Solved: Re: Multiple Sip trunks to One Cube and CUCM for different Sip providers for Migration - Page 2

umutyasar · ‎01-30-2020

Hi,

We're planning to move from one sip provider to the other.

We connect the cube to the other sip trunk and configure inbound/outbound dial peers.

The new provider gives us a new number range for testing.

I didn't configure new dial peers for the new sip trunk to cube and also didn't configure a new sip trunk to cucm.

I think it's possible to use existing sip trunk and dial peers.

For testing purposes, I choose a branch phone and assigned a new Directory number to cucm phone with the number the incoming call will have. I'm using the existing sip trunk (RG), CCS and Partition that the branch is using in production.

I configured a Translation pattern for the new number as well with the same CSS and Partition that the branch is using in production.

The problem is I can't get any dial tonne, once I dial the test number. What do you think is the reason of the problem?

Do I need to configure new sip trunk on cucm and new dial peers to cube for cucm for the new sip trunk?

Regards,

U

TONY SMITH · ‎02-06-2020

Let's see the SIP debugs "debug ccsip mess" for an outbound call, and your CUBE configuration.

umutyasar · ‎02-10-2020

Hi Tony,

The issue is, there is a dial tone but no 2-way media for both inbound and outbound calls.

Please see attached log for an outbound call. I don't see any issue related to codec or dmtf.

Can you see any issue?

TONY SMITH · ‎02-11-2020

Let's confirm the devices ...

192.168.28.30 is that the CUBE?
192.168.28.2 CUCM?
192.168.178.153 IP Phone?
202.10.26.33 Service provider?

It looks as if the call connects, so while it's up can you check it's using the dial peers that you expect ..

show call active voice comp
show call active voice | i PeerId

For future debugs, can you post them without the timestamp at the beginning of every line? Ideally just as the came from the gateway. Those timestamps bugger up presentation in some tools.

umutyasar · ‎02-14-2020

Hi Tony,

Thanks for the commands.

I solved the problem.

The cube was configured for media flow-around and I didn't notice this in the first place.

The existing provider was working since the ITSP sip server and voip phones have the routing each other.

But new ITSP sip server and voip phones don't have routing tables for each other.

I changed the config of the cube to media flow-through and both ITSP sip trunks are working now.

Now I need to plan a migration of DIDs from existing to new ITSP. New ITSP will do the porting.

I want to plan for a single number range porting firstly (12345600-99) after that I want to go for a state by state approach, like 01..., 02..., 03... But for this, I need to influence the outgoing calls according to calling (internal) DID.

Is it possible to do that with cucm and cube?

If not do you know any other solution?

TONY SMITH · ‎02-19-2020

@umutyasar wrote:
Now I need to plan a migration of DIDs from existing to new ITSP. New ITSP will do the porting.
I want to plan for a single number range porting firstly (12345600-99) after that I want to go for a state by state approach, like 01..., 02..., 03... But for this, I need to influence the outgoing calls according to calling (internal) DID.
Is it possible to do that with cucm and cube?
If not do you know any other solution?

Ideally you would get the new SIP provide to honour the calling numbers for all your existing numbers, then you can switch all your outbound calls even when the incoming number ranges are not yet ported.

However if that's not possible, I think you could route outbound calls based on calling number as follows ..

Create a second CUCM facing dial peer which will match calls based on calling number, if you use "incoming calling e164-pattern-map" you'll be able to match multiple ranges and add to the list as your numbers port.

Use Class of Restriction or Dial Peer Groups for force calls matched on that dial peer to use the new SIP service

When the migration is all complete you can strip back a lot of this configuration.

Does that make sense? I'd be interested to see if others have any neater ideas.

umutyasar · ‎02-25-2020

Hi Tony,

I got the new SIP provider to allow all existing DIDs to use the calling numbers for all my existing numbers, then I'll influence all outbound calls to go through the new provider.

But in case of a problem, I was thinking to use your second option.

I'm using the dial-peer group for influencing inbound calls coming from one provider, go out to the same provider for both providers. this part is OK.

But "incoming calling e164-pattern-map" didn't work for influencing outbound calls using Internal numbers. I found a solution for this with CUCM as below but since I'm new to CUCM I'm not quite sure how can I configure CUCM for it. If you know CUCM can you please help me?

It's saying "

you need to create different partitions to include on new CSS that need to be assigned to dedicated users.

Than two route patterns for the same destination will exist but within different partitions and one with a prefix attached and the other without any prefix.

Than pots dial peers will strip specified digits if you don't add forward-digits all "

In my case i want to influence just site-1 DIDs to use new ITSP for outbound calls, the rest of the sites will still going to use existing ITSP. i have an existing partition and css assigned for all the DIDs in this site. In my case, i think i need to create a new css with a new partition with the prefix. How is the route pattern connected to the new css?

Regards,

Umut Yasar

TONY SMITH · ‎02-26-2020

"But "incoming calling e164-pattern-map" didn't work for influencing outbound calls using Internal numbers"

How did you determine that this wasn't working?

Doing this in CUCM would be to my mind a little messy, but possible. It's messy because in CUCM there's only one route to the gateway so you need some way of coding the calls so the gateway will then route them to your preferred carrier. For example you could ..

(1) Create a duplicate set of PSTN Route Patterns, in new Partitions. Configure these Route Patterns so they prefix the called number with something unique, for arguments sake let's say "#98". These Route Patterns will point to the same gateway as normal.

(2) On the gateway create dial peer for calls prefixed "#98", pointing to the new carrier and of course stripping off that prefix on outbound.

(3) Back in CUCM create a new CSS containing the new partitions created in (1), and assign this CSS to the phones

Note that this will NOT work if you're using Dial Peer Groups, and they over-ride any dial peer matching by number.

umutyasar · ‎02-26-2020

Hi Tony, I'll look at the CUCM part but I want to answer your question first about e164. I tried it but it didn't work.

In cisco doc, it's saying that "incoming calling" can be used for inbound dial peers but I believe I need to do something on outbound dial-peer to influence outbound calls.

I used "destination calling e164-pattern-map 10" command for it on outbound dial-peer. And 10 was configured as below, I tried without prefix "*00" and it didn't work as well,

voice class e164-pattern-map 10

e164 *000111111..

I have a strip config on outbound dial-peer to strip "*00" I'm not sure why it's there since route-pattern is "#.@", It hasn't got *00 at the beginning and "destination-pattern *00.T" configured on dial-peer. Do you have an idea why it's configured like this way?

I configured as below and it didn't work, I couldn't call outside using number 0111111123, no 2-way voice. I didn't do debug at that time, I'm not sure why it didn't work. I thought it's not possible to influence outbound dial-peer for an internal number calling outside with e164 and passed this option. (I just placed the requirement I mentioned on this post for the below outbound dial-peer, I am not putting all of its configs)

dial-peer voice 10 voip

translation-profile outgoing strip-steeringcode

destination-pattern *00.T

destination calling e164-pattern-map 10

umutyasar · ‎02-26-2020

Hi Tony,

I'll look at the CUCM part but I want to answer your question first about e164. I tried it but it didn't work.

In cisco doc, it's saying that "incoming calling" can be used for inbound dial peers but I believe I need to do something on outbound dial-peer to influence outbound calls.

I used "destination calling e164-pattern-map 10" command for it on outbound dial-peer. And 10 was configured as below, I tried without prefix "*00" and it didn't work as well,

voice class e164-pattern-map 10

e164 *000111111..

I have a strip config on outbound dial-peer to strip "*00" I'm not sure why it's there since route-pattern is "#.@", It hasn't got *00 at the beginning and "destination-pattern *00.T" configured on dial-peer. Do you have an idea why it's configured like this way?

I configured as below and it didn't work, I couldn't call outside using number 0111111123, no 2-way voice. I didn't do debug at that time, I'm not sure why it didn't work. I thought it's not possible to influence outbound dial-peer for an internal number calling outside with e164 and passed this option. I'm not putting all of the dial-peer config it's just showing the things that I mentioned here.

dial-peer voice 10 voip

translation-profile outgoing strip-steeringcode

destination-pattern *00.T

destination calling e164-pattern-map 10

TONY SMITH · ‎02-26-2020

@umutyasar wrote:
Hi Tony,
I'll look at the CUCM part but I want to answer your question first about e164. I tried it but it didn't work.
In cisco doc, it's saying that "incoming calling" can be used for inbound dial peers but I believe I need to do something on outbound dial-peer to influence outbound calls.

First things first. To do this on the gateway the way I was suggesting you first need your calls to match different inbound dial peers on the CUCM side. Once that's working you can add configuration so that these different dial peers route the outbound calls as you wish. There are different ways of doing this, you could have different Dial Peer Groups, or different COR settings, or different Translation Profiles to tweak the called number.

Did you get the inbound matching working?

Easiest way to check this would be to add your new dial peer for the migrated numbers, but don't do anything to change outbound. Place a call from one of the numbers that should match, and check dial peers used in the call.

If it's not matching then you need to look at which dial peer it's hitting, and why that match is being preferred.

gachilleas · ‎02-27-2020

Please check the following figures they will help you understand the matching criteria and priorities.
[cid:image002.jpg@01D5ED63.389C4BD0]

[cid:image003.jpg@01D5ED63.389C4BD0]
Regards,
Achilleas Grigoriadis

umutyasar · ‎04-08-2020

Hi Tony,

It has been sometime after you help me resolve this problem. I am accepting your answer as the solution.

As I said I got the new SIP to provide to honor the calling numbers for all existing numbers, and It's become very easy for migration. Inbound calls use old SP and Outbound calls I directed to the new SP using Preference 1 option in the dial-peer config with it. With this, I do not need to do any config change after porting even if we move numbers step by step.

Also the other option you mentioned using Dial Peer Groups for outbound calls; matching calling number using "incoming calling e164-pattern-map" from a second CUCM facing dial-peer and using dial-peer groups to direct to new SP should work, but I did not use it as the first option solved my problem.

Regards,

Umut Yasar

Mohammed al Baqari · ‎01-31-2020

Look at cucm sip call tree to see if the call exits.

gachilleas · ‎02-08-2020

Check the use of Tenants in CUBE in order to separate traffic to/from each provider.
Then in dial peers you can specify the tenant you want.
e.g voice class tenant 1
etc

Separate dial peers for each traffic flow is better as you can manipulate data according to your needs.