help with new setup with multiple sips

[turuncu]

I am trying to move from spa3000 to obi110.  From what I read it should be able to everything that spa can do.

I have set up gv with no problem and can receive calls from it with no problem.  I have setup with callcentric for the 911 on the 2nd line.  I would like to set up a 3rd sip with a Betamax account.  Betamax can place calls without registration.

*I would like my local calls with 10 digits or 1xxxxxxxxxx to go through gv (SP1)
*911 thorugh callcentric (Sp2) 
*Anything that is 011 to go throught betamax (SP3)

I know I can add a **3 but I was not able to do it and I could not do the dial plan.  I have been reading the dial plan stuff for the last 3 days and I am getting more and more confused. 

I would really appreciated any help.

Turuncu

[RonR]

Physical Iterfaces -> PHONE Port -> DigitMap:

([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|**1(Msp1)|**2(Msp2)|**3(Mvg3)|**8(Mli)|**9(Mpp)|(Mpli))


Physical Iterfaces -> PHONE Port -> OutboundCallRoute:

{([1-9]x?*(Mpli)):pp},{(<#:>):li},{911:sp2},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},
{(<**2:>(Msp2)):sp2},{(<**3:>(Mvg3)):vg3},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}


Service Providers -> ITSP Profile A -> General -> DigitMap:

(<1>[2-9]xxxxxxxxx|1xxxxxxxxxx|<**3>011xx.)


Gateways and Trunk Groups -> Voice Gateway3
Name : Service_Provider_Name
AccessNumber : SP2(SIP_Server_Proxy)
DigitMap : (<011:00>xx.|00xx.)
AuthUserID : SIP_Username
AuthPassword : SIP_Password


911                 ->  SP2
10 or 11 digits  ->  SP1
011+               ->  VG3

[turuncu]

Thank you so much for your help.  I will try this as soon as I get home.

turuncu

[turuncu]

Physical Iterfaces -> PHONE Port -> DigitMap:

([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|**1(Msp1)|**2(Msp2)|**3(Mvg3)|**8(Mli)|**9(Mpp)|(Mpli))


Physical Iterfaces -> PHONE Port -> OutboundCallRoute:

{([1-9]x?*(Mpli)):pp},{(<#:>):li},{911:sp2},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},
{(<**2:>(Msp2)):sp2},{(<**3:>(Mvg3)):vg3},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}


Service Providers -> ITSP Profile A -> General -> DigitMap:

(<1>[2-9]xxxxxxxxx|1xxxxxxxxxx|<**3>011xx.)


Gateways and Trunk Groups -> Voice Gateway3
Name : Service_Provider_Name
AccessNumber : SP2(SIP_Server_Proxy)
DigitMap : (011xx.)
AuthUserID : SIP_Username
AuthPassword : SIP_Password


911                 ->  SP2
10 or 11 digits  ->  SP1
011+               ->  VG3

I have tried this and when I dield 011+ the call went through the GV (i saw the gv number on the recipientens cid)

Is there a log that i can see that shows where each call is rauted to as I dial the number or after the call?

Also I just remembered that the VG that I using needs to have 00 as oppose to 011 is there a way to replace 011 that the user dials and put 00 when we pass the call to VG3?

Thanks

[RonR]

I have tried this and when I dield 011+ the call went through the GV (i saw the gv number on the recipientens cid)

I assumed you have:

Physical Interfaces -> PHONE Port -> PrimaryLine: SP1 Service

If so, dialing 011+ should result in **3011+ being sent to the OutboundCallRoute where it would go to VG3 with the **3 removed in the process (and 011 changed to 00 with the revised VG3 DigitMap).

Is there a log that i can see that shows where each call is rauted to as I dial the number or after the call?

While the call is in progress:

Status -> Call Status

After the call has completed:

Status -> Call History

Also I just remembered that the VG that I using needs to have 00 as oppose to 011 is there a way to replace 011 that the user dials and put 00 when we pass the call to VG3?

See the revised VG3 -> DigitMap in reply #1.

[DocM]

I have tried this and when I dield 011+ the call went through the GV (i saw the gv number on the recipientens cid)

I assumed you have:

Physical Interfaces -> PHONE Port -> PrimaryLine: SP1 Service

If so, dialing 011+ should result in **3011+ being sent to the OutboundCallRoute where it would go to VG3 with the **3 removed in the process (and 011 changed to 00 with the revised VG3 DigitMap).

Is there a log that i can see that shows where each call is rauted to as I dial the number or after the call?

While the call is in progress:

Status -> Call Status

After the call has completed:

Status -> Call History

Also I just remembered that the VG that I using needs to have 00 as oppose to 011 is there a way to replace 011 that the user dials and put 00 when we pass the call to VG3?

See the revised VG3 -> DigitMap in reply #1.

With the current config, wouldn't you have to dial **3011 (not 011) to correctly connect the call?

[RonR]

Give a PrimaryLine of SP1 Service and the following DigitMap:

Service Providers -> ITSP Profile A -> General -> DigitMap : (<1>[2-9]xxxxxxxxx|1xxxxxxxxxx|<**3>011xx.)

011+ will be matched by the '<**3>011xx.' rule, adding **3 to it.

The '{(<**3:>(Mvg3)):vg3}' OutboundCallRoute rule will be matched, replacing 011 with 00 in the process by the VG3 DigitMap rule '<011:00>xx.', with **3 being stripped before sending the remainder to VG3.

[DocM]

So, when 011+phone number dialed > Phone Port > Primary Line > SP1 and Corresponding ITSP Profile > **3 added > then back to Phone Port > OutboundCallRoute > **3 matched and stripped > VG3 > replaces 011 with 00 > resulting number dialed. Hope I got that right.

Wow, I never knew it functioned like that. This will be very valuable for my own project.

[turuncu]

I have tried this and when I dield 011+ the call went through the GV (i saw the gv number on the recipientens cid)

I assumed you have:

Physical Interfaces -> PHONE Port -> PrimaryLine: SP1 Service

If so, dialing 011+ should result in **3011+ being sent to the OutboundCallRoute where it would go to VG3 with the **3 removed in the process (and 011 changed to 00 with the revised VG3 DigitMap).

Is there a log that i can see that shows where each call is rauted to as I dial the number or after the call?

While the call is in progress:

Status -> Call Status

After the call has completed:

Status -> Call History

Also I just remembered that the VG that I using needs to have 00 as oppose to 011 is there a way to replace 011 that the user dials and put 00 when we pass the call to VG3?

See the revised VG3 -> DigitMap in reply #1.

Thanks I can now see that the calls are being routed correctly.  Betamax is giving me headache but that is another matter.  Thanks for your help

[Stewart]

]Betamax is giving me headache but that is another matter.

Possible alternatives to Betamax (usually somewhat higher rates, but better quality and reliability): Voxbeam, CallWithUs, Future-Nine.  Where are you calling?  Landline or mobile?  If mobile, which carrier?

[turuncu]

]Betamax is giving me headache but that is another matter.

Possible alternatives to Betamax (usually somewhat higher rates, but better quality and reliability): Voxbeam, CallWithUs, Future-Nine.  Where are you calling?  Landline or mobile?  If mobile, which carrier?

Sorry just saw your message.  I call Turkey and Azerbaijan.  and betamax usually have some free stuff for Turkey.  I used to use callwithus but the customer service was as helpful as betamax and I had to pay for every minute. 

I call a combination mobile and landline.

Thanks for your help

[turuncu]

Physical Iterfaces -> PHONE Port -> DigitMap:

([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|**1(Msp1)|**2(Msp2)|**3(Mvg3)|**8(Mli)|**9(Mpp)|(Mpli))


Physical Iterfaces -> PHONE Port -> OutboundCallRoute:

{([1-9]x?*(Mpli)):pp},{(<#:>):li},{911:sp2},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},
{(<**2:>(Msp2)):sp2},{(<**3:>(Mvg3)):vg3},{(<**4:>(Mvg4)):vg4},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}


Service Providers -> ITSP Profile A -> General -> DigitMap:

(<1>[2-9]xxxxxxxxx|1xxxxxxxxxx||<**4>011905.|<**3>011xx.)


Gateways and Trunk Groups -> Voice Gateway3
Name : Service_Provider_Name
AccessNumber : SP2(SIP_Server_Proxy)
DigitMap : (<011:00>xx.|00xx.)
AuthUserID : SIP_Username
AuthPassword : SIP_Password

Gateways and Trunk Groups -> Voice Gateway4
Name : Service_Provider_Name
AccessNumber : SP3(SIP_Server_Proxy)
DigitMap : (<011:00>xx.|00xx.)
AuthUserID : SIP_Username
AuthPassword : SIP_Password


911                 ->  SP2
10 or 11 digits  ->  SP1
011+               ->  VG3
011905+            ->  VG4

If I wanted to add one more sip provider and route number that start with 011905+ would I do it like above.  What I tried to do was add a new sip  assign **4.  And for the digimap I have put it infornt of **3.  Would this work

Thanks