Help on Asterisk -> Obi110 #1 -> Obi110 #2 -> pstn setup
RonR:
Let me state up front that I only know enough about Asterisk to be dangerous.
I think the real problem is as I described in Reply #1 above. The problem is not with the OBi. The problem is how do you get the desired number through SipClient(CSipSimple) -> Asterisk(1.8.4) PBX to the OBi. If you can get the desired number through to the OBi, the rest is easy.
The ideal situation would be to have anything you dial at SipClient(CSipSimple) (such as 18005551212 or **218005551212 or 10*18005551212) be received by Asterisk(1.8.4) PBX and sent directly to OBi #1 as DialedNumber@w.x.z.y:5061 using a SIP INVITE. Then you could have OBi #1 do anything you want with it by evaluating it in the SP2 InboundCallRoute. This could include sending it on to OBi #2 via OBiTALK. The processing in the OBi is easy. What I don't know is if it's possible to get Asterisk to pass the dialed number from SipClient(CSipSimple) to OBi #1 using a SIP INVITE. If you can accomplish this Asterisk task, the rest is trivial.
Is there a particular reason you've got Asterisk in the loop? If SipClient(CSipSimple) (or some other client) can make and receive SIP calls without registration, it should be able to talk to the OBi directly and I think all your problems go away. I have a PAP2 doing this and can handle almost any dialing sequence from the phone attached to the PAP2. I have it pretty much mimicking the phone attached to the OBi's PHONE Port.
freewilly:
I like CSipSimple because it simply can be integrated to Android phone dialer and uses bluetooth while Obion can't.
It is pretty simple pass string(your obi number and pstn dial out number) from asterisk to Obi110 #1's Sp2
exten => 3334,n,Dial(SIP/200123123*18005551212@1111,20)
where
3334 is the extension I dial out from CSipSimple,
1111 is the extension of Sip at Obi110 #1's Sp2
200123123 is Obi110 #2
18005551212 is the number I want dial out from pstn
I know Obi110 #1's Sp2 receive data send from asterisk, because it try to dial out from P2P1 200123123*18005551212 under call history session.
from CLI, I see
-- Called 200123123*18005551212@1111
-- SIP/1111-0000010a is ringing
-- Got SIP response 486 "Busy Here" back from 192.168.1.109:5061
-- SIP/1111-0000010a is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [3334@normal-out:3] Hangup("SIP/1001-00000109", "") in new stack
== Spawn extension (normal-out, 3334, 3) exited non-zero on 'SIP/1001-00000109'
Maybe something in the DigitMap of ITSP Profile B need to be modified?
RonR:
The ITSP Profile B DigitMap doesn't come into play, so it's not of interest.
The way I envision this working, you don't want or need anyone to know about OBi #2 other than OBi #1.
The crucial question is:
Can you make Asterisk deliver any number you dial on CSipSimple unchanged to OBi #1 via a SIP INVITE?
If so, all the work goes on in the OBI's. I already have it implemented and tested here and can provide it to you. You'll be able to call out any trunk on OBi #1 or OBi #2 (or OBi #3, #4, #5, ...) from CSipSimple without using the AA (i.e single stage dialing) using the same methods used on the telephone connected to the OBi PHONE Port:
18005551212 -> OBi #1 PrimaryLine
**118005551212 -> OBi #1 SP1 Service
**218005551212 -> OBi #1 SP2 Service
**818005551212 -> OBi #1 LINE Port
2*18005551212 -> OBi #2 PrimaryLine
2**118005551212 -> OBi #2 SP1 Service
2**218005551212 -> OBi #2 SP2 Service
2**818005551212 -> OBi #2 LINE Port
freewilly:
I can't get this work. I believe OBi device doesn't support it.
simple test,
Obi#1.sp2.X_InboundCallRoute: {(xx.):pp(2*18005551212)}
Call Obi#1.sp2 using cell phone or any phone NOT attached to Obi#1.
It tells Obi that any incoming call of Obi#1.sp2, route to 2*18005551212, which dial 18005551212 from Obi#2.PrimaryLine.
It just simply doesn't work. Obi#1 hangup on me right way.
RonR:
With:
OBi #1 -> SP2 Service -> X_InboundCallRoute : {(xx.):pp(2*18005551212)}
and
OBi #2 -> OBiTALK Service -> InboundCallRoute : {(200123456)>(Msp1):sp1}
where 200123456 is OBi #1 OBiTALK number.
Calling OBi #1 SP2 (a VoIP provider number) from my cell phone causes 18005551212 to be dialed out SP1 of OBi #2.
Trust me, this works.
The crucial question is:
Can you make Asterisk deliver any number you dial on CSipSimple unchanged to OBi #1 via a SIP INVITE?
If not, then this may be a dead end. If so, the solution is easy.
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