VOL1 3.7 Globalization
Globalized Call Routing Step 8: Device > Gateway. Select the MGCP Gateway for R1. Choose the Called and Calling party transformation CSS's
It shows CSS_HQ_CALLED and CSS_HQ_CALLING as the two used. We were not instructed to create these CSS's a few steps earlier. It would have to be which two of the following?
CSS_HQ_CALLED_IN
CSS_HQ_CALLED_OUT
CSS_HQ_CALLING_IN
CSS_HQ_CALLING_OUT
Comments
Hmm, this task (3.7) feels very "disconnected" from all the previous tasks. I had to undo everything I did in 3.1 - 3.6 as this one uses all new PT and CSS. I'm going to go through it again but at this point nothing is working.
On Dec 14, 2009, at 20:05, MarkH <[email protected]> wrote:
It's Called_out per Josh's reply to someone a week or so ago.
http://ieoc.com/forums/p/9294/82526.aspx#82526
and yeah, i actually just did globalization in another session...everything is different.
A couple of things to note. Josh's verification in the other post is for Step 7. I went with Mark's suggestion for Step 8 and assigned CSS_HQ_CALLED and CSS_HQ_CALLING to Device > Gateway > Transformation CSS. When I place a call from HqPh1 to PSTN 3122012222 I get 'Your call cannot be completed as dialed.'
Below are ISDN debugs on both r1-hq and PSTN routers for a test call that fails.
EDIT: VoIP to PSTN is still not functioning but PSTN to VoIP is working as expected. For the very last step I tried pattern +1[2-9]XX.[2-9]XXXXXX just to make sure my head is screwed on straight. Thankfully it worked. Now, that pesky VoIP to PSTN is really gettin' to me. Will resume in the morning. [|-)]
My apologies about the reference to the wrong step.
As for the calling number, somewhere the "Use External Phone Number Mask" box is not checked...or, is it set on extension 1001 at all?
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@aquabrown - No worries mate. Just want to make sure the INE folks know to update both step 7 and 8.
@mark - From what I can tell the CSS provided in your response for Step 8 is correct. Calls towards the PSTN are making it from CCM > r1-hq > PSTN router, but they are failing to ring the PSTN phone due to the following (debug q931 on PSTN router)
Called Party Number i = 0xA0, '+13122012222'
Dec 15 16:08:23.579: ISDN Se0/2/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x8004
Cause i = 0x8281 - Unallocated/unassigned number
So, the cause code for call setup failure on PSTN router is 'Unallocated/unassigned number'. If the call is sent to the PSTN router without the + symbol the PSTN phone rings. Does something need to be different on the PSTN router compared to the previous 3.x tasks to support + in the called number and ring the PSTN IP phone (7971)? My PSTN router config is from the Members area and all the default configs are dated August 11. Just want to make sure there isn't something missing. The default PSTN router config from the member's area is essentially a slim CME config with one IP phone and 6 ephone-dn's.
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Hey Mark,
You are correct, the + is not accepted as a valid digit in the DNIS
(Called Number) field for the gateway. The CSS on the gateway for the Called
Party Transformation CSS should be CSS_HQ_CALLED_OUT which should contain PT_HQ_CALLED_OUT
which should be assigned to a Called Party Transformation Pattern of +.! With Digit
Discard set to PreDot.
This way all calls sent to the gateway will have the + stripped from the
called number.
HTH
Thanks everyone, I am back on track now. I rolled back my VM's and went through the entire lab again and completed everything. For those of us using our own lab equipment there are some uncertainties going into this task because it doesn't start where the previous task left off. Overall I would say this is a great, but challenging, task if you have never worked with + before. However, it warrants further reading in my opinion. There are 3 guides that include excellent information. UCM7 SRND provides a good overview, UCM7 Features and Services Guide and UCM7 System Guide provide excellent technical explanations.
On Dec 15, 2009, at 21:35, MarkH <[email protected]> wrote:
It's not just because you have your own equipment.....INE doesn't do the "config files for each exercise" bit so it's awkward for everyone (I'm not complaining, btw). It would have been nice to be its own "section" but that wouldn't have fit with the way the guide is organized.
Anyway, as I mentioned, the way I did it was by doing the first 6 tasks as one lab and then a different lab session later on *starting* with 3.7. So I didn't have to worry about "leftovers" getting in the way of my execution/understanding of the task.
Glad to see Part 2 of the dial plan series up as well [Y]
So, I rolled back my VM's and I'm going through several 3.x tasks again to work on my speed. One thing I didn't note about task 3.7 before rolling back is what CSS should r1 and r2 be associated with? I'm stuck, where VoIP to PSTN works but not PSTN to VoIP. I know it's an oversight on my part.
The task itself is quite hefty, but since it doesn't pick up where 3.6 left off it feels like in addition to the task we also need to take a stab at what else needs to be undone without any real sanity check to make sure we didn't undo too much to the point the task will never work. Am I making sense? Would it be possible to have a short addendum for configuration modification going into the task?
Device > Gateways > r1-hq, T1PRI > Call Routing Information - Inbound Calls
Significant Digits = 4
CSS <NONE>
AAR
PREFIX
The CSS needs to be able to see the internal partitions. I don't have my WB with me at the moment but make sure the CSS you use includes the internal lines and test from there.
Ok, it's all sorted out thanks to Josh. [Y]
r1-hq T1 port needs to be assigned to CSS_HQ_DEVICE (assuming phones are assigned to PT_HQ_DEVICE)
r2-br1 T1 port needs to be assigned to CSS_BR1_DEVICE (assuming phones are assigned to PT_BR1_DEVICE)
After save/reset, issue no mgcp/mgcp commands on each gateway as stated in troubleshooting section.
At first I didn't realize I needed to assign the gateways to CSS_HQ_DEVICE and CSS_BR1_DEVICE. I should have realized it much sooner. But, even when I did assign CSS I did not initiate the mgcp commands on the routers immediate after which is why I ran into so many instances of it partially working or in some cases, worked on one site but not the other.
That's the one!
Like I said above, you're home free once the call reaches the gateway. Now as long as it can "see" the destination phone, the call will make it all the way. Whatever partition the phones are in should be in the CSS you use for the gateway.
This saga has the added bonus of burning that no mgcp/mgcp tango into your fingers! [8-|]
Just to understand this conversation.. The configs in the book are correct? Please Advise
The configuration in the workbook along with the comments here regarding the CSSs applied to the gateway will work for you. If you have any question, you can e-mail me directly, and I'll be glad to help.
Thanks
Thanks