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
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
Thank you so much for your help. I will try this as soon as I get home.
turuncu
Quote from: RonR on March 04, 2012, 01:56:06 PM
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
Quote from: turuncu on March 04, 2012, 04:29:47 PM
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).
Quote from: turuncu on March 04, 2012, 04:29:47 PM
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
Quote from: turuncu on March 04, 2012, 04:29:47 PM
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.
Quote from: RonR on March 04, 2012, 05:02:13 PM
Quote from: turuncu on March 04, 2012, 04:29:47 PM
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).
Quote from: turuncu on March 04, 2012, 04:29:47 PM
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
Quote from: turuncu on March 04, 2012, 04:29:47 PM
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?
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.
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.
Quote from: RonR on March 04, 2012, 05:02:13 PM
Quote from: turuncu on March 04, 2012, 04:29:47 PM
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).
Quote from: turuncu on March 04, 2012, 04:29:47 PM
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
Quote from: turuncu on March 04, 2012, 04:29:47 PM
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
Quote from: turuncu on March 11, 2012, 08:04:49 AM]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?
Quote from: Stewart on March 11, 2012, 08:51:12 AM
Quote from: turuncu on March 11, 2012, 08:04:49 AM]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
Quote from: RonR on March 04, 2012, 01:56:06 PM
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