken).
For now, I've at least locked the FAQ thread (SONYA - Are all previously locked threads unlocked by default? that could be a pain)
I can see this may take some time to resolve...
BillyVerreynne wrote:
Some URLs are broken (specifically the PL/SQL forum's FAQ - not sure if Blu fixed that in the meantime) - any means to get old postings' URLs to work consistently?

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Welcome to the Cisco Networking Professionals Unified Computing topic

Welcome to the Cisco Networking Professionals Unified Computing topic. This topic will provide you the opportunity to discuss issues such as building and running server virtualization, mobility management, stateless computing, application provisioning, implementation of FCoE and other solutions based on the Cisco Unified Computing System and Cisco Unified Fabric. We encourage everyone to share their knowledge and start conversations about challenges you might be facing.
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We encourage you to tell your fellow colleagues about the site.
Dan Bruhn
NetPro Community Manager

This is easily done with dial peer statements . The dial peer in your originating router must route the calls to the terminating router first. That would look like :
dial-peer voice xxxxx voip ( the xxxxx is just a tag)
destination-pattern 45... (that would route any 5-digit calls beginning with 45)
session-target ipv4:xxx.xxx.xxx.xxx (ip address of the terminating router)
If digitones are to be dialed after the connection is established, use the statement:
dtmf-relay-h.245-alphanumeric
You could also use a statement to specify the codec to be used:
codec g711ulaw
You would need multiple voip dial peers if the calls were going to different routers based on the dialed digits. If all calls are sent to the same terminating router, use all wild cards in the dest-pattern statement.
At the terminating router configure pots dial peers:
dial-peer voice xxxxx pots
dest-pattern 45...
port x/x (whichever port the call is to be terminated on)
prefix 45 (this re-inserts matched digits which are stripped off by the pots dp)
Repeat for other ports which will receive calls.
Paul

Welcome to the Cisco Networking Professionals Digital Media System topic

Welcome to the Cisco Networking Professionals Digital Media System topic. This forum will provide you the opportunity to discuss topics such as digital signage, Enterprise TV, and desktop video solutions. We encourage everyone to share their knowledge and start conversations about challenges you might be facing.
Remember, just like in the workplace, be courteous to your fellow forum participants. Please refrain from using disparaging or obscene language or posting advertisements.
We encourage you to tell your fellow colleagues about the site.
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NetPro Community Manager

You are not alone in your opinion. The DMD is pretty clunky and useless. We gave up trying to use it. We code our apps with HTML and Javascript. You are correct in thinking that it seems Cisco pushed DMS out the door before it was really ready for market. That was my opinion also. I guess they had their reasons. I've seen the roadmap for the product so I am of the belief that in a few years they will have surpassed most of their competitors and worked out the issues.
5.0 released today. Upgrades will not be available until the end of the month. You can check out the new features on the Cisco website. Its a complete re-design of the GUI. We shall see just how much a step forward it is for the product set.

Welcome to the Cisco CSR (Cloud Service Router) Discussion Forum

Welcome to the Cisco CSR (Cloud Service Router) Discussion Forum!
This forum helps CSR users interact, share knowledge and build communities with one another.
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Best Regards,
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hi,I have a question on sql database high availability. I have tried using database mirroring, where I am using sql standard edition, in this database mirroring of synchronous mode is the only option available, and it is giving problem, like sql time out errors on my applicatons since i had put in the database mirroring, as asynchronous is only available on enterprise version, is there any suggestions on this. thanks ---vijay

Does the Cisco Unified Communications Sizing Tool 8.5(3) work for you ?

The previous version of this sizing tool worked just fine thank you very much, but now when I fire it up, it says it has expired and I should upgrade to the nice shiny new one. So I downloaded the new tool from
http://tools.cisco.com/cucst/faces/landing.jsp
and installed it, but when it runs it displays an Excel error and dies. Don't you love those sorts of circles?
I am on Windows 7 64-bit with Excel 2010 32-bit - a compatible combination, according to the release notes.
Any ideas?
Regards,
Geoff

It's got to be something with you excel security settings.
I am sure you are absolutely correct. I just installed it on a colleague's computer (he works for a different company so his Excel does not have the same settings as mine, which comes from the HP corporate load).
I've played with quite a few different settings in the Trust Center but haven't cracked it yet.
Regards,
Geoff

Ask the Expert: Upgrading Cisco Unified Communications Manager (CUCM) to Version 9.1 (Drive to 9)

Welcome to the Cisco Support Community Ask the Expert conversation. Learn from experts Vijay Rao and Amit Singh about simplified upgrade process and focused support from Cisco to migrate to version 9.1.
This is a continuation of the live Webcast
Drive to 9 is a comprehensive and holistic program designed to help you upgrade the current Cisco® Unified Communications Manager installed base to version 9.1 or higher. This upgrade will enable customers to have next-generation collaboration experiences.
During the live event, Cisco subject matter experts Vijay Rao and Amit Singh focussed on the simplified upgrade process and focused support from Cisco to migrate to version 9.1. They also talked about the changes made to the licensing model of User Connect Licensing and Cisco Unified Workspace Licensing.
Vijay Rao is a Network Consulting Engineer and is currently a unified communications (UC) consultant for Bank of America. He has been providing consulting assistance to the bank for the past 6 years. He helps design complex UC networks for large enterprise customers. He was previously part of Cisco IT in the Asia Pacific, Japan, and China (APJC) region and was instrumental in designing and implementing the Bangalore campus. He has been working with Cisco for 9 years and has 12 years of UC experience. He has a Cisco CCVP® certification.
Amit Singh is a customer support engineer at the Cisco Technical Assistance Center in Bangalore, India. He has 7 years of experience in his areas of expertise: wireless, Cisco Unified Communications Manager, multiservices, Cisco Unity®, and Cisco Unified Contact Center Express. He has been involved in various escalation requests from India, Singapore, and Australia and is currently working as a technical lead for the Voice team in Bangalore, India. He is a computer science graduate.
Remember to use the rating system to let Vijay and Amit know if you have received an adequate response.
Vijay and Amit might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation on the Collaboration, Voice and Video sub-community forum shortly after the event. This event lasts through July 19, 2013. Visit this forum often to view responses to your questions and the questions of other community members.
Webcast related links:
Webcast Video
FAQ from the live webcast
Slides from the live webcast

Hello Robert,
Apologies for a delayed response, some days get very hectic.
In CallManager, we only define the SRST reference, and CUCM version and SRST version are independent of each other.
The only thing, which is related and will change with CUCM upgrade is Phone F/w version.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/compat/ccmcompmatr1.pdf
You may just want to check your, phone f/w compatibility with the SRST version running on your ISR G1 Gateways:
http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_device_support_tables_list.html
For Example: SRST version 7.1
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps2169/data_sheet_c78-520521.html
You may want to do some lab testing with CUCM 9.1 and an SRST supported f/w on your phones.
If you decide to run the old Phone/F/w to support the SRST version, you may not be able to take advantage of new features.
Also, you can try and upgrade your phones(Wih CUCM 9.1) and test them with your SRST version.
It should work fine, but from a troubleshooting perspective, TAC may request you to come into a Cisco Supported combination.
Please, let me know if this clarifies your doubt or we can have a quick phone call.
Regards
Amit Singh

Ask the Expert : Call Recording with Cisco Unified Communication Manager (UCM)

Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to learn and ask questions about Cisco Unified CM call recording solution that provides the ability to record customer conversations for compliance purpose. This topic will cover an overview, configuration and troubleshooting of the call recording feature.
Monday, January 19th, 2015 to Friday, January 30th, 2015
Harmit Singh is a technical leader with the High Touch Technical Services (HTTS) and Technical Assistance Center (TAC) Unified Communications teams based in Bangalore. He has broad experience in Cisco Unified Communications infrastructure solutions. He has 10 years of experience working with large enterprise and service provider networks. He also holds CCIE certifications (#20012) in Voice and Collaboration as well as Red Hat and VMware certifications.
Mohammed Noorulla Khan is a customer support engineer in High-Touch Technical Services (HTTS) Unified Communications teams based in Bangalore. His areas of expertise include Cisco Unified Communications Manager, Gateways, and Jabber. He has over 6 years of industry experience working with large enterprise and service provider networks. He also holds CCIE certifications (#35741) in Voice and VMware certifications.
** Remember to use the rating system to let Mohammed and Harmit know if you've received an adequate response. **
Because of the volume expected during this event, the experts might not be able to answer every question. Remember that you can continue the conversation in the Collaboration, Voice and Video community, subcommunity, IP Telephony, shortly after the event. This event lasts through January 30th 2015. Visit this forum often to view responses to your questions and those of other Cisco Support Community members.

Hi Maheshwar,
Thank you for your query. Please find my response below:
1> Do we support recording with HCS environment and which 3rd party vendors are validated with HCS based call control 10.1.1?
Answer: Whether you use a standalone UCM cluster, UCCE or HCS, call recording would be supported across the board in the same manner.
Please refer to the following link:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/hcs-cc/10_0_1/Install_and_Config/CHCS_BK_ICC270D0_00_installing-and-configuring-cisco-hcs/CHCS_BK_ICC270D0_00_installing-and-configuring-cisco-hcs_chapter_011.html#CHCS_RF_T1105284_00
Option
Notes
Recording
All Recording applications that are supported by Unified CCE are supported on HCS for CC. For details, see Recording section in Agent and Supervisor Capabilities.
With respect to which 3rd party vendors have been validated, marketplace.cisco.com is a good place to crosscheck this info. You will find a Cisco Compatible Logo against the partners listed there. The logo is used to signify that the PARTNER product has undergone technical interoperability testing with the Cisco product specified. The interoperability testing is conducted by a third party laboratory based on testing criteria set forth by Cisco. PARTNER is solely responsible for the support and warranty of its product. Placement of the PARTNER product or information pertaining thereto, on the Cisco Marketplace website does not constitute an offer to sell the PARTNER product in any way. For further information on the PARTNER products, please visit the PARTNER company website.
Please refer to the following link and use the search field under Collaboration Technology:
https://marketplace.cisco.com/catalog/search?utf8=%E2%9C%93&search[q]=&search[technology_category_ids]=23%2C24%2C197%2C1940%2C1941%2C1921%2C1576%2C1897%2C1983%2C2418%2C26%2C198%2C1904&search[order]=tier&per_page=20&_=1421663854257&ts=1421663855441
2> Which end points are supported for recording via HCS call control?
Answer: The following link should help clarify this:
http://solutionpartner.cisco.com/web/sip/wiki/-/wiki/Main/Unified+CM+Silent+Monitoring+Recording+Supported+Device+Matrix
Please let us know if you have any follow up questions. Hope this is helpful.
Regards,
Harmit Singh.

Ask the Expert: Plan, Design, and Implement Mobile Remote Access, the Cisco Collaboration Edge Architecture

Welcome to the Cisco® Support Community Ask the Expert conversation. This is an opportunity to learn and ask questions about planning, designing, and implementing mobile remote access (Cisco Collaboration Edge Architecture) with Cisco subject matter experts Aashish Jolly and Abhijit Anand.
Cisco Collaboration Edge Architecture is an architecture that provides VPN-less access of Cisco Unified Communications resources to Cisco Jabber® users. This discussion is dedicated to addressing questions about design best practices while implementing mobile remote access.
For more information, refer to the Unified Communications Mobile and Remote Access via Cisco VCS deployment guide.
Aashish Jolly is a network consulting engineer who is currently serving as the Cisco Unified Communications consultant for the ExxonMobil Global account. Earlier at Cisco, he was part of the Cisco Technical Assistance Center (TAC), where he helped Cisco partners with installation, configuring, and troubleshooting Cisco Unified Communications products such as Cisco Unified Communications Manager and Manager Express, Cisco Unity® solutions, Cisco Unified Border Element, voice gateways and gatekeepers, and more. He has been associated with Cisco Unified Communications for more than seven years. He holds a bachelor of technology degree as well as Cisco CCIE® Voice (#18500), CCNP® Voice, and CCNA® certifications and VMware VCP5 and Red Hat RHCE certifications.
Abhijit Singh Anand is a network consulting engineer with the Cisco Advanced Services field delivery team in New Delhi. His current role involves designing, implementing, and optimizing large-scale collaboration solutions for enterprise and defense customers. He has also been an engineer at the Cisco TAC. Having worked on multiple technologies including wireless and LAN switching, he has been associated with Cisco Unified Communications technologies since 2006. He holds a master’s degree in computer applications and multiple certifications, including CCIE Voice (#19590), RHCE, and CWSP and CWNP.
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Because of the volume expected during this event, our experts might not be able to answer every question. Remember that you can continue the conversation on the Cisco Support Community Collaboration, Voice and Video page, in the Jabber Clients subcommunity, shortly after the event. This event lasts through June 20, 2014. Visit this forum often to view responses to your questions and the questions of other Cisco Support Community members.

Hi Marcelo,
Yes, there are some requirements for certificates in Expressway.
Expressway Core (Exp-C)
- Can be signed by either External or Internal CA
- Better to use a cluster name even if you start with 1 peer in Exp-C cluster. In the future, if more peers are added, changes would be minimal.
- Better to use FQDN of cluster as CN of certificate, this way the traversal zone configuration on Expressway-E won't require any change even if new peers are added to Exp-C cluster.
- If CUCM is mixed mode, include security profile names (in FQDN format) as Subject Alternate Names
- The Chat Node Aliases that are configured on the IM and Presence servers. They will be required only for Unified Communications XMPP federation deployments that intend to use both TLS and group chat. (Note that Unified Communications XMPP federation will be supported in a future Expressway release). The Expressway-C automatically includes the chat node aliases in the CSR, providing it has discovered a set of IM&P servers.
- For TLS b/w CUCM, IM-P & Exp-C
+ If using self-signed certificates on CUCM, IM/P. Load Cisco Tomcat, cup, cup-xmpp certificates from IM-P on Exp-C. Load callmanager, Cisco Tomcat certificates from CUCM on Exp-C.
+ If using Internal CA signed certificates on CUCM, IM/P. Load Root CA certificates on Exp-C.
+ Load CA certificate under tomcat-trust, cup-trust, cup-xmpp-trust on IM-P.
+ Load CA certificate under tomcat-trust, callmanager-trust on CUCM.
Expressway Edge (Exp-E)
- Signed by External CA
- Configured Unified Communications domain as Subject Alternate Name
- If using a cluster, select FQDN of this peer as CN and FQDN of Cluster + this peer as Subject Alternate Name.
- If XMPP federation is being deployed, enter the same Chat Node Aliases as entered in Exp-C.
For more details, please refer to the Certificate Creation Guide for Cisco Expressway x8.1.1
http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X8-1/Cisco-Expressway-Certificate-Creation-and-Use-Deployment-Guide-X8-1.pdf
- Aashish

Ask the Expert: Configuration and Troubleshooting the Cisco Application Control Engine (ACE) load balancer

With Ajay Kumar and Telmo Pereira
Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to learn and ask questions about configuration and troubleshooting the Cisco Application Control Engine (ACE) load balancer with Cisco expert Ajay Kumar and Telmo Pereira. The Cisco ACE Application Control Engine Module for Cisco Catalyst 6500 Series Switches and Cisco 7600 Series Routers is a next-generation load-balancing and application-delivery solution. A member of the Cisco family of Data Center 3.0 solutions, the module: Helps ensure business continuity by increasing application availability Improves business productivity by accelerating application and server performance Reduces data center power, space, and cooling needs through a virtualized architecture Helps lower operational costs associated with application provisioning and scaling
Ajay Kumar is a customer support engineer in the Cisco Technical Assistance Center in Brussels, covering content delivery network technologies including Cisco Application Control Engine, Cisco Wide Area Application Services, Cisco Content Switching Module, Cisco Content Services Switches, and others. He has been with Cisco for more than four years, working with major customers to help resolve their issues related to content products. He holds DCASI and VCP certifications.
Telmo Pereira is a customer support engineer in the Cisco Technical Assistance Center in Brussels, where he covers all Cisco content delivery network technologies including Cisco Application Control Engine (ACE), Cisco Wide Area Application Services (WAAS), and Digital Media Suite. He has worked with multiple customers around the globe, helping them solve interesting and often highly complex issues. Pereira has worked in the networking field for more than 7 years. He holds a computer science degree as well as multiple certifications including CCNP, DCASI, DCUCI, and VCP
Remember to use the rating system to let Ajay know if you have received an adequate response.
Ajay and Telmo might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation on the Data Center sub-community discussion forum Application Networking shortly after the event.
This event lasts through July 26, 2013. Visit this forum often to view responses to your questions and the questions of other community members.

Hello Krzysztof,
Another set of good/interesting questions posted. Thanks!
I will try to clarify your doubts.
In the output below both resources (proxy-connections and ssl-connections rate) are configured with a min percentage of resources (column Min), while 'Max' is set to equal to the min.
ACE/Context# show resource usage
                                                     Allocation
        Resource         Current       Peak        Min        Max       Denied
-- outputs omitted for brevity --
proxy-connections             0      16358      16358      16358      17872
ssl-connections rate          0        626        626        626      23204
Most columns are self explanatory, 'Current' is current usage, 'Peak' is the maximum value reached, and the most important counter to monitor 'Denied' represents the amount of packets denied/dropped due to exceeding the configured limits.
On the resources themselves, Proxy-connections is simply the amount of proxied connections, in other words all connections handled at layer 7 (SSL connections are proxied, as are any connections with layer 7 load balance policies, or inspection).
So in this particular case for the proxy-connections we see that Peak is equal to the Max allocated, and as we have denies we can conclude that you have surpassed the limits for this resource. We see there were 17872 connections dropped due to that.
ssl-connections rate should be read in the same manner, however all values for this resource are in bytes/s, except for Denied counter, that is simply the amount of packets that were dropped due to exceeding this resource.
For your particular tests you have allocated a min percentage and set max equal to min, this way you make sure that this context will not use any other additional resources.
If you had set the max to unlimited during resource allocation, ACE would be allowed to use additional resources on top of those guaranteed, if those resources were available.
This might sound a great idea, but resource planning on ACE should be done carefully to avoid any sort of oversubscription, specially if you have business critical contexts.
We have a good reference for ACE resource planning that contains also description of all resources (this will help to understand the output better):
http://www.cisco.com/en/US/docs/interfaces_modules/services_modules/ace/v3.00_A2/configuration/virtualization/guide/config.html#wp1008224
1) When a resource is utilized to its maximum limit, the ACE denies additional requests made by any context for that resource. In other words, the action is to Drop. ACE should in theory silently drop (No RST is sent back to the client). So unless we changed something on the code, this is what you should see.
To give more context, seeing resets with SSL connections is not necessarily synonym of drops. As it is usual to see them during normal transactions.
For instance Microsoft servers are usually ungracefully terminating SSL connections with RESET. Also when there is renegotiation during an SSL transaction you may see RESETS, but this will pass unnoticed for end users.
2) ACE will simply drop/ignore new connections when we reach the maximum amount of proxied connections for that context. Exisiting connections will continue there.
As ACE doesn't respond back, client would simply retransmit, and if he is lucky maybe in the next attempt he will be able to establish the connection.
To overcome the denies, you will definitely have to increase the resource allocation. This of course, assuming you are not reaching any physical limit of the box.
As mentioned setting max as unlimited might work for you, assuming there are a lot of unused resources on the box.
3) If a new connection comes in with a sticky value, that matches the sticky entry of a real server, which is already in MAXCONNS state, then both the ACE module/appliance should reject the connection and that sticky entry would be removed.
The client would at that point reestablish a new connection and ACE would associate a new sticky entry with the flow for a new RSERVER after the loadbalancing decision.
I hope this makes things clearer! Uff...
Regards,
Telmo

Welcome to the IPv6 Community

Welcome to the CSC IPv6 Community. We encourage everyone to share their knowledge and start conversations on issues such as adding IPv6 capability to your networks and discuss challenges and successes around network services migration. All topics are welcome, including routing address management, translation, tunneling, firewall, security, DNS, MTU, applications and protocol stack issues.
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Cheers,
Dan Bruhn
CSC Community Manager

hiLisa_K8 wrote:
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Welcome to the Service Providers General discussion

Welcome to the Cisco Networking Professionals Connection Service Providers Forum. This topic will provide you the opportunity to discuss issues surrounding General Service Provider solutions. We encourage everyone to share their knowledge and start conversations on issues such as storage Cable, DSL, WiMAX, and any other topic concerning SP networking.
Remember, just like in the workplace, be courteous to your fellow forum participants. Please refrain from using disparaging or obscene language or posting advertisements.
We encourage you to tell your fellow networking professionals about the site. If you would like us to send them a personal invitation simply send their names and e-mail addresses along with your name to us at [email protected]
Dan Bruhn
NetPro Community Manager

Welcome to the Cisco Networking Professionals Connection Service Providers Forum. This topic will provide you the opportunity to discuss issues surrounding General Service Provider solutions. We encourage everyone to share their knowledge and start conversations on issues such as storage Cable, DSL, WiMAX, and any other topic concerning SP networking.
Remember, just like in the workplace, be courteous to your fellow forum participants. Please refrain from using disparaging or obscene language or posting advertisements.
We encourage you to tell your fellow networking professionals about the site. If you would like us to send them a personal invitation simply send their names and e-mail addresses along with your name to us at [email protected].
Dan Bruhn
NetPro Community Manager

Welcome to the MARS Discussion

Welcome to the Cisco Networking Professionals Cisco Security MARS Forum. This conversation will provide you the opportunity to discuss the product, solutions and issues surrounding Cisco Security MARS deployments, maintenance and integration. We encourage everyone to share their knowledge and start conversations about topics involving the Cisco Security MARS.
Remember, just like in the workplace, be courteous to your fellow forum participants. Please refrain from using disparaging or obscene language or posting advertisements.
We encourage you to tell your fellow networking professionals about the site.
Dan Bruhn
NetPro Community Manager

Great to have a special forum for MARS !
Are there any best practices documents that would get me up to speed with customizing MARS to show more focused alerts? For instance until I've configured IPS events into MARS, syslog events from ASA/FWSM were not enough for MARS to conclude that a scan is being performed.
I tried to make a lot of "noise" with hacking tools such as Nessus across the firewalls w/o any special "Critical" level alerts.
Thanks,
Yigal

Cisco Unified Communications Manager 10.0 - New Features Google Hangout Q&A Session

Engineer, Customer Support, Cisco
Welcome the Cisco Support Community Google Hangout Q&A Session about Cisco Unified Communications Manager 10.0. If you have any questions regarding the product or the content of the Google Hangout, please post them here.
Cisco UCM Product: http://cs.co/cucm10 http://cs.co/cucm10
Google Hangout Page : https://plus.google.com/events/cmt40r73djuhnauh5blg31k13ic

1.Licensing on v10 is not different from version9. But yes it has been renamed to Cisco Prime License Manager.
If you compare licenses on v10 to CUCM version 8.X or earlier, then in that case you will see a BIG change. If you need more info about this change please do let me know.
2.IM and Presence Service Release 10.0(1) is supported only with Cisco Unified Communications Manager Release 10.0(1).
Note IM and Presence Service Release 9.1(x) is supported only with Cisco Unified Communications Manager Release 9.1(x).
Source: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/compat/10_x/CUCM_BK_CD1DB914_00_compat_matrix/CUCM_BK_CD1DB914_00_compat_matrix_chapter_011.html#CUP0_RF_SECB5848_00
3. End user features include SAML SSO, vido on hold (contact center environment), etc. This session on routing enhancement covered urgent priority which would also translate to end user experience as they might not face the delay in call completion.

Welcome to the Voice over IP Discussion

Welcome to the Cisco Networking Professionals Connection Service Provider Forum. This conversation will provide you the opportunity to discuss issues surrounding Voice over IP. We encourage everyone to share their knowledge and start conversations on issues such as VoIP Transport, VoIP Access, Managed Voice Services, H.323, SIP, VXML, MGCP, SS7 and any other topic concerning Voice over IP.
Remember, just like in the workplace, be courteous to your fellow forum participants. Please refrain from using disparaging or obscene language or posting advertisements.
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This is easily done with dial peer statements . The dial peer in your originating router must route the calls to the terminating router first. That would look like :
dial-peer voice xxxxx voip ( the xxxxx is just a tag)
destination-pattern 45... (that would route any 5-digit calls beginning with 45)
session-target ipv4:xxx.xxx.xxx.xxx (ip address of the terminating router)
If digitones are to be dialed after the connection is established, use the statement:
dtmf-relay-h.245-alphanumeric
You could also use a statement to specify the codec to be used:
codec g711ulaw
You would need multiple voip dial peers if the calls were going to different routers based on the dialed digits. If all calls are sent to the same terminating router, use all wild cards in the dest-pattern statement.
At the terminating router configure pots dial peers:
dial-peer voice xxxxx pots
dest-pattern 45...
port x/x (whichever port the call is to be terminated on)
prefix 45 (this re-inserts matched digits which are stripped off by the pots dp)
Repeat for other ports which will receive calls.
Paul

Welcome to the MPLS over ATM Discussion

Welcome to the Cisco Networking Professionals Connection Service Provider Forum. This conversation will provide you the opportunity to discuss issues surrounding MPLS over ATM. We encourage everyone to share their knowledge and start conversations on issues such as Frame-based MPLS networks, multiservice networks, VPN scalability, multiple service classes, multicast, VoIP and any other topic concerning MPLS over ATM.
Remember, just like in the workplace, be courteous to your fellow forum participants. Please refrain from using disparaging or obscene language or posting advertisements.
We encourage you to tell your fellow networking professionals about the site. If you would like us to send them a personal invitation simply send their names and e-mail addresses along with your name to us at [email protected]

This is easily done with dial peer statements . The dial peer in your originating router must route the calls to the terminating router first. That would look like :
dial-peer voice xxxxx voip ( the xxxxx is just a tag)
destination-pattern 45... (that would route any 5-digit calls beginning with 45)
session-target ipv4:xxx.xxx.xxx.xxx (ip address of the terminating router)
If digitones are to be dialed after the connection is established, use the statement:
dtmf-relay-h.245-alphanumeric
You could also use a statement to specify the codec to be used:
codec g711ulaw
You would need multiple voip dial peers if the calls were going to different routers based on the dialed digits. If all calls are sent to the same terminating router, use all wild cards in the dest-pattern statement.
At the terminating router configure pots dial peers:
dial-peer voice xxxxx pots
dest-pattern 45...
port x/x (whichever port the call is to be terminated on)
prefix 45 (this re-inserts matched digits which are stripped off by the pots dp)
Repeat for other ports which will receive calls.
Paul

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