Single-Stage Dialing Through Any OBi Trunk Using SIP
RonR:
This article illustrates how to configure any number of OBi100/OBi110/OBi202 devices plus any number of SIP clients for single-stage dialing capability. Any OBi or SIP client can initiate a call through any trunk of an OBi using SIP exclusively. The OBiTALK Service and OBiON Apps are not used. SIP clients may be softphone's, IP phones, ATA's, etc.
Dialing from an OBi or its SIP clients uses the following format:
18005551212 -> Local PrimaryLine
**1 18005551212 -> Local SP1 Service
**2 18005551212 -> Local SP2 Service
**8 18005551212 -> Local LINE Port
**9 200123456 -> Local OBiTALK Service
n*18005551212 -> OBi n PrimaryLine
n **1 18005551212 -> OBi n SP1 Service
n **2 18005551212 -> OBi n SP2 Service
n **8 18005551212 -> OBi n LINE Port
n **9 200123456 -> OBi n OBiTALK Service
Voice Services -> SPx Service -> X_InboundCallRoute (SPx must be configured for SIP) (OBi100/OBi110):
{(Mtsc)>(<*1:>(Msp1)),(Mtsc)>(<**1:>(Msp1)):sp1},{(Mtsc)>(<*2:>(Msp2)),(Mtsc)>(<**2:>(Msp2)):sp2},
{(Mtsc)>(<*8:>(Mli)),(Mtsc)>(<**8:>(Mli)):li},{(Mtsc)>(<*9:>(Mpp)),(Mtsc)>(<**9:>(Mpp)):pp},
{(Mtsc)>**0:aa},{(Mtsc)>***:aa2},{(Mtsc)>(Mp2p):spx},{(Mtsc)>(Mpli):pli},{(Mtsc)>0:ph},{(Mtsc):},{ph}
Voice Services -> SPx Service -> X_InboundCallRoute (SPx must be configured for SIP) (OBi202):
{(Mtsc)>(<*1:>(Msp1)),(Mtsc)>(<**1:>(Msp1)):sp1},{(Mtsc)>(<*2:>(Msp2)),(Mtsc)>(<**2:>(Msp2)):sp2},
{(Mtsc)>(<*3:>(Msp3)),(Mtsc)>(<**3:>(Msp3)):sp3},{(Mtsc)>(<*4:>(Msp4)),(Mtsc)>(<**4:>(Msp4)):sp4},
{(Mtsc)>(<*8:>(Mli)),(Mtsc)>(<**8:>(Mli)):li},{(Mtsc)>(<*9:>(Mpp)),(Mtsc)>(<**9:>(Mpp)):pp},
{(Mtsc)>**0:aa},{(Mtsc)>***:aa2},{(Mtsc)>(Mp2p):spx},{(Mtsc)>(Mpli):pli},{(Mtsc)>1:ph1},{(Mtsc)>2:ph2},
{(Mtsc):},{phx}
Note: Replace spx above (1 place) with the OBi SIP service (sp1,sp2,sp3,sp4). Replace pli above (2 places) with the trunk name (sp1,sp2,sp3,sp4,li,pp,tg1) to be used as the PrimaryLine. Replace phx above (1 place) with the desired PHONE Port (ph1,ph2).
User Settings -> User Defined DigitMaps -> User Defined Digit MapX
Label : tsc
DigitMap : (userid1|userid2)
This is a list of trusted OBi and SIP client userid's that are allowed to use single-stage dialing on this OBi. Any reserved characters (m, M, s, S, x, X) in userid's must be surrounded by single quotes. If no other OBi's or SIP clients are allowed to use single-stage dialing on this OBi, the DigitMap may be left empty: ()
User Settings -> User Defined DigitMaps -> User Defined Digit MapX
Label : p2p
DigitMap : (<n*:>(@@.)<:@192.168.1.140:sip_port>|<n*:>(@@.)<:@hostname:sip_port>)
where n is a number to be associated with a destination OBi's IP address or hostname.
Note: Replace sip_port above with 5061 if the destination OBi has SP2 configured for SIP (:sip_port may be omitted for 5060/SP1).
This is a list of destination OBi's through which calls from this OBi or its SIP clients can be initited. If you do not wish to initiate calls through other OBi's from this OBi or its SIP clients, the DigitMap may be left empty: ()
Change [1-9]x?*@@. to <P2P>(Mp2p) in the PHONE Port and Auto Attendant DigitMap's:
Physical Interfaces -> PHONE Port -> DigitMap:
(<P2P>(Mp2p)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|**1(Msp1)|**2(Msp2)|**8(Mli)|**9(Mpp)|(Mpli))
Voice Services -> Auto Attendant -> DigitMaP:
(<P2P>(Mp2p)|[1-9]|[1-9][0-9]|<00:$1>|0|**1(Msp1)|**2(Msp2)|**8(Mli)|**9(Mpp)|(Mpli))
Change {([1-9]x?*@@.):pp} to {(<P2P:>@@.):spx} in the PHONE Port and Auto Attendant OutboundCallRoute's:
Physical Interfaces -> PHONE Port -> OutboundCallRoute:
{(<P2P:>@@.):spx},{(<#:>|911):li},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},
{(<**2:>(Msp2)):sp2},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}
Voice Services -> Auto Attendant -> OutboundCallRoute:
{(<P2P:>@@.):spx},{0:ph},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},
{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}
Note: Replace spx above with the OBi SIP service (sp1,sp2,sp3,sp4).
SIP clients should be configured to make calls without SIP registration. The SIP proxy should be the IP address or hostname of the OBi (with :5061 appended for SP2). If SIP clients are located outside the OBi's LAN, ports 5060 and 5061 will need to be forwarded to the OBi in the router.
If you don't have a SIP provider and SP2 is unused, the following will enable SP2 for SIP:
Service Providers -> ITSP Profile B -> SIP -> ProxyServer : 127.0.0.1
Voice Services -> SP2 Service -> AuthUserName : (any userid)
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP2 Service -> X_ServProvProfile : B
freewilly:
Good to know, I will try it later.
This will make land line(or Ooma) work with a sip phone without a pbx.
Obi110 is configured as a pbx which can servers a single sip client.
fatbrain:
no luck with this configuration... no matter what number I dial on my sip phone it ends up in {ph}
I have GV in SP1, a Polycom IP 550 in SP2 and an active Line port. I'd like to be able to single-stage dial from my Polycom that is within the same network and be able to 2-stage dial with authentication and/or caller ID on my LINE port.
Thanks guys!
RonR:
The fact that you're reaching the PHONE Port means you're almost there.
The problem is likely that you don't have the username from your IP Phone in the tsc User Defined DigitMap. Look at the OBi Call History and make sure you're using the correct username.
Also, if there are any reserved characters (m, M, s, S, x, X) in the username, they have to be surrounded by single quotes. For example, if the username is xlite, it would have to be entered in the tsc User Defined DigitMap as : 'x'lite
fatbrain:
Outstanding!!. Now it works. Thanks so much RonR!!!
What is the sequence of translations that occur from the call flow: Polycom->SP2->SP1->GV
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