Can a pair of obi110's be used to create a bridge without Internet access?
I own a large farm which has a private Ethernet network interconnecting the various buildings, but no Internet.
I would just like to take the PSTN and export it elsewhere on the network.
Building 1
PSTN Line <-> Obi110#1 <->EthernetNetwork
Building 2
EthernetNetwork<->Obi110#2<->POTS Phone
Can this be done with no Internet access?
Incoming calls from the PSTN Line to the phone connected to OBi #2 isn't a problem. Outgoing calls from the phone connected to OBi #2 to the PSTN Line isn't as easy. The only solution I came up with is to have OBi #2 hot dial the Auto Attendant in OBi #1 in order to place an outgoing call.
After revisiting this, I just discovered that Obihai apparently decided to quietly support SIP/PSTN, SIP/SIP, SIP/GV, GV/PSTN, etc. gateway calling after all. I had previously been told this would not be supported in the current product. This simplifies the problem and makes the whole process totally transparent.
Don't register the OBi's on the OBiTALK Web Portal.
Starting from Factory Defaults...
1. Disable Auto Provisioning in each OBi:
System Management -> Auto Provisioning -> ITSP Provisioning -> Method : Disabled
System Management -> Auto Provisioning -> OBiTALK Provisioning -> Method : Disabled
2. Enable SIP in each OBi (SP2/ITSP Profile B):
Service Providers -> ITSP Profile B -> SIP -> ProxyServer : 127.0.0.1
Voice Services -> SP2 Service -> AuthUserName : (put anything here)
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP2 Service -> X_ServProvProfile : B
3. Set Voice Gateway 1 in each OBi to point to the other OBi:
Voice Services -> Gateways and Trunk Groups -> Voice Gateway1
Name : OBi
AccessNumber : SP2(w.x.y.z:5061)
where w.x.y.z is the IP address of the other OBi.
4. Set the LINE Port in OBi #1 to bridge incoming calls to OBi #2:
Physical Interfaces -> LINE Port -> InboundCallRoute : {vg1,ph}
Remove ,ph if you don't want the phone connected to OBi #1 to ring on incoming calls.
5. Set the SP2 Service in OBi #1 to bridge incoming calls to the LINE Port:
Voice Services -> SP2 Service -> X_InboundCallRoute : {>(Mli):li}
6. Set the PHONE Port in OBi #2 to bridge calls to OBi #1:
Physical Interfaces -> PHONE Port -> OutboundCallRoute:
{([1-9]x?*(Mpli)):pp},{(<#:>|911):li},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},
{(<**2:>(Msp2)):sp2},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):vg1}
Wow thanks for the guidance. I will give it a try and let you know what comes of it.
The 2 OBi110s are up and running. Totally amazing.
Though I have to say that no amount of reading the manual would have ever gotten me to the configuration you suggested.
Thanks for the concise directions.
What an absolutely fantastic guide!
This is exactly what I was looking for.
Incoming calls are perfect (although can anyone confirm if caller id is passed across?)
However, is it possible to shut the attendant up and on picking up the phone on OBi #2 just get a dial-tone and allow an immediate call? i.e. pick up the phone and dial 0xxxxxxxxxx
I understand the workaround to allow outgoing calls via the auto-attendant, maybe I was naive to think I could just set-up one as fxo and fxs and it would be totally transparent.
My ultimate goal is modem/fax bridging for these two obi110's.
Thanks again for the great guide, sometimes what sounds simple in theory needs a particular genius to make it happen, as snblitz said, who would've worked this method out from the instructions.
Regards,
flipdee
snblitz & flipdee,
Please revisit Reply #1 above.
EXCELLENT WORK! (Excuse the caps)
This solves ALL my problems as I was having great difficulty with OBI #1 taking too long to answer using the {li(1470)} method, chopping off the first few numbers manually dialled, accidently calling the emergency services! AHHHH! not good (especially on a device without handset).
This is exactly what I was hoping for.
I don't want to go off-topic in this very useful thread, just about to post a new topic about tone profiles if you guys wouldn't taking a look for me?
Thank you so much again, really appreciate it!
flipdee
FWIW, this scheme should support sharing the PSTN Line with up to 3 remote OBi's (or 4 is you don't need to ring phone connected to OBi #1). Simply set Voice Gateways 2, 3, and possibly 4 in OBi #1 to point to the additional OBi's and add them to the LINE Port InboundCallRoute:
Physical Interfaces -> LINE Port -> InboundCallRoute : {vg1,vg2,vg3,ph}
or
Physical Interfaces -> LINE Port -> InboundCallRoute : {vg1,vg2,vg3,vg4}
Set the PHONE Port in OBi #3, #4, and possibly #5 the same as OBi #2.
Also set the following in OBi #1:
Voice Services -> SP2 Service -> MaxSessions : 4
Nice work indeed!
Quick question, should Caller ID be automatically transfered from (PSTN) OBi #1 to FXS device on OBi #2 (3,4 etc) ?
Cheers,
flipdee
Quote from: flipdee on December 11, 2011, 05:28:08 PM
Quick question, should Caller ID be automatically transfered from (PSTN) OBi #1 to FXS device on OBi #2
I can't say. That's another feature Obihai was witholding (except on the OBiTALK trunk) for a future product, but maybe they changed their mind on it too. I don't have CallerID on my PSTN line, so I can't test it. If you discover one way or the other, please let us know.
I get callerid on OBi #1 using all default callerid settings (I was expecting to have to change FSK (Bell 202) with being in the UK.
Call 1 12/12/2011 01:30:56
Terminal ID LINE1
Peer Name
Peer Number 0xxxxxxxxxx
Direction Inbound Inbound
01:30:56 Ringing Forking to:SP2(_g1192.168.1.62:5061*), PHONE1
01:31:04 End Call
However I just get the following on OBi #2
Call 1 12/12/2011 01:30:57
Terminal ID SP2 PHONE1
Peer Name
Peer Number
Direction Inbound Inbound
01:30:57 Ringing
01:31:05 End Call
And no callerid appearing on the pots telephone.
Not absolutely necessary but it would be nice.
flipdee
P.S. I meant to say, I assume callerid does work on an FXS device connected to an OBi110?
RonR, two questions:
On outbound calls to the PSTN, couldn't you set up the system so it "hotlines" to a number on the other OBi that dials "nothing" on the LINE port, so you would hear PSTN dial tone and could dial through? This would have several advantages:
1. Faster call setup time; you wouldn't have to accumulate DTMF and then send it back out.
2. If the digit map otherwise couldn't determine end-of-number, you would avoid an interdigit timeout.
3. On calls that would be dialed manually, if the PSTN line is in use elsewhere, you'd find out before taking the effort to dial.
On calls from the PSTN to remote OBi(s), couldn't you set up SPx (instead of VGx) to point to the other OBi(s), so you could pass caller ID? Of course, this would be limited to a maximum of two remotes -- additional units would be called via VGx and would not receive caller ID.
Quote from: Stewart on December 11, 2011, 05:45:01 PM
On outbound calls to the PSTN, couldn't you set up the system so it "hotlines" to a number on the other OBi that dials "nothing" on the LINE port, so you would hear PSTN dial tone and could dial through? This would have several advantages:
1. Faster call setup time; you wouldn't have to accumulate DTMF and then send it back out.
2. If the digit map otherwise couldn't determine end-of-number, you would avoid an interdigit timeout.
3. On calls that would be dialed manually, if the PSTN line is in use elsewhere, you'd find out before taking the effort to dial.
That approach might work. Give it a try and see.
Quote from: Stewart on December 11, 2011, 05:45:01 PM
On calls from the PSTN to remote OBi(s), couldn't you set up SPx (instead of VGx) to point to the other OBi(s), so you could pass caller ID? Of course, this would be limited to a maximum of two remotes -- additional units would be called via VGx and would not receive caller ID.
AFAIK, CallerID can only be passed when one of the trunks being bridged is the OBiTALK trunk. The last I knew, Obihai intentionally retricts passing CallerID when the OBiTALK trunk is not involved:
Enhancements & Fixes in Maintenance Release 1.3.0(2575):
Version 1.3 highlights:
- Allow Caller-id spoofing for calls bridged via OBiTALK service. But use the obi number for circle-of-trust authentication.
When v1.3 first came out, it was noted that CallerID was not being passed on things like SIP/SIP and GV/SIP.
Though my OBi is presently down so I can't test, its SP2 had been registered to PBXes as a sub-PBX and successfully passed caller ID on incoming PSTN calls. PBXes then forked the call to IP phones, registered as extensions.
Stewart,
The approach you suggested works given the following changes:
OBi #1:
Voice Services -> SP2 Service -> X_InboundCallRoute : {>(<+:>):li}
OBi #2:
Physical Interfaces -> PHONE Port -> DigitMap : (<S0:+>|...)
Physical Interfaces -> PHONE Port -> OutboundCallRoute : {+:vg1},...
OBi #2 hot dials OBi #1 and you get PSTN dialtone. However, you lose the ability to use Speed Dials and Auto Attendant 2 (*** configuration) from OBi #2.
If it's important have the ability to get real PSTN dialtone on OBi #2, I would suggest the following changes:
OBi #1:
Voice Services -> SP2 Service -> X_InboundCallRoute : {>(<+:>|(Mli)):li}
OBi #2:
Physical Interfaces -> PHONE Port -> OutboundCallRoute:
{([1-9]x?*(Mpli)):pp},{(<#:+>|911):vg1},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},
{(<**2:>(Msp2)):sp2},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):vg1}
This allows dialing # from OBi #2 to get real PSTN dialtone from OBi #1. This also allows 911 from OBi #2 to go out the LINE Port of OBi #1.
Quote from: Stewart on December 11, 2011, 06:18:57 PM
Though my OBi is presently down so I can't test, its SP2 had been registered to PBXes as a sub-PBX and successfully passed caller ID on incoming PSTN calls. PBXes then forked the call to IP phones, registered as extensions.
Were the incoming calls on the LINE Port bridged to SP2 using a LINE Port -> InboundCallRoute rule?
Quote from: RonR on December 11, 2011, 07:43:44 PM
OBi #1:
Voice Services -> SP2 Service -> X_InboundCallRoute : {>(<+:>):li}
OBi #2:
Physical Interfaces -> PHONE Port -> DigitMap : (<S0:+>|...)
Physical Interfaces -> PHONE Port -> OutboundCallRoute : {+:vg1},...
OBi #2 hot dials OBi #1 and you get PSTN dialtone. However, you lose the ability to use Speed Dials and Auto Attendant 2 (*** configuration) from OBi #2.
If it's important have the ability to get real PSTN dialtone on OBi #2, I would suggest the following changes:
OBi #1:
Voice Services -> SP2 Service -> X_InboundCallRoute : {>(<+:>|(Mli)):li}
OBi #2:
Physical Interfaces -> PHONE Port -> OutboundCallRoute:
{([1-9]x?*(Mpli)):pp},{(<#:+>|911):vg1},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},
{(<**2:>(Msp2)):sp2},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):vg1}
This allows dialing # from OBi #2 to get real PSTN dialtone from OBi #1. This also allows 911 from OBi #2 to go out the LINE Port of OBi #1.
Hi RonR,
Your first suggested change above works well giving near instant pstn dial-tone on OBi #2.
However when I make the second suggested changes I get "no call route available" on OBi #2.
Should I have not make the second set of changes on top of the first?
I think the first change is better for me as the OBi bridge will only be used for PSTN>VOIP<PSTN and nothing else.
Thanks again,
flipdee
Quote from: RonR on December 12, 2011, 12:26:56 AM
Were the incoming calls on the LINE Port bridged to SP2 using a LINE Port -> InboundCallRoute rule?
Yes, I believe that the rule was just {sp2(1)}, where 1 is a ring group on PBXes that points to the IP phones. Spoofing was set in the OBi; nothing special was needed at the PBXes end.
Also, I had to increase the ring delay by one second, and change the disconnect tone to match the standard for France.
It's not a real POTS line, but a triple-play FTTH package from Orange. The ISP-supplied Livebox router has an FXS port, which I connected to the OBi Line port. Of course, this is an inelegant solution -- I would rather have connected directly to Orange by SIP. However, not only are the credentials hidden (and likely difficult to discover), the phone uses a separate VLAN, so I'd have to put a box between ONT and Livebox. (Replacing the Livebox is not practical; it's in the TV path, too, with even more security nonsense.) The intermediate box would need quite a bit of horsepower, for it not to degrade the 100 Mbps Internet connection. So, I took the lazy way out and got an OBi.
Quote from: flipdee on December 12, 2011, 03:09:15 AM
Your first suggested change above works well giving near instant pstn dial-tone on OBi #2.
However when I make the second suggested changes I get "no call route available" on OBi #2.
Should I have not make the second set of changes on top of the first?
The two sets of changes are separate, not to be combined.
is it possible with one OBi110 and another non-obihai device such as spa-1001? or even a softphone on my iphone?
Quote from: imbilly on September 29, 2014, 04:03:30 AM
is it possible with one OBi110 and another non-obihai device such as spa-1001? or even a softphone on my iphone?
Not sure what you mean by "is it possible". You need to define what you mean by "it".
If you are talking about direct ip calling between two locations on a private network without accessing other servers, that type of calling requires devices or adapters that will make calls without registering with a sip server. Using direct ip communications, OBihai adapters can communicate with each other or with Linksys/Cisco adapters such as the SPA1001, SPA3102, etc. The adapter configurations are more complex and generally will require troubleshooting to get it working.
Many softphones require registration with a sip server and consequently will not work in a direct ip calling environment. There are a few softphone exceptions. CSipSimple is an android smartphone softphone that does have the capability to be configured without registration for direct ip calling.