Pablo is using his Cisco 7961 IP phone. He goes to look at his 'Missed Calls' and sees that he missed a call that had come in from a local PSTN number. The number in his missed call looked exactly like this:

He needs the ability to look at this missed call and simply press the 'Dial' softkey --and nothing more-- to return the call immediately - with no inter-digit timeout, and of course successful call routing back to the PSTN caller.

Pablo has the same CSS on his Line, as he does on his Phone Device - which is:

And the CSS has in it, the Partition of:


There is a Route Pattern that points straight to a gateway, that matches his desired return call:
Pattern: \+!

Partition: PT_CorpHQ-LOCAL

Urgent Priority: YES

Gateway: CorpHQ-MGCP-GW

And the MGCP Gateway has a Called Party Transformation CSS:
Gateway: CorpHQ-MGCP-GW

Outbound Called Party Transformation CSS: CSS_CdPTP-CorpHQ-GW

And the CSS with CSS has the PT:

--> PT_CdPTP-CorpHQ-GW

And the CdPTP to localize the call egress from the MGCP GW to the carrier (the carrier is expecting 10 DNIS digits) is:
Called Party Transformation Pattern: \+1.206!

Partition: PT_CorpHQ-LOCAL

Urgent Priority: YES!

Discard: PreDot

So as you can see, the call should be returned to the PSTN with no issues.

So Pablo looks at that missed call, and goes to press 'Dial'. No sooner than he has pressed it, his display quickly pulses out the digits '+12065015111' -- then it briefly changes to show only '+1' -- and then the call simply disappears, as the screen returns to a normal, idle state.

He is quite shocked at this, most especially because of the fact that Tom, his cube mate, has the same Cisco 7961 IP phone, with the same dialing permissions (Tom also has the exact same CSS as Pablo, on both his Line and Phone Device), and Tom has absolutely no problem returning this exact same missed call pattern.

There is one small difference between Pablo and Tom's phones -- Tom has a 7961 SCCP phone, while Pablo has a 7961 SIP phone.

BTW, I should probably also note that neither one of them has any trouble placing any sort of normal, local calls when dialing from their keypads.

So, your contest question for the weekend is a three-parter:
  1. Why does Tom's phone work, but Pablo's does not, when attempting to return the very same missed call?
  2. What CUCM entity could you add to this equation to make Pablo's phone work for returning this call (without changing any of the entities or values listed above)?
  3. What would the value be of this newly created entity?
The winner of this contest will have their choice of any one of these items:
  • $100USD Amazon Gift Card
  • $100USD in GradedLabs Tokens (which is just a bit more than 3 Voice rack sessions)
  • $100USD worth of INE.com online store credit
The way we will hold the contest is as follows:
  • You must answer all questions correctly, submitting your answers in the comments section of this post
  • If there are multiple, correct respondents, then we will place all of the correct respondents names into an online randomizer - the modern day 'hat', if you will
  • We will not allow any of the response comments to be seen here publicly until Monday morning when we choose the winner, so as not to give an unfair advantage to anyone
  • It might be good if you posted with a registered email account -- you know, that way we can notify of your glorious prizes should you win? :-)
I'll return on Monday to gather up the winners, choose a random name, and post all of the comments with my own replies and comments.

Also, we will be holding these sorts of contests bi-weekly, so check back often to increase your chance of winning!

Good Luck!
Mark Snow

Congratulations to Roger Carpio, the winner of this contest!

As promised below in my comments, here is an updated post with the screenshot for the SIP Dial Rule to both fix this problem, and avoid a 10 second inter-digit timeout for all other normal keypad dialed calls:

Click to see full-sized image

Mark Snow, CCIE #14073
About Mark Snow, CCIE #14073

You might say that Mark Snow began his networking career at the age of five, when his father, a patented research scientist at AT&T Bell Laboratories, started sharing his knowledge with Mark. He has been working with data and voice technology ever since, beginning with Unix System V and basic analog telephony and progressing to large data networking projects and large phone systems in both enterprise and 911 PSAP environments around the world. You’ll see Mark in all of INE’s Voice video courses and live Bootcamps. Mark Snow is also an accomplished pilot, and when he isn’t learning, labing, consulting, or teaching, he can be found jumping out of a perfectly good airplane, hanging off a rock somewhere, skiing out west, or just enjoying a quiet day at the beach with his wife and two wonderful kids. You may contact Mark Snow at msnow@ine.com, follow him on Twitter, or find him helping others in INE’s IEOC Community Forum.

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