Tag Archives: lua

Lua Script Solves Call Forward Masking

Had to implement an Auto Attendant in Unity today.  The client had a requirement to send after-hours calls out to the PSTN, but had to be routed in to their Call Centre, which had it’s own IVR which matched on calling number, that needed to be the switchboard number for their site.

Call Flow was as follows:

  1. Inbound call from PSTN
  2. CTI RP re-direct to Unity
  3. Unity Call Handler, with Close and Holiday transfer to 2nd Call Handler
  4. After-Hours Call Handler transfers back to After-Hours CTI RP
  5. After-Hours CTI RP as Call Forward to a PSTN number

 

The behavior that we see is controlled by SIP Trunk Calling Party Selection parameter, which is set to “Originator” by default:

pic1

 

The SIP INVITE sent to the CUBE looked like this:

INVITE sip:0112223344@mygateway.mydomain.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.5:5060;branch=z9hG4bK55cd222ad94e
From: <sip:0255667787@myhostname.mydomain.com>;tag=336868~0d10edf8-0553-4009-a728-c065d8394e46-24136542
To: <sip:0112223344@mygateway.mydomain.com>
Date: Mon, 07 Dec 2015 10:38:00 GMT
Call-ID: 94dfb980-66516188-5120-247410ac@192.168.5.5
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM11.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:192.168.5.5:5060>;method=”NOTIFY;Event=telephone-event;Duration=500″
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Session-ID: 97632eb36d62419cbdc5fe71aa336867;remote=00000000000000000000000000000000
Cisco-Guid: 2497689984-0000065536-0000000110-0611586220
Session-Expires: 1800
Diversion: “After-Hours Re-Direct” <sip:**999@mydomain.com>;reason=unconditional;privacy=off;screen=yes
P-Asserted-Identity: <sip:0255667787@myhostname.mydomain.com>
Remote-Party-ID: <sip:0255667787@myhostname.mydomain.com>;party=calling;screen=yes;privacy=off
Contact: <sip:0255667787@192.168.5.5:5060>
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 227

v=0
o=CiscoSystemsCCM-SIP 336868 1 IN IP4 192.168.5.5
s=SIP Call
c=IN IP4 172.16.115.23
b=TIAS:8000
b=AS:8
t=0 0
m=audio 26842 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

 

  • In this case the calling party is sent out at 0255667787.  We want this sent out as 0255667000 to me the client’s requirement.
  • To do this, we need to correct the From, PAI, RPID, and Contact headers
  • The Diversion Header contains the Internal DN of the CTI RP – not very pretty, and something that our ITSP may bark at!  We must change it.
  • The headers MUST ONLY be modified when the call is forwarded FROM THIS CTI RP – my “hook” here will be the offending Diversion header.

 

I opted for a Lua script.  I find complex SIP Profiles unwieldly, which could have been another option.  Vomit.

 

So, here’s the script that achieved it.  Pretty simple in fact:

M = {}

trace.enable()

function M.outbound_INVITE(msg)

— Input for script
local ctirpdn = scriptParameters.getValue(“ctirpdn”)
local mainnumber = scriptParameters.getValue(“mainnumber”)
local rdnis = scriptParameters.getValue(“rdnis”)

— Get Diversion header
local diversion = msg:getHeader(“Diversion”)

— Check if Diversion Header exists and if the CTI RP DN is matched
if diversion and diversion:find(ctirpdn) then
trace.format(“Successful match on diversion – applying masking”)

trace.format(“CTI RP DN is : %s”, ctirpdn)
trace.format(“Main Number is : %s”, mainnumber)

— Apply masking to affected headers – From, PAI, RPID, Contact
msg:applyNumberMask(“From”, mainnumber)
msg:applyNumberMask(“P-Asserted-Identity”, mainnumber)
msg:applyNumberMask(“Remote-Party-ID”, mainnumber)
msg:applyNumberMask(“Contact”, mainnumber)
— Apply masking to Diversion to mask internal CTI RP DN
msg:applyNumberMask(“Diversion”, rdnis)

end
end

return M

The bold script params were applied on the SIP Trunk.  I decided to use Unity as the forwarding station, as set this as the “rdnis”.  The key is the IF STATEMENT, that only applied changes if the CTI RP DN is matched in the header.  Otherwise, the script is skipped.

 

Now, for testing… Let’s open up RTMT and trace this in SDL:

 

 

03149762.001 |13:00:37.979 |AppInfo |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 192.168.1.3:[5060]:
[990360,NET]
INVITE sip:0112223344@mygateway.@mydomain.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.5:5060;branch=z9hG4bK5634188efa43
From: <sip:0255667000@myhostname.@mydomain.com>;tag=337995~0d10edf8-0553-4009-a728-c065d8394e46-24136584
To: <sip:0112223344@mygateway.@mydomain.com>
Date: Mon, 07 Dec 2015 11:00:37 GMT
Call-ID: bdb57e00-665166d5-516d-247410ac@192.168.5.5
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM11.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:192.168.5.5:5060>;method=”NOTIFY;Event=telephone-event;Duration=500″
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Session-ID: 97632eb36d62419cbdc5fe71aa337994;remote=00000000000000000000000000000000
Cisco-Guid: 3182788096-0000065536-0000000119-0611586220
Session-Expires: 1800
Diversion: “After-Hours Re-Direct” <sip:0255667800@@mydomain.com>;reason=unconditional;privacy=off;screen=yes
P-Asserted-Identity: <sip:0255667000@myhostname.@mydomain.com>
Remote-Party-ID: <sip:0255667000@myhostname.@mydomain.com>;party=calling;screen=yes;privacy=off
Contact: <sip:0255667000@192.168.5.5:5060>
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 227

v=0
o=CiscoSystemsCCM-SIP 337995 1 IN IP4 192.168.5.5
s=SIP Call
c=IN IP4 172.16.115.23
b=TIAS:8000
b=AS:8
t=0 0
m=audio 22640 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

 

Working perfectly! 🙂

 

#dontcalltac

 

 

 

 

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Lua Scripting – Replacing FQDN with IP in SIP Headers

A friend of mine at Cisco TAC pinged me a few days ago, asking to assist him with a case he’s working on that required a sip normalization script.

The call flow included a SIP Trunk to a conferencing system that was DNS-enabled.  CUCM was unable to lookup the DNS address (in this case due to a bug), and since the system was integrated directly to CUCM (not via CUBE), the only option for normalization was a Lua script.

The requirements of the script were:

  1. Find/Replace a number of potential FQDN/IP combinations
  2. Apply required normalization to “Contact” header
  3. Contact head must be modified for initial inbound INVITE, and all subsequent inbound responses to the INVITE
  4. Other messages should be ignored

Script:

–[[

Author : Jonathan Els
Email : jonathanelscpt@gmail.com
Version : 3.0

Description:

Find and replace FQDN with IP in Contact headers.
Script is applied to Inbound INVITEs and all Inbound INVITE responses

Future development:

None planned

–]]

M = {}

trace.enable()

— Customer-specific DNS table initialization – edit on per-script basis
local dns = {}
dns[“1.1.1.1”] = “hostname1%.rootdomain%.com”
dns[“2.2.2.2”] = “hostname2%.rootdomain%.com”
dns[“3.3.3.3”] = “hostname3%.subdomain%.rootdomain%.com”
local function process_contact_headers

— Get Contact header
local contact = msg:getHeader(“Contact”)
local iptest = “@(%d+%.%d+%.%d+%.%d+)”

— Check if exists and if URI host portion is FQDN
if contact and not contact:match(iptest) then
trace.format(” — Contact URI host portion matched FQDN”)
trace.format(” — Contact header is : %s”, contact)

— Iterate over domain and substitute if matched
for ip,fqdn in pairs(dns) do
if contact:match(fqdn) then
contact = contact:gsub(fqdn, ip)
trace.format(” — Matched on : %s”, fqdn)
trace.format(” — Modified to : %s”, ip)

— Modify contact header
msg:modifyHeader(“Contact”, contact)
break
end
end
end
end

— Apply to INVITE request and all INVITE responses
M.inbound_INVITE = process_contact_headers
M.inbound_ANY_INVITE = process_contact_headers
return M

Some Notes:

  • The requirement called for both request and response handling
    • M.inbound_INVITE
    • M.inbound_ANY_INVITE
  • The multiple handling requirement resulted in a “purposed-based” function being used to avoid duplication of code.
  • Some checks were added to skip and exit in the event that
    1. Contact header did not exist
    2. URI host portion was an IP Address.
  • Lua does not support full regex (regex is slow), so a standard IP approximation was used.  The “@” prefix in the match is used to ensure the match is on the URI.

Useful Resources:

The last resource from Cisco Live is possibly the best “How-To” out there!

Happy scripting.

#dontcalltac

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Lua Scripting – Proxied Manager Breaks MWI for Centralized CUC Behind SME

I’ve been working as design and deployment lead on a globally significant full Collaboration on-premise deployment for the last 3 months (think 120 CUCM across 9 Leaf Clusters on 101 UCS C240’s) that’s been keeping me quite busy 🙂

As part of the deployment design phase, a decision was taken to centrally host Unity Connection for all 9 Leaf Clusters.  As I commenced with the dial plan design, while I reviewed the business requirements I identified a call flow that I knew would immediately create a problem.  Some like a board, pen and paper or even spreadsheets for call routing – I like to do these in my head as I jot down the configuration elements in notepad++.  Your preferred method is irrelevant – the point is always to model your call flow design prior to implementation – with the intention to find inherent call flow failures.

The client had a strong use case for a proxied manager configuration, with a secretary available to take calls on their behalf.  There are a number of ways to configure this, but from a routing perspective, a separate partition is always required for the manager, with reach-ability only from the secretary’s CSS.

The Challenge:

  • Call flow required is CUC > SME > Leaf > Manager Phone
  • MWI is routed from CUC via an Out-of-Dialog SIP NOTIFY
  • In the CUCM SIP construct, this uses the same Inbound CSS as a standard call

The Likely Result:

  • The manager’s MWI will never light up – no reach-ability in the inbound CSS without violating an inter-cluster proxied routing requirement
  • Depending on your configuration, the proxied line on the secretary’s phone is probably going to light up instead!

The requirement is not unique, but the advent of SIP, and wanting a scale-able solution posed a challenge.


Due to the scale of deployment and tight timelines, our local SE was kind enough to arrange some time with a BU dial plan specialist to review my dial plan design, with some recommendations along the way (what an awesome experience!).  I was keen to pose my dial plan issue to the BU, as I was quite sure that a special method would be needed to accommodate this.

After some discussion, a Lua script was proposed.  I was quite keen on the idea, as I’ve had some experience writing these before.  The call flow idea was actually quite simple:

  1. CUC > SME > Leaf (end-to-end SIP)
  2. SIP Normalization Script (on Leaf Trunk) > Custom Routing Prefix on Manager DN (think “**”)
  3. Translation Pattern (PreDot strip with correct re-prefixing) > Custom CSS with prioritization of Manager Partition(s)

The BU provided me a script for a similar case, which I customized to support my requirement:

–[[

Author : Jonathan Els

Description:

Script for separate handling of MWI to accommodate proxied partitions for centralized voicemail designs with SME – e.g. with Proxied Manager/Secretary configurations
Script strips initial chars (“0” hard-coded in this script) to add a prefix for custom routing

Future development:

1. Add contextual checks to specifically limit to MWI – possible matching on Content (“Messages-Waiting” etc.) or CUC source address could apply
1. Use prefix as script param.

–]]
M={}

trace.enable()

function M.inbound_NOTIFY(msg)

— Set prefix for MWI
prefix = “**”

— Need to convert E.164 to **-prefixed string in request URI and To header
local m,r,v = msg:getRequestLine()
r=string.gsub (r,”sip:0(%d*)@”, “sip:” .. prefix .. “%1@”)
msg:setRequestUri (r )

local t = msg:getHeader(“To”)
t=string.gsub (t,”sip:0(%d*)@”, “sip:” .. prefix .. “%1@”)
msg:modifyHeader (“To”, t)
end

return M

In our dial plan, we deviated from the +-E.164 CVD best practice due to a limitation on the client’s billing system.  We preferred localized DID’s instead.  As a result, the above script uses the following pattern match:

sip:0(%d*)@”

Without excessive deviation, if your call routing requirement dictates the need for Lua scripting, please take the time to understand Lua Pattern Matching.  Lua’s about *speed*.  Regex is damn slow.


During testing, I hit a snag and reached out to the BU once again…

Looking at SDL showed the script to be working:

00240075.000 |17:09:45.622 |SdlSig   |SIPNormalizeReq                        |wait                           |SIPNormalization(2,100,78,1)     |SIPHandler(2,100,79,1)           |2,100,14,91.8^192.168.116.12^*           |*TraceFlagOverrode

00240075.001 |17:09:45.622 |AppInfo  |//SIP/SIPNormalization/trace_sip_message: After inbound SIP Normalization msg is:

[20427,INT]

NOTIFY sip:**111001001@192.168.116.21:5060 SIP/2.0

Date: Wed, 24 Jun 2015 15:09:45 GMT

From: <sip:voicemail@192.168.116.12>;tag=36349161

Event: message-summary

Content-Length: 23

User-Agent: Cisco-CUCM10.5

To: <sip:**111001001@192.168.116.21>

Contact: <sip:voicemail@192.168.116.12:5060;transport=tcp>

Content-Type: application/simple-message-summary

Call-ID: ad6c800-58a1c839-13de1-c74a8c0@192.168.116.12

Subscription-State: active

Via: SIP/2.0/TCP 192.168.116.12:5060;branch=z9hG4bK13ecf35a0eda8

CSeq: 102 NOTIFY

Max-Forwards: 70

 

Messages-Waiting: yes

However, I soon found that the standard digit analysis engine was not being applied.  The TP’s in my CSS’s were not being matched!  I actually confirmed this by configuring my expected pattern – **111001001 – on an entirely different phone and watched its MWI light up.

The BU suggested that I try setting the Advanced CallManager Service Parameter Multiple Tenant MWI Mode to “True”.  In our initial discussions, I’d been informed that this was not required in the SIP Trunking use case, but this ultimately did not prove to be the case.  After doing this, the translation pattern kicked in and that elusive MWI lit up immediately.

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