Showing posts with label Calling. Show all posts
Showing posts with label Calling. Show all posts

Wednesday, October 27, 2021

Integrating WebEx Calling and Communications Manager Express 2/2

This is the second post in the two post series. It will go into more detail on the configuration of the solutions and workarounds put in place.

 


  • The Communications Manager Express (WebEx local gateway / CUBE) requires a security license. There's not much else to add. Make sure it has one so the CUBE will accept the TLS commands.

 

  • The CME phone directory numbers will by default register to the WebEx Control Hub. According to TAC this can confuse the Webex Control Hub and cause issues. The symptom of the phones trying to register with Webex is evident with a “show sip-ua register status”. It will show the directory numbers of all the CME phones trying to register to Webex. TAC explained that it should only show the SIP trunk trying to register to Webex. (It’s also normal to see registration for a SIP IP Telephony Service Provider too if it's configured for authentication) TAC offered a configuration sample to prevent the SIP devices from registering with Webex and I looked up how to prevent the SCCP devices and POTS device ports. Please see the following three configuration samples.

 

    • SIP phones on the CME will need their directory numbers configured as “no-reg”

    voice register dn <>

    no-reg

 

    • SCCP phones on CME will need their directory numbers configured for “no-reg”
                   ephone-dn <>
        number <> no-reg primary


    • Analog phones connected to CME will need to have their pots dial-peer configured not to “sip-register” it's destination-pattern

        dial-peer voice <> pots

         destination-pattern <>

         port <>/<>/<>

         no sip-register


            Note: In my case, I had to reboot the CUBE before the configuration changes took full effect.


After configuring the system to not pass the phone registration along to Webex I still have two registration attempts listed. The first one is to the IP Telephony Service Provider for the CME phones and the second is to Webex for the integration. The following CLI sample output is similar to my registration CLI output now.


ExampleCME#show sip-ua register status


Tenant:  10

--------------------- Registrar-Index  1 ---------------------

Line                             peer      expires(sec) reg survival  P-Associ-URI

================================ ========= ============ === ========  ============

5555551212                       -1         239          yes normal


Tenant:  200

--------------------- Registrar-Index  1 ---------------------

Line                             peer      expires(sec) reg survival  P-Associ-URI

================================ ========= ============ === ========  ============

ExampleCME.Trunk1234_LGU         -1         39           yes normal

ExampleCME#


  • The CME location in the WebEx Control Hub needs to have the main phone number associated with it. This is kind of an interesting step because the CME is not hosting Webex Calling phones but rather it's hosting its own CME phones. However, Webex won’t pass calls without the main phone number configured. I chose an unused DID the customer owns, added it to the control hub, and associated it to the location the trunk is tied to.

 

  • The CME dial-peers directed at the WebEx cloud didn't present the SIP signaling in a way that the WebEx cloud expected to see it. According to Cisco TAC their configuration guide was designed for the CUBE to front end a Communications Manager or other PBX device. Cisco TAC suggested hair-pinning the dial peers to obfuscate CME from WebEx Calling. TAC also helped me do a little bit of dial-plan manipulation prefixing digits so that the CUBE didn't drop the calls as a "loop". Technically TAC said CME isn't explicitly listed as being supported however, this workaround presented the SIP signaling to the Webex cloud in the Webex expects to see it. Please see the following information on how the dial-peers and translations were configured:

 

Inbound calls from WxC to CME

The "Configure Local Gateway on IOS-XE for Webex Calling" guide from Cisco includes dial-peers without any dial-plan numbers tied to them. In their example, they show dial-peer 200201 matching inbound calls from Webex Calling and then sending it along to dial-peer group 300.

 

Example of a portion of the reference configuration in the "Configure Local Gateway on IOS-XE for Webex Calling" guide from Cisco:

dial-peer voice 200201 voip

 description Inbound/Outbound Webex Calling

 destination dpg 300

 

I took Cisco’s example config and modified the dial-peer group so that it passed the calls to my dial-peer 444001 below. This dial-peer hairpins the calls back onto the CME and prefixes a 444 so the CUBE doesn't drop the call as a duplicate.

dial-peer voice 4440001 voip

 description Inbound calls from WxC to CME - Prefix 444 and loop via CUBE

 translation-profile outgoing add444

 destination-pattern 8040..$

 session protocol sipv2

 session target ipv4: <CME IP Address>

 voice-class sip early-offer forced

 voice-class sip bind control source-interface Loopback0

 voice-class sip bind media source-interface Loopback0

 dtmf-relay rtp-nte sip-kpml sip-notify

 codec g711ulaw

 no vad

 

This inbound dial-peer matches the call with the 444 prefix and strips it off. Then the CME phone will match the outbound leg.

dial-peer voice 4440002 voip

 description Inbound calls from WxC to CME - Strip 777 Webex Calling Strip 777 - WxC Side

 translation-profile incoming strip444

 session protocol sipv2

 session target sip-server

 incoming called-number 444T

 voice-class sip early-offer forced

 voice-class sip bind control source-interface Loopback0

 voice-class sip bind media source-interface Loopback0

 dtmf-relay rtp-nte sip-kpml sip-notify

 codec g711ulaw

 no vad

 

Outbound Calls from CME to WxC

This outbound dial-peer matches the directory number of phones on WxC, then prefixes a 777, and then loops the call back through the CUBE.

 

     dial-peer voice 7770001 voip

      description Outbound calls to WxC - Prefix 777 and loop via CUBE

      translation-profile outgoing add777

      destination-pattern 8070..$

      session protocol sipv2

      session target ipv4:<CME IP Address>

      voice-class sip early-offer forced

      voice-class sip bind control source-interface Loopback0

      voice-class sip bind media source-interface Loopback0

      dtmf-relay rtp-nte sip-kpml sip-notify

      codec g711ulaw

      no vad

 

This dial-peer matches the inbound call leg for calls routed to the 777 prefix.

dial-peer voice 7770002 voip

 session protocol sipv2

 session target sip-server

 incoming called-number 777T

 voice-class sip early-offer forced

 voice-class sip bind control source-interface Loopback0

 voice-class sip bind media source-interface Loopback0

 dtmf-relay rtp-nte sip-kpml sip-notify

 codec g711ulaw

 no vad

 

This dial-peer matches the outbound call leg for calls with the 777 prefix. Strips off the prefix and the routes the call out to Webex Calling

 

dial-peer voice 7770003 voip

 description Outbound calls to Webex Calling Strip 777 - WxC Side

 translation-profile outgoing strip777

 max-conn 150

 destination-pattern 7778070..$

 session protocol sipv2

 session target sip-server

 destination dpg 300

 voice-class codec 99 

 voice-class stun-usage 200

 no voice-class sip localhost

 voice-class sip tenant 2000

 dtmf-relay rtp-nte

 srtp

 no vad

 

The information below shows the translations config: 

voice translation-rule 4441

 rule 1 /\(......\)/ /444\1/

 

voice translation-rule 4442

 rule 1 /^444\(......$\)/ /\1/

 

voice translation-rule 7771

 rule 1 /\(......\)/ /777\1/


voice translation-rule 7772

 rule 1 /^777\(......$\)/ /\1/

 

voice translation-profile add444

 translate called 4441

 

voice translation-profile strip444

 translate called 4442

 

voice translation-profile add777

 translate called 7771

 

voice translation-profile strip777

 translate called 7772


Thursday, July 15, 2021

Integrating Webex Calling and Communications Manager Express 1/2

 Cisco now supports WebEx Calling locations that use Cloud Connected PSTN to also integrate with a local gateway (CUBE) for on-net dialing to a PBX. The new trunk, route group, and dial plan configuration options in the WebEx Control Hub facilitate this. Cisco has guides published that reference the configuration to support Communications Manager as that PBX specifically. These two posts will take it in a slightly different direction and explore integrating with Communications Manager Express. While not explicitly called out as supported by Cisco I've had success with the integration with a little assistance from Cisco TAC. This integration may help folks migrating off of existing CME deployments and onto WebEx Calling.


This first post will call out some caveats I ran into at a high level. The second post of two will go into more detail about how I addressed the caveats.


Gotchas

The Communications Manager Express (WebEx local gateway / CUBE) requires a security license. While this makes sense now it wasn't explicitly called out in the documentation and I missed this requirement initially. The documentation does say, "The trunk between the local gateway and the Webex cloud is always secured using SIP TLS transport and SRTP for media between the local gateway and the Webex Calling Access SBC". Read, get a security license on that CUBE. (along with the CUBE licensing that is stated as required in the documentation)


The CME phone directory numbers will by default register to the WebEx Control Hub. This can cause confusion/issues with the trunk authentication between the CUBE and the WebEx Control Hub. The first time I integrated a CME it didn't present as an issue but the 2nd time I integrated one the trunk wouldn't authenticate. Cisco TAC requested that I removed the registration from the CME phone directory numbers to the WebEx Control Hub.


The WebEx Calling location in the WebEx Control Hub needs to have a main phone number associated with it. "You will not be able to make or receive calls until this number is added" Even though this location will not host phones in the WebEx Control Hub the WebEx Calling location must have a main number defined or the location will not process calls. 


The CME dial-peers directed at the WebEx cloud didn't present the SIP signaling in a way that the WebEx cloud expected to see it. Cisco TAC suggested hair-pinning the dial peers to obfuscate CME from WebEx Calling. We also needed to do a little bit of dial-plan manipulation prefixing digits so that the CUBE didn't drop the calls as a "loop". Technically TAC said CME isn't specially called out as being supported however, this workaround allowed the SIP signaling to present to the WebEx cloud in the way they typically expect to see it.


Monday, July 29, 2019

Marrying Cloud Video with On-Premises Calling 2/2

This blog is working under the assumption that the following on-premises pieces have already been built out and are working independently of the WebEx cloud piece:

  • Cisco Unified Communications Manager solution on-premises
  • On-premises PSTN connectivity via a Cisco Voice Gateway or CUBE

This blog also assumes that the WebEx piece has already been provisioned and that Video endpoints are already registered to the cloud.
Part 2 of this blog adds some details around the pieces that added to what’s listed above to pull this together. Here’s the high-level list of technologies mentioned again.

  • WebEx Control Hub
  • Cisco CUCM
  • Cisco Expressways
  • Firewalls
  • NAT
  • DNS
  • Certificates
  • Cisco Directory Connector software

Please read all of the Cisco guides thoroughly before attempting design or installation. With that said here’s a little bit of information about each technology with my tech notes.

The curve balls I ran into in the process are in orange text. 

WebEx Control Hub
The WebEx Control Hub is the partner/customer configuration touchpoint for the WebEx cloud. Below is a high-level list of most of the things configuring in the WebEx Control Hub for Hybrid Calling.

  • Created a custom SIP subdomain (for example joshualearn.com)


  • “Verify” the subdomain via a DNS text record with a special hash value provided in the WebEx Control Hub
  • Optionally “claim” the subdomain with a TAC case. (If the subdomain is claimed then any new WebEx Teams account with that email domain has to be created in the official Company organization)
  • Call Connect
    • It gets enabled pointed to the SRV record for the MTLS connection to the Expressway-E For example, _sips._tcp.sipmtls.joshualearn.com (It’s also possible to use an IP and port in the case of a quick POC)
    • Upload the expressway certs to WebEx (if they’re self-signed)
  • Active Hybrid Call Service
    • Add a default SIP destination for Hybrid Call Connect Service
    • Enable Hybrid Calling for the user in the WebEx cloud. (The directory URI of the Spark Remote Destination should match the user's mail id in CUCM which should also match the user-id in the WebEx control hub

Cisco CUCM
The Cisco CUCM should already be in place but there a few things that to be validated and or remediated to start and there a few things that need to be configured specifically to the integration build-out.

General Prerequisites

  • The CUCM database replication needs to be built out using FQDNs. The Expressway will connect to the CUCMs using that same value and need a FQDN that can be resolved via DNS. This caught me off guard the first time and required additional planning to update the system to FQDN entries and to re-establish the database replication between CUCMs.


  • Cluster Fully Qualified Domain Name in the Enterprise Parameters needs to be configured to something unique. For example cluster1.joshualearn.com

New build out for Hybrid Calling

  • The CUCM needs a service account with AXL permissions (which is pretty standard for most applications)


  • Cisco CUCM needs a SIP trunk built out to the Expressway C server
  • Cisco CUCM needs SIP route patterns to send calls to WebEx.
  • *.calls.webex.com


  • *.rooms.webex.com


  • Configure the Spark remote device
  • Configure a Spark Remote Destination for the user

Cisco Expressway
The Cisco Expressways is a bit of Swiss Army Knife for Cisco when it comes to additional software required for Hybrid Calling. Most of the specific connector software runs on the Expressway-C Connector host. There is also a transit pair required to create the secure transport between CUCM on the WebEx cloud.
Cisco Expressway-C Services Host

  • Cisco recommends that the Expressway-C Services Host is updated to the latest version of code. This will allow for interoperability with the WebEx Control Hub. This is also the reason why it’s helpful to have the Services host on its own dedicated Expressway. (Some customers have very small deployments and are tempted to use the same Expressway-C as the Connector host and the transit host. However, these can cause additional challenges with keeping up the latest Expressway code on the C and E. If the Expressway-C Connector host is separate than the transit pair for the Expressways can be updated on a different interval.


  • The Management Connector is installed to establish connectivity to the WebEx Control Hub from the Expressway


  • The Call Connector is installed to create the data hook between the CUCM and the WebEx control hub


  • Linking the Call Connector to CUCM pulls the IP address or the FQDN of the CUCM. If the system was defined using a hostname in CUCM the connection will fail. This is an issue I ran once. For whatever reason, the folks that deployed the initial CUCM decided to use hostnames. The Expressway-C Connector host could not resolve the hostname and therefore could not connect the CUCM.

Cisco Expressway Traversal Pair
The traversal pair is basically a way for the SIP traffic to traverse from the Cisco CUCM on-premises environment to the WebEx Control Hub and cloud registered video endpoints. The expressway traversal hosts can be dedicated or shared with existing MRA and or B2B expressways. Refer to the Expressway guides for details on how to spin up and license them.
Cisco Expressway-C Traversal Host

  • The Expressway-C host sits on the internal voice server network.

Cisco Expressway-E Traversal Host (with 2 NICs)

  • The Expressway-E ideally sits in a firewall environment with a transit DMZ and a service DMZ. However, I have also seen it deployed with one NIC in a service DMZ and the other NIC on the inside network.
  • Create an MTLS zone between the Expressway-E and the WebEx cloud.
  • Follow the guide to configure zones and search rules on the Expressways.
  • One caveat I ran into once was with using the automatic WebEx DNS zone. It ended up causing an error message to pop up. I don’t remember any more specifically what the message was. It may have been that is was looking for additional SIP ports to be available instead of just the MTLS SIP port 5062. In any case, I manually built this rule and I didn’t receive any errors.
  • Another caveat I ran into was the tool to validate the connection to the WebEx control hub was bugged out and didn't support MTLS. That warning can be ignored. (At least until Cisco updates that portion of the code). Bug ID: 

Firewall

  • Two firewall DMZs are recommended by Cisco however, some customers have opted to implement one and put the “transport DMZ NIC” on the inside network.


  • Firewall policies
  • Open up port 5062 on the firewall for traffic towards the Expressway-E (for the WebEx Control Hub)

NAT

  • NAT the “Service DMZ” interface of the Expressway-E to a public IP address

DNS

  • A record – create an A record for the Expressways. For example expe.us1.joshualearn.com.


  • SRV record – create an MTLS SIP service record. For example _sips._tcp.sipmtls.joshualearn.com. 300 IN SRV 5 10 5062 expe.us1.joshualearn.com.

Certificates

  • Internal certificates are allowable if the Expressway-E is only hosting the Hybrid Calling, however, the WebEx Control Hub needs to trust the issuing CA chain. If the Expressway-E is also hosting MRA and or B2B then a publicly signed certificate is required on the Expressway-E


Have you implemented this before, have thoughts about trying it out, or do you just like to leave comments? If so, please leave one below.

Wednesday, May 29, 2019

Marrying Cloud Video with On-Premises Calling 1/2

Cisco's current line of Video endpoints (Room Kit, Room Kit Plus, SX, DX, MX, and WebEx Board) can be registered to the WebEx Calling Cloud for signaling and SIP URI dialing. This allows a video endpoint to call another endpoint or call into a WebEx meeting using a URI.

(A URI looks like an email address. For example, I might set up a URI as dx80@joshualearn.com for folks to reach me on a Cisco DX-80. The URIs on WebEx are actually a bit longer. One might look more like dx80@joshualearn.rooms.webex.com on WebEx. Most of the time folks don't type out the URIs so that's not usually a concern.)

So what happens when you want to dial a PSTN telephone number?  How do we associate a PSTN telephone number to a video device?  Previously, the only option was to turn up a SIP trunk with a telco provider within the WebEx Control Hub.  Then one would order new DIDs, assign a DID, and assign an internal number to a video endpoint.   That process involves starting a service contract with a telco provider that has an established relationship with Cisco WebEx. Cisco has termed this type of company as a "Cisco Preferred Media Provider".

While that is still an option, some folks have asked how they can bring their own PSTN by integrating with their existing on-premises Cisco Collaboration investment.  Before there was not an option.  Now those folks are in luck.  Utilizing what's known as Cisco WebEx Hybrid Call Service the Cloud Video environment can be married to On-Premises Calling environment.

What's involved to make that work? I'll write a follow-up post with more detail.  In the meantime, here is a high-level list of what's under the hood.
  • Cisco WebEx Cloud registered video endpoints
  • Cisco Unified Communications Manager solution on-premises
  • On-premises PSTN connectivity via a Cisco Voice Gateway or CUBE
  • Cisco Expressways
    • Cisco Expressway-C Services Host
    • Cisco Expressway-C Traversal Host
    • Cisco Expressway-E Traversal Host (with 2 NICs)
  • Firewalls
    • 2 firewalls are recommended, however, I have also seen it designed with 1.
    • Firewall policies
    • NAT
  • DNS
    • A records
    • SRV records
  • Certificates
    • Internal certificates are allowable however the WebEx Control Hub needs to trust the issuing CA chain.

Part 2 of this blog series will include more of the detail on some of the specifics related to the "how-to" design and deploy the solution.

If you have tried this already, have thoughts about trying this, or just like to leave comments, please leave one below.

Integrating WebEx Calling and Communications Manager Express 2/2

This is the second post in the two post series. It will go into more detail on the configuration of the solutions and workarounds put in pla...