Single-Stage Dialing Through Any OBi Trunk Using SIP
RonR:
Quote from: fatbrain on February 15, 2012, 06:42:58 am
What is the sequence of translations that occur from the call flow: Polycom->SP2->SP1->GV
I'm assuming you have your Polycom dialplan set up to pass whatever you dial unchanged.
The only translations that would occur in the OBi are those provided by the DigitMap for the target trunk. In the case you cited, SP1, it would typically be:
Service Providers -> ITSP Profile A -> General -> DigitMap
RonR:
fatbrain,
If you'd like to also receive incoming calls on the Polycom:
On any trunk that you wish to have the Polycom ring in addition to the PHONE Port:
InboundCallRoute : {sp2(192.168.1.145:5062),ph}
Use your actual Polycom IP address and port number in place of '192.168.1.145:5062' above.
Stewart:
Quote from: RonR on February 15, 2012, 10:43:28 am
If you'd like to also receive incoming calls on the Polycom:
1. Set up a Voice Gateway to point to the Polycom ...
Unfortunately, that approach cannot deliver caller ID to the Polycom, because of a limitation in the OBi.
I have a similar setup with two Polycom phones, and tried to work around the problem by using an InboundCallRoute of e.g. {SP2(polycom@192.168.1.145:5062),ph}, combined with setting X_SpoofCallerID. Unfortunately, there was a subtle problem (I don't remember the details); I gave up and ended up putting a free PBXes account in the path (OBi registers as a sub-PBX, Polycoms register as extensions, PBXes routes the OBi "trunk" to the Polycoms via a ring group).
RonR:
The following is a better solution for ringing a SIP client that allows passing CallerID:
On any trunk that you wish to have the SIP client ring in addition to the PHONE Port:
InboundCallRoute : {spx(userid@192.168.1.145:5062),ph}
Service Providers -> ITSP Profile x -> SIP : X_SpoofCallerID : (checked)
Diana:
I had setup my OBi110 in the past for "Single-Stage Dialing" based on the older method ( http://www.obitalk.com/forum/index.php?topic=1103.0 ) and would like to use the method in this thread which appears to be more general. However, I have a few questions base on my current setup, which is:
SIP Line: SP2 - setup with anveo; no DID
SIP Port: 5060 (set for 5060 even though Anveo uses 5010)
SP1: GV
SP1 Port: 5064
Voice Gateway 3 (vg3): CallWithUs
Primary Line: tg1
TrunkList: sp1,sp2,vg3
3 OBion Apps running on 3 Android Phones (290xxxxxA;290xxxxxB;290xxxxxC)
1 SIP App (CSimple) running on 1 Android Phone (username: 123456)
1 SIP Phone external to my LAN (Work) [username: 123789]
1 OBi110 on my LAN (192.168.1.211)
I have a free URL address with DYNDNS: xxxxxx.dyndns.org
Questions:
Quote
User Settings -> User Defined DigitMaps -> User Defined Digit MapX
Label : p2p
DigitMap : (<n*:>(@@.)<:@192.168.1.140:sip_port>|<n*:>(@@.)<:@hostname:sip_port>)
(1) If I'm only using one OBi (OBi No: 200OBiNo), should I set n = 1?
(2) I'm not sure what info should be substitute here: "@hostname:sip_port>"; should it be @xxxxxx.dyndns.org:5064?
DigitMap : (<1*:>(@@.)<:@192.168.1.211:5060>|<1*:>(@@.)<:@hostname:5060>)?
(3) Can I still use the OBiON App and a SIP Client on the three Android Devices?
The other changes that were made are:
Quote
Voice Services -> SP2 Service -> X_InboundCallRoute (SPx must be configured for SIP):
{(Mtsc)>(Mp2p):sp2},{(Mtsc)>**0:aa},{(Mtsc)>***:aa2},{(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)>(Mtg1):tg1},{(Mtsc)>0:ph},{(Mtsc):},{ph}
User Settings -> User Defined DigitMaps -> User Defined Digit Map4
Label : tsc
DigitMap : (200OBiNo|290xxxxxA|290xxxxxB|290xxxxxC|123456|123789)
This is a list of trusted OBi and SIP client userid's that are allowed to use single-stage dialing on this OBi. If no other OBi's or SIP clients are allowed to use single-stage dialing on this OBi, the DigitMap may be left empty: ()
Quote
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:>@@.):sp2} in the PHONE Port and Auto Attendant OutboundCallRoute's:
Physical Interfaces -> PHONE Port -> OutboundCallRoute:
{(<P2P:>@@.):sp2},{(<#:>|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:>@@.):sp2},{0:ph},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},
{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}
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 :5064 appended for SP2). If SIP clients are located outside the OBi's LAN, ports 5060, 5010 and 5064 forwarded to the OBi in the router!!
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