Call Forward (unconditional) & dialling out on Obitalk 200 ---Help needed.
azrobert:
You won't be able to pass a number to the remote site using Skype.
The only way I know how to accomplish your requirements is to use the Auto Attendant at the remote site to determine routing.
You definitely can only have 1 OBiLine connected to the OBi200, so you will need an OBi110.
Your setup would look like this:
OBi200>OBiLine>FreeTalk>Skype>FreeTalk>OBi110>OBi200>AA>
OBiLine>PSTN
OBiBT -> Cell
Phone Port
I helped someone with a setup like this:
OBi202>OBiTalk>OBi202->OBi110>PSTN
This is the closest I came to the above setup, but it should work.
I didn't know SipSorcery and PBXes supported TLS.
SipSorcery would greatly reduce the complexity.
It would look like this:
OBi200>SipSorcery>OBi200>OBiLine->PSTN
OBiBT -> Cell
Phone Port
Using SS you can pass a number to the remote OBi200 and determine routing based on that number. I have an SS account, so I can help with the dialplan. I tried connecting my OBi to SS using TLS and it failed. I just changed the transport method to TLS and the registration port to 9712. Don't know if additional changes are required. I have a Free SS account, so maybe TLS is not supported with these accounts.
I don't know how to do the same with PBXes. You would have to use the AA to determine routing.
azrobert:
I tried registering with SipSorcery with TLS transport again and it worked!
I tried using registration port 5061 this time.
There is a TLS bug at SipSorcery, but when it happens they automatically restart the server.
I saw a post on their forum.
I don't know if this bug has been fixed.
azrobert:
If you want to use SipSorcery it would be a very easy setup.
Each OBi200 would register at SipSorcery as an extension.
Then you could do something like the following:
Dial 0 to ring the remote OBi200
Dial 9 to get dial tone on the remote PSTN
Dial 9 + a number (98005551212) to call out on the remote PSTN
Dial a number with a different prefix to call out the remote BT.
If you want to go the route, tell me what prefixes you want to use and the format of the outbound number.
SipSorcery cost $69 per year.
http://www.sipsorcery.com/mainsite/Home/Pricing
The no longer offer free accounts.
I believe they have a 30 day money back guarantee.
realjohny:
Thank you for your reply.
I used SS till last year, forgetting if it was working on different ports in this restricted VoIP zone. But as SS doesn't handle media it has less capability to bypass restrictions to my understanding.
So If I renew my SS will I use it with AA? I am wondering if I call paired BT phone with local Obi from my mobile will AA attend this call And then SS will route it to remote Obi (also connected with PSTN & BT Phone) giving me a dial tone ?
Before renewing SS I am wondering if this all can be easily achieved.
I can recall that PBXES was able to somehow bypass restrictions than SS.
azrobert:
I lied to you about PBXes.
I forgot about a function they have, so you can so the same as SipSorcery.
I have to refresh my memory on how it works.
I tried to register to PBXes using TLS and port 5070 and it failed.
I changed it to UDP and port 5060 and it registered, so I have the credentials correct.
You might have to figure out how to register to PBXes.
I have a free account and I don't know if TLS is included.
For both SipSorcery and PBXes:
You don't need the AA at the remote site.
From the local phone port you don't need the AA locally.
If you want to call the cell attached to the local BT then you need the local AA.
FYI, when I was testing PBXes the latency was horrible.
I found I was registering to their European server.
Switching to the west coast server improved the latency, but it was still noticeable.
This was a long time ago.
I don't know if anything has improved.
I don't know where your sites are, but you might have latency problems.
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