911 calls failing

[pbrackett1]

I am working with a network phone system running CUCM 10, alongside CER and some other similar systems. Inside the phone network, calls to 911 are failing. The caller gets a rapid busy signal when they try. I can see the attempts in the CER call history, and I do receive the emergency alert notification ("Alert, caller at extension XXXX is in emergency."). The call router (a Cisco C2911) shows the calls being made (see the attached log for an example). My telecom is reporting that the calls are being denied because they are not properly defined. They have pointed to a couple of specific lines in the log file from my router:

Called Party Number i = 0x80, '911'
Plan:Unknown, Type:Unknown

Anyone have any ideas. I'm in over my head and could really use some assistance.

[Daniel Bosch]

Sorry to hear that you're in this situation.  You explained things well enough that I assumed your level of expertise was a little more extensive (take this as a compliment.)  If you have CUCM GUI access but can only guess if SIP or MGCP is being used, you're definitely in over all our heads in this case. Unfortunately, it's not likely you'll be able to get the level of help you need in a forum for something this complex if our suggestions seem like a foreign language.  If it were me, this is what I would do next:

1. Find out who does have CLI access on the C2911 and have him/her ready for one of the next 2 steps.

2. Open a case with Cisco TAC to request configuration assistance (you do have a support contract, right?)

3. Plea your case to get professional services help from a local or national consultant who can help with CUCM/VoIP.

If you have to go with option 3 only - you might use this situation to convince your company that paying for support is worth it.  I'd bet it would be a lot less expensive than litigation from a failed 911 call that ended up costing someone dearly, right?

[pbrackett1]

The C2911 is probably the only piece in this whole infrastructure that I can access via CLI. I have a logging computer hooked up to the thing right now in my server room, for logging the calls that I make to 911 to watch how they fail. So, would I create the e164 pattern map on the C2911? I feel like I am so close to getting this figured out (thanks to the help of all of you). Cisco TAC won't help because we don't have a support contract and I'm not sure what else to try.

[Roger Kallberg]

Before you commit to any changes in configuration it would be best if you could share the current configuration with us so that we can see what type of gateway you have. Please save it in a text file and attach it to your response.

[pbrackett1]

Attached is the running config for my C2911. Thanks for taking a look.

[Roger Kallberg]

What we have suggested is for SIP, that would not work for you as you're having an H.323 gateway. It was many years ago since I last did something with H.323 gateways, so of the top of my head I don't know how to handle this in such a gateway.

[Steven L]

all my H323 notes are on stone tablets

[pbrackett1]

UGH. I was hoping I would be able to keep this system running. Thanks, everyone, for trying to help.

[Roger Kallberg]

You can convert your gateway to SIP. It is a quite simple process. Basically you need to change the protocol on your voip dial peers and create a SIP trunk in CM that you add to the same RG as where your current H.323 gateway is placed in. For settings on the trunk, like inbound CSS and so on, look at the values used on the H.323 gateway and use the same.

[pbrackett1]

So, Roger --> I need to 1) "change the protocol on your voip dial peers" & 2) "create a SIP trunk in CM that you add to the same RG as where your current H.323 is placed in"

Would the changes for #1 happen in the config of the C2911? It can't possibly be as simple as replacing all of the H323 dial peers in that config with SIP dial peers? Could you explain that out a little bit, please?

For #2, I could probably Google some instructions for creating a SIP trunk in CUCM, but how do I "add [it] to the same RG as where your current H.323 is placed in"? Is this a route group? A little bit more, please?

Thanks so much.

[Roger Kallberg]

For item 2 have a look at the current H.323 gateway and look at it's dependency records. This is an example of this from one of our CMs.

For item 1, aka the changes needed in the gateway this could work for you.

voice service voip
 no allow-connections h323 to h323
 no allow-connections h323 to sip
 no allow-connections sip to h323
 default supplementary-service h450.2
 default supplementary-service h450.3
 default supplementary-service h450.12
 sip
  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0
  early-offer forced
!
voice class e164-pattern-map 1
 description E164 Pattern Map for called number to CUCM
  e164 2694610083
  e164 2694610088
  e164 2694613121
  e164 2694613351
  e164 2694613567
  e164 2694614350
  e164 2694616191
  e164 2694616947
  e164 2694616997
  e164 26959338..
  e164 269593390.
  e164 2694616577
!
dial-peer voice 102 voip
 no preference 1
 no destination-pattern 2694610083
 destination e164-pattern-map 1
 no voice-class h323 1
 session protocol sipv2
 dtmf-relay rtp-nte sip-kpml
!
no dial-peer voice 103 voip
!
no dial-peer voice 104 voip
!
no dial-peer voice 105 voip
!
no dial-peer voice 106 voip
!
no dial-peer voice 107 voip
!
no dial-peer voice 108 voip
!
no dial-peer voice 109 voip
!
no dial-peer voice 110 voip
!
no dial-peer voice 111 voip
!
no dial-peer voice 112 voip
!
no dial-peer voice 113 voip
!
voice class e164-pattern-map 911
 description Emergency numbers to PSTN
 e164 911
 e164 9.911
!
dial-peer voice 911 voip
 description Incoming dial peer for 911 EMERGENCY
 session protocol sipv2
 incoming called e164-pattern-map 911
 voice-class codec 1  
 dtmf-relay rtp-nte sip-kpml
 no vad
!
voice class e164-pattern-map 100
 description Standard numbers to PSTN
 e164 9[2-9]......
 e164 9[2-9]..[2-9]......
 e164 [2-9]..[2-9]......
 e164 91[2-9]..[2-9]......
 e164 1[2-9]..[2-9]......
 e164 9011T
!
dial-peer voice 100 voip
 description Incoming dial peer for standard PSTN calls
 session protocol sipv2
 incoming called e164-pattern-map 100
 voice-class codec 1  
 dtmf-relay rtp-nte sip-kpml
 no vad
!
interface GigabitEthernet0/0
 no h323-gateway voip bind srcaddr 10.19.254.2
 no h323-gateway voip interface
!
no voice class h323 1
!
sip-ua 
 no remote-party-id
 retry invite 2
 timers trying 300
 registrar ipv4:10.19.254.2 expires 3600
 g729-annexb override
!

For more details on how call routing in IOS operates have a look at this document Explain Cisco IOS and IOS XE Call Routing and for information on how to configure a SIP trunk in IOS have a look at this document Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 

A general observation on this whole interaction, based on what you write I think that you might be in over your head with this and if I where you I would reach out to a reputable Cisco partner that has a specialization in Collaboration to procure their help with this.

[pbrackett1]

So, I apparently already have some SIP trunk profiles in my CUCM, if that helps my situation at all.

[Roger Kallberg]

Not relevant to this as such. When you create the SIP trunk it will use a SIP trunk security profile as that is one of the mandatory configuration elements.

[pbrackett1]

So, I found the route group for the H.323 gateway --> listed as "HS_RG". I created a SIP trunk and gave it the default settings. But, when I pull up the settings for the HS_RG route group configuration, I can't add the SIP trunk that I created (called SIP_Trunk_ECPS_001) because it doesn't show up in the list of devices.

[pbrackett1]

WAIT. I think I did that wrong.

[pbrackett1]

So, I hadn't created a SIP before; I'd created a SIP Trunk security profile (oops). I corrected myself and created a SIP trunk and gave it the default settings. Then, I added it to the RG that the H.323 gateway is on. So, Roger --> I think I've done #2 of your suggested process. Any thoughts, or should I work on editing the config of my C2911?