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Alittlegreen
join:2008-07-24
Montreal, QC

Alittlegreen

Member

Ho do we use Anveo & VoIP.ms in an automated fashion via Obi302

While I was reading Mc Lovin's post -Awesome name btw- it got me thinking. In particular Ozark's posting »Re: [Anveo] Hard time configuring Obi to work with Anveo :-( of using trunkgroups.

My present setup as I've been trying the different ITSPs is slightly different in that, sp1 is Anveo, and sp2 is VoIP.ms

Is it possible to set up the Obi so that :
1-Anveo remains the primary means of placing and receiving calls, and that in the event that their network doesn't work, that it failovers the inbound call to a voip.ms account@sip.voip.ms (or the like)?
2-If when placing an outbound call the Anveo network is down/or simply not registered with the ATA, that the Obi automatically places the outbound call via voip.ms(sp2)?
2.2-If the above two are possible, is it possible to cancel VoIP.ms' DID number and simply have the calls go to and from your sip address -with the exception of course of 911.
3-Calls to certain numbers automatically bypass Anveo (sp1) and go through voip.ms(sp2)?
Mango
Use DMZ and you get a kick in the dick.
Premium Member
join:2008-12-25
www.toao.net

Mango

Premium Member

1) Unfortunately no. If Anveo's entire network is offline or otherwise down, they will not be able to route calls to VoIP.ms or anyone else. However, this is unlikely as they do have automatic geographic redundancy.

2) Yes. Use the Trunk Groups feature.

2.2) Yes, you do not require a DID to use VoIP.ms.

3) Yes, exclude those numbers from your Anveo DigitMap.

m.

Alittlegreen
join:2008-07-24
Montreal, QC

1 edit

Alittlegreen

Member

Are there any tutorials, or crash courses anywhere for setting up the trunk groups?
And how can we add exclusion to a dialing plan? My current one for Anveo is below, bu just not quite sure how to modify it.
(933|*xx|1xxxxxxxxxx|[2-9]xxxxxxxxx|011xx.|xx.)
VoIP2Go
join:2013-12-14

VoIP2Go

Member

I have this thread stored in my favorites...

»[Equipment] Using Obi for partial automatic failover

Most of the Obi experts on here offer their input
Mango
Use DMZ and you get a kick in the dick.
Premium Member
join:2008-12-25
www.toao.net

Mango to Alittlegreen

Premium Member

to Alittlegreen
Trunk groups are quite easy: »toao.net/510

My best guess at a barring rule (test it to see if I'm right).

(933|*xx|!15552345678|1xxxxxxxxxx|[2-9]xxxxxxxxx|011xx.|xx.)

Alittlegreen
join:2008-07-24
Montreal, QC

Alittlegreen

Member


What if we take a match to it?
Hi there,
Ok, So I had some time to test it all out last night. Here's what going on so far:

In a perfect world:
1-In the event that a call cannot be placed via sp1, it should go through sp2. And if neither of the the two are avail, then on to sp3.

2-In addition to this, all 311, 511, 811, 4443, 4747, and *225 calls need to automatically be routed to sp2 or sp3.

Current Setup:

Sp1 - Anveo
Sp2 - VoIP.ms yyz server
Sp3 - VoIP.ms yul server

----------
Service Providers->ITSP Profile A(Anveo)->Digit Map->
(9(11|33)|*xx|1xxxxxxxxxx|[2-9]xxxxxxxxx|011xx.|xx.)

Service Providers->ITSP Profile B(VoIP.ms yyz server)->Digit Map->
(*75xx|*xx|1xxxxxxxxxx|[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*]@@.)

Service Providers->ITSP Profile C(VoIP.ms yul server)->Digit Map->
(*75xx|*xx|1xxxxxxxxxx|[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*]@@.)

----------
Voice Services->Gateways & TrunkGrps->Trunk Group1->TrunkList->
sp1,sp2,sp3

Voice Services->Gateways & TrunkGrps->Trunk Group1->DigitMap->
((Msp1)|(Msp2)|(Msp3))

----------

Physical Interfaces->Phone 1->Phone Port->DigitMap->
([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|##|**70(Mli)|**8(Mbt)|**81(Mbt)|**82(Mbt2)|**1(Msp1)|**2(Msp2)|**3(Msp3)|**4(Msp4)|**9(Mpp)|(Mpli))

Physical Interfaces->Phone 1->Phone Port->OutboundCallRoute->
{911:sp1},{933:sp1},{([1-9]x?*(Mpli)):pp},{():li},{():ph2},{((Mli)):li},{((Mbt2)):bt2},{((Mbt)):bt},{((Mbt)):bt},{**0:aa},{***:aa2},{((Msp1)):sp1},{((Msp2)):sp2},{((Msp3)):sp3},{((Msp4)):sp4},{((Mpp)):pp},{(Mpli):pli}

Physical Interfaces->Phone 2->Phone Port->DigitMap->
([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|##|**70(Mli)|**8(Mbt)|**81(Mbt)|**82(Mbt2)|**1(Msp1)|**2(Msp2)|**3(Msp3)|**4(Msp4)|**9(Mpp)|(Mpli))

Physical Interfaces->Phone 2->Phone Port->OutboundCallRoute->
{911:sp1},{933:sp1},{([1-9]x?*(Mpli)):pp},{():li},{():ph1},{((Mli)):li},{((Mbt2)):bt2},{((Mbt)):bt},{((Mbt)):bt},{**0:aa},{***:aa2},{((Msp1)):sp1},{((Msp2)):sp2},{((Msp3)):sp3},{((Msp4)):sp4},{((Mpp)):pp},{(Mpli):pli}

------------------------------------------

Last night I tried testing by changing the following:

1- Service Providers->ITSP Profile A(Anveo)->Digit Map->
(9(11|33)|*xx|!311|!511|!811|!4443|!4747|1xxxxxxxxxx|[2-9]xxxxxxxxx|011xx.|xx.)

2- Voice Services->Gateways & TrunkGrps->Trunk Group1->TrunkList->
sp1,sp2
DigitMap: ((Msp1)|(Msp2))

3- Physical Interfaces->PHONE Port->OutboundCallRoute->
{(Mtg1):tg1},{(Msp2):sp2}

But whenever I dialed 811 I simply got a fast busy.
then I tried entering the wrong sip account number for Anveo so that it wouldn't register, and when I dialed 811 again, but no dice.

Does anyone here have any insight as to what I'm doing wrong? I dunno about you guys, but to me, it feels like looking for the proverbial needle
OzarkEdge
join:2014-02-23
USA

OzarkEdge

Member

Given VoIP.ms servers have the same profile requirement, you could omit ITSP Profile C and simply reference ITSP Profile B where required.

BTW, Obihai's recommended firmware parameter format...

Group Name - Sub Group Name::ParameterName = Parameter Value

... avoids using those pesky html tags that can make information go missing in these forum posts.

OE
RonR
join:2003-10-10
Ash Flat, AR

RonR to Alittlegreen

Member

to Alittlegreen
I took a quick look at your DigitMaps and didn't see an obvious problem (doesn't mean I didn't miss it). Then I wondered if there's a problem with Trunk Groups not rolling over on barred numbers, but some testing on an OBi202 revealed that it appears to work properly.

Any chance you mistyped something in your testing last night? I think it's worth another try.
RonR

RonR to OzarkEdge

Member

to OzarkEdge
said by OzarkEdge:

Given VoIP.ms servers have the same profile requirement, you could omit ITSP Profile C and simply reference ITSP Profile B where required.

If SP2 (VoIP.ms yyz server) and SP3 (VoIP.ms yul server) use different ProxyServer's, then separate ITSP profiles are needed.

Alittlegreen
join:2008-07-24
Montreal, QC

Alittlegreen to RonR

Member

to RonR
Hey RonR,
Just to double check, in your testing with the 202 you had:

Voice Services->Gateways & TrunkGrps->Trunk Group1->TrunkList->
sp1,sp2,sp3

Voice Services->Gateways & TrunkGrps->Trunk Group1->DigitMap->
((Msp1)|(Msp2)|(Msp3))

Physical Interfaces->Phone 1->PHONE Port->OutboundCallRoute->
{911:sp1},{933:sp1},{(Mtg1):tg1},{([1-9]x?*(Mpli)):pp},{():li},{():ph2},{((Mli)):li},{((Mbt2)):bt2},{((Mbt)):bt},{((Mbt)):bt},{**0:aa},{***:aa2},{((Msp1)):sp1},{((Msp2)):sp2},{((Msp3)):sp3},{((Msp4)):sp4},{((Mpp)):pp},{(Mpli):pli}

Physical Interfaces->Phone 2->PHONE Port->OutboundCallRoute->
{911:sp1},{933:sp1},{(Mtg1):tg1},{([1-9]x?*(Mpli)):pp},{():li},{():ph1},{((Mli)):li},{((Mbt2)):bt2},{((Mbt)):bt},{((Mbt)):bt},{**0:aa},{***:aa2},{((Msp1)):sp1},{((Msp2)):sp2},{((Msp3)):sp3},{((Msp4)):sp4},{((Mpp)):pp},{(Mpli):pli}

Thanks again all for all your brain power.
twinclouds
join:2010-06-12
San Diego, CA

twinclouds

Member

I tried to use the trunk group feature for a similar purpose. I put a wrong registration for one of the entry of the trunk group thought it would fall over to the next entry. However, it didn't. As RonR said, the Obi trunk group feature has a limited functionality and may not always work. Eventually, I gave up. If anyone has good experience, please let me know what is your configuration and how it works.
RonR
join:2003-10-10
Ash Flat, AR

RonR to Alittlegreen

Member

to Alittlegreen
said by Alittlegreen:

Just to double check, in your testing with the 202 you had:

1. You need to use code directives around DigitMaps to keep them from getting mangled by the forum software. What you posted didn't come across correctly.

2. No, I don't use anything close to the default values in my OBi. I did test with a Trunk Group as the PrimaryLine. The Trunk Group had two members (SP1,SP2). SP1's DigiMap had a barred number. When dialing anything other than the barred number, it went out SP1. When dialing the barred number, it went out SP2.

I suggest you try a simplified test case as I did. Get something simple working first and build from there.
OzarkEdge
join:2014-02-23
USA

OzarkEdge to RonR

Member

to RonR
said by RonR:

If SP2 (VoIP.ms yyz server) and SP3 (VoIP.ms yul server) use different ProxyServer's, then separate ITSP profiles are needed.

Yes, you are right, and likely so... one is his pop, the second/failover is not his pop. My bad.

OE

Alittlegreen
join:2008-07-24
Montreal, QC

Alittlegreen

Member

Fiiiiinnnallly figure out why the calls that were barred on sp1 weren't rolling over to sp2.
Physical interfaces: Phone 1: Phone Port: Primary Line: Trunk Group1

The default had the above as simply SP1.

What a sigh of releif...
now if i could only get them to roll over to SP3! I think for the time being i'll just live with 1 fall back sp