Avaya Aura System Mgr 7.1/CM 7.1 - Need help with SIP phone dialing outbound to INTL

[marcyg]

Hi,
I have TDM soft and hard phones that work fine (dial 9 011 xxxxx). However, my SIP phones that register to System Manager, can only dial local and toll free. I need them to dial INTL.

1) SIP phones do not require to dial a 9 for an outside line. How do I fix this?
2) When I dial 011 on the SIP phone naturally it only sees the 0 and goes to the operator. How do I fix this?
3) How do I change the outbound caller ID on SIP phones?
4) Does SIP phone use ARS ANA on my CM? Or is there something I have to setup for routing/elements Sys Mgr?

Thank you,
Marcy

 

[ZeroZeroOne]

Yes, System Manager / Session Manager needs to know how to process SIP calls. Take a look under Elements / Routing.

If you are more familiar with CM than Session Manager, you can loosely think of Dial Patterns as "ARS analysis", Routing Policies as "Route Patterns" and Entity Links as "Trunk Groups".
Session Manager will match the dial string to a Dial Pattern, use the associated Routing Policy, and send the call to the correct Entity.

Look at any existing Dial Patterns to get an idea of how they work. You can then create new Dial Patterns to match what you need. For example, you can create a pattern for "9" with a minimum of 12 and a maximum of 20 and route that to CM. That should handle normal long distance (9 and 11-digits) and international (9 011 ...) and route the calls to CM for additional processing via ARS Analysis.

Outbound Caller ID for SIP phones can be set in CM with the "public unknown numbering" OR via an Adaptation under Elements / Routing. If your SIP phones are using PRI or other trunks in CM then you'll definitely need to update the public unknown numbering table.

This is just a very high level to get you started.

 

[kyle555]

If you traceSM on Session Manager, and from SMGR go Session Manager-->System Status-->User Regsitrations and "reload complete" for a SIP phone or 2, you'll see Personal Profile Manager data get pushed to the phone. It's not service affecting.

Thing about SIP is that the phone has to send a complete INVITE which is different than TDM/H323 where the switch processes each dialed digit.

That said, PPM contains all the possible dialable strings for that phone based on your dialplan analysis table, feature codes, aar, ars, etc.

So, if in dial plan analysis you have 2 length 10 UDP and the same for 3/4/5/6/7/8/9, that would explain why PPM would contain dial patterns for the phone that look like 2xxxxxxxxx and 3xxxxxxxxx so the phone knows when it's hit a complete match and should quit waiting for more digits and send the call.

Presumably whatever you got for 0 in there is getting to the phone as 0, a 1 digit number, not 011xxxxxxxxxxxxx where the phone should wait a bit.

When you change dialplan analysis, you need to do a complete init sync of CM to SMGR - more than a nightly incremental one - and then SM gets all that data and crunches it into those strings that your SIP phones recognize.

 

[marcyg]

Thank you both! I'm still struggling here.

This was very helpufl!! "If you are more familiar with CM than Session Manager, you can loosely think of Dial Patterns as "ARS analysis", Routing Policies as "Route Patterns" and Entity Links as "Trunk Groups". Session Manager will match the dial string to a Dial Pattern, use the associated Routing Policy, and send the call to the correct Entity."

I have a dial pattern for 011, min 9/max 36, SIP domain all, there are no denied originating locations, and several originating locations/routing policies. For the Origination Locations and Routing policies, I went into each routing policy and made sure the dial pattern for 011 was included. Maybe I need to type +011 for the dial pattern?


And for the Entity Links/Trunk groups,they are all listed correctly.
a-New Englewood SBC - LD, 10.130.42.35, SIP Trunk, A1 Interface for CenturyLink-LD
a-New Englewood SBC - Local, 10.130.42.34, SIP Trunk, A1 Interface for CenturyLink-Local
Canada - Cisco, 10.2.16.150, SIP Trunk,
Canada - Cisco2, 10.2.16.151, SIP Trunk
Canada - Cisco3, 10.30.9.151, SIP Trunk
Clive SBC - LD, 10.226.0.252, SIP Trunk, New SBC at Clive, IA, for LD
Clive SBC - Local, 10.226.0.251, SIP Trunk, New SBC at Clive, IA, for Local
CliveSM, 10.226.0.17, Session Manager,
Englewood O365, 10.130.42.9,SIP Trunk, ENG Audio Codes-O365
Englewood SBC, 10.130.40.140, SIP Trunk,
Englewood SM-new, 10.130.40.148, Session Manager, Englewood SM100
Harland CM, cm.local, CM, Clive and ENG PE
India - Avaya, ipo.local, SIP Trunk,
Lake Mary O365, 172.26.240.55, SIP Trunk, LM Audio Codes-O365, new, was 172.26.240.55
Lake Mary SBC, 10.134.4.40, SIP Trunk
__________________________________
I'm starting to think the issue is that I need to configure SIP phones to dial 9 for outside line. They do not require 9, so when I go to make an INTL call, I am dialing 011-852-1-830-830 for example. The call rings to our operator.

Suggestions please?

 

[kyle555]

Your SIP phones, if they have their Session Manager profile configured with originating and terminating application sequence to CM, will never trigger off of SM dial patterns.

Session Manager does sets and trunks. If a set is registered with an application sequence to CM, it doesn't matter what the dial patterns say, SM pitches to CM.

That said, hypothetically, you could have a system with 100 SIP phones with an app sequence to a Entity/Entity Link of CM and have no dial patterns at all. If that CM had PRI out to the PSTN, those SIP phones could make calls. And if those SIP phones had DIDs and CM sent that up to Session Manager, SM would make those registered phones ring without any dial patterns either.

Regardless of the routing rules that you have in place, you have a more fundamental layer of PPM where the phone decides what it believes to be a complete enough number to send out and those rules are ultimately defined by what's in dialplan analysis and UDP and AAR/ARS.

If you did a display dialplan analysis, can 10 digit numbers go straight to UDP?

 

[marcyg]

Here's the information on the attachment. Current dial plan, ARS, route pattern, and a simple TraceSM when I dial out on the SIP phone to a number. Feeling pretty dumb here LOL. Thank you for the replies.

 

[ZeroZeroOne]

Kyle, That's for Avaya SIP phones, right? Is that the case for other brands, such as Polycoms?
I hope I don't sound like I'm arguing - I can always learn something new - I've always set up dial patterns for other systems (Skype) and just assumed they were necessary for non-Avaya phones.

But that does remind me that the digit map may need to be manually edited in some SIP phones. Last month we had to log into a SpectraLink phone's web admin and change the digit map to allow both 4-digit and 7-digit extension dialing.

So much learning...

 

[kyle555]

Correct. PPM is for Avaya phones only. Polycoms have an internal dial plan within them too - something like 2xxx|9xxxxxxxxxx|91xxxxxxxxxx|911|911 so it knows when to quit dialing, but it's not automagic like the PBX telling the endpoint what those strings are.

Dial patterns are for trunk calls, not station calls. So in Marcy's trace we see the SIP phone send an INVITE to SM which pumps it to CM by virtue of the application sequence. CM has that get to ARS somehow presumably and CM sends it back to SM. At that point - once CM sends it back to SM - the dial pattern kicks in and decides to use the routing policy to go to her SBC. If the dial pattern kicked in as soon as the SIP phone went off hook, her call would gone from the SIP phone to the SBC directly and circumvented CM thereby bypassing things like COR and FRL.

Regardless, 0 goes to reception because PPM is telling the Avaya phone that there are no other dialable strings starting with 0 longer than 1 digit.

What about a station trace? What if you traceSM and enable PPM too when you press s to start and you go in SMGR-->SM-->System Status-->User Registrations and filter on your 2132 and click 'reload complete'. You'll see a "GetAllEndpointConfiguration" from the phone to SM and a response from SM to the phone. In there, key down, and you'll see all the patterns SM told your phone it should know how to dial.


I also had a thought... how long does it take once you dial 1+NPA+NXX+XXXX for the line to start ringing? If you display/list locations, you might have a proxy selection route pattern defined for any given location - like that of your SIP phone - and that's the path of last resort if nothing else matched. You'll know you're using it when you look at a station trace and you see something like "location 5 route 22 preference 0" instead of "preference 1" like when you pick the route normally thru ARS.

If that's in fact the case - where you're using the proxy selection route pattern and not even using ARS at all - then the answer to "how do I stop letting calls go out without a 9"? is "by removing the proxy selection route pattern from the specific location that phone belongs to in the locations table"

 

[kyle555]

And, if you go in SMGR and click Session Manager - what release are you on? And if you go in user registrations and pick your phone and expand the details and click 'device', what firmware is your phone running?

 

[gwebster]

Whatever you do, test that a phone can dial just 911 and reach emergency services.

 

[marcyg]

Thank you all for your help. It was in fact a digit mapping on the web interface for the Polycom VVX601 phone. I am now able to dial using 9, outbound caller ID works, and INTL calling works.

0T|011xxx.T|9011xxx.T|1[2-9]xxxxxxxxx|91[2-9]xxxxxxxxx|9[2-9]xxxxxxxxx|*xx[2-7]xxx|[2-7]xxx