Re: Call Flow from Unity to CUCM out to Cell phones for Unity Web Inbox Phone feature

RAustin70 · ‎07-17-2020

I'm brain-dead here today. A long way back, I set up the Unity Web Inbox and it works great for on-site customers when they choose 'phone' and put in their internal number.

Now, with the pandemic, many employees are teleworking and want to use their cell phone in that field and it isn't working.

I can set call-forwarding on the desk phones to their cell phones, but the calling party ID '3153300809' is not in our owned DN group, so our PSTN Provider (yes, we are still in T1 dark ages) drops that ID and inserts the billing DN and address to the call.

so 1. I can't find where to open up Unity to allow off-site calling from the web inbox. Checked the restriction tables and they look good

and 2. I don't remember where I set up that 3153300809 number as the Unity calling party ID so I can't change it to an owned DN. I've looked in SCH's, looked in the Port Groups where the Remote party-ID lives, etc.

Below is a test call coming across my gateway on it's way to my cell phone, you can see it originally calls '7447' from '3153300809' and my 8865 is redirecting the call to my cell 3158885555

1004444: Jul 17 12:00:40.372 EDT: ISDN Se0/1/2:23 Q931: RX <- SETUP pd = 8 callref = 0x2618
Bearer Capability i = 0x9090A2
Standard = CCITT
Transfer Capability = 3.1kHz Audio
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA18381
Preferred, Channel 1
Facility i = 0x9F8B0100A10F02010106072A8648CE1500040A0100
Protocol Profile = Networking Extensions
0xA10F02010106072A8648CE1500040A0100
Component = Invoke component
Invoke Id = 1
Operation = InformationFollowing (calling_name)
Name information in subsequent FACILITY message
Progress Ind i = 0x8483 - Origination address is non-ISDN
Calling Party Number i = 0x2183, '3153300809'
Plan:ISDN, Type:National
Called Party Number i = 0x80, '7447'
Plan:Unknown, Type:Unknown
1004445: Jul 17 12:00:40.377 EDT: ISDN Se0/1/2:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0xA618
Cause i = 0x8081 - Unallocated/unassigned number
1004446: Jul 17 12:00:41.566 EDT: ISDN Se0/1/3:23 Q931: TX -> SETUP pd = 8 callref = 0x0B95
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98394
Exclusive, Channel 20
Calling Party Number i = 0x2183, N/A
Plan:ISDN, Type:National
Called Party Number i = 0x80, '3158885555'

Any insight would be greatly appreciated.

Rob

Maren Mahoney · ‎07-17-2020

If you have a SCCP integration, the callerID and CSS on the voicemail ports will be part of the equation when dialing out from Unity Connection.

If you have a SIP integration, take a look at the Rerouting Calling Search Space on the SIP Trunk.

Other things may be in play in your environment, but I'd suggest starting there.

Maren

Jaime Valencia · ‎07-17-2020

You use TRAP ports for that

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/11x/integration/guide/cucm_sip/b_cucintcucmsip/b_cucintcucmsip_chapter_0100.html

HTH

java

if this helps, please rate

RAustin70 · ‎07-20-2020

We are SIP integrated here, I will check the rerouting CSS. TRAP, all my ports are configured for TRAP and it works fine on-site for recording and playback. I don't see any other TRAP configs beyond checking TAP enabled on the ports.

If a customer uses their Cell phone in the Web Inbox, the call fails, leads me to look at the CSS for this one.

If an internal DN is set to forward to an off-site location, and the customer uses the on-site DN for Web Inbox, the call does go through, but the commercial carrier uses the Billing Telephone Number (BTN) because they do not recognize the Calling Party ID as a valid DN coming from our system. So I want to find where that setting lives and set it to a valid DN if possible so I can set up a caller-ID for the DN that the customer will recognize.

RAustin70 · ‎07-20-2020

I guess what it all comes down to, is how do I change

"Calling Party Number i = 0x2183, N/A"

to

"Calling Party Number i = 0x2183, 3151234567"

For calls coming from our Unity Connections Server, into CUCM over SIP trunks, out to PSTN through MGCP controlled Voice Gateways

Maren Mahoney · ‎07-20-2020

Called and Calling Party information can be set in CUCM. You can use the Dialed Number Analyzer to see what is happening. Start with the information being sent to CUCM by CUC in the trace file. Transformations can be:

At the Route Pattern being targeted with the outbound call.
As the Route Group is added to the Route List after a Route Pattern is matched
Using transformations on the SIP trunk itself. (Look for "Called Party Transformation CSS" and "Calling Party Transformation CSS".)
(Keep in mind that only one of these takes effect - the "last" one configured in the order above)

Called and Calling Party information can be set at the router/SBC.

Called and Calling Party information can be set by your provider.

It will take a step-by-step analysis to figure out what your system is doing.

Maren

RAustin70 · ‎07-22-2020

Okay so I corrected Call Flow by creating a new CSS for the inbound Calls CSS on the CUC SIP Trunk that adds the PSTN Partition.

For Reference, the call flow I am discussing goes CUC ---> SIP Trunk ---> CUCM ---> MGCP T1 --->Telco

conversation trace from CUC shows this:

##### invite from CUC sent to CUCM From: sip:111.222.333.20:5060
09:29:30.364 |23915,PhoneSystem-1-070,,MiuSIPStack,11,Outgoing Sip Message--> INVITE sip:993158682701@111.222.333.10 SIP/2.0
09:29:30.364 |23915,PhoneSystem-1-070,,MiuSIPStack,11, From: sip:111.222.333.20:5060;tag=1a3befd654dc49b7bddc837e893b0f17
09:29:30.364 |23915,PhoneSystem-1-070,,MiuSIPStack,11, To: sip:993151234567@111.222.333.10
09:29:30.364 |23915,PhoneSystem-1-070,,MiuSIPStack,11, Via: SIP/2.0/TCP 111.222.333.20:5060;branch=z9hG4bK5860359a5e924b84a2610885f43b6003
09:29:30.364 |23915,PhoneSystem-1-070,,MiuSIPStack,11, Max-Forwards: 70
09:29:30.364 |23915,PhoneSystem-1-070,,MiuSIPStack,11, User-Agent: Cisco-UnityConnection/8.5
09:29:30.364 |23915,PhoneSystem-1-070,,MiuSIPStack,11, Contact: <sip:111.222.333.20:5060;transport=tcp>;automata
09:29:30.364 |23915,PhoneSystem-1-070,,MiuSIPStack,11, Call-ID: 5978009ad37d4d83910a77046ae5307d@111.222.333.10
09:29:30.364 |23915,PhoneSystem-1-070,,MiuSIPStack,11, CSeq: 200 INVITE
09:29:30.364 |23915,PhoneSystem-1-070,,MiuSIPStack,11, Allow-Events: kpml
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,SUBSCRIBE
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, Remote-Party-ID: "AFRL_Voicemail" <sip:111.222.333.20:5060>
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, Content-Length: 280
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, Content-Type: application/sdp
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, v=0
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, o=CiscoSystemsUCXN 1991705096 1991705097 IN IP4 111.222.333.20
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, s=No Subject
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, c=IN IP4 111.222.333.20
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, t=0 0
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, m=audio 20436 RTP/AVP 0 18 101
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, a=rtpmap:0 pcmu/8000
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, a=ptime:20
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, a=rtpmap:18 G729/8000
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, a=ptime:20
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, a=rtpmap:101 telephone-event/8000
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, a=fmtp:101 0-15
09:29:30.365 |23915,PhoneSystem-1-070,,MiuSIPStack,11, a=sendrecv

Unity is only sending a from:<IP of Unity>:5060 instead of <DN of Unity@IP of Unity>:5060

So, I am sort of thinking maybe I play with a normalization script on the SIP Trunks that finds 111.222.333.20, and appends <VMPilotDN@*> to the from field.

Any other ideas on how to assign the DN to the invite?