[Help]Deployment Preparation

Qnewbie

This post looks like cross different topics. Just pick this as Obi110 is unavailable for me currently.

I am planing to set up system for voice communication between HK, Mainland, and aboard.
Main system is located aboard with 1 IP01, 1 Obi110. Mainland China(1 GOIP) and HK(1 Obi110) is used as two branches.

Wanted:
Redundancy: one peer is down, other two peers can communicate with each other. The only exemption is mainland China, as GOIP or ET263 is down, then every thing from/to mainland is down. It can only be compensated by VSP.

Preliminary planing:

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Deployment 1

2011-5-22 18:19

Help is needed for:
1. Call route for Obi110@HK.
2. Better deployment.

TIA!

ckleea

I have not had time to test delay but overall the performance of a voice call from HK to a USB stick that attached to the server is quite good. Voice quality is really satisfactory.

I am not sure about how to carry callerID from one to another. However, having read your requirement, it seems to me that GOIP may be one of the problems. Internet connection quality is another.

I have tried this configuration and found quality is acceptable except the volume of voice is a bit lower.

laptop on a UK train -> wireless UK -> IP01 (UK) -> asterisk server (HK) -> HKBN -> mobile phone in HK

For SMS, our configuration has no problem on incoming. Chinese characters are received correctly and then sent out by email. For outgoing, we have two options, either command line or through a web gui program. No Chinese character can be sent.

Qnewbie

Ask the sale personnel in ATCOM, they should give you at least the best possible delay time. So you know the scale of delay time.

BTW, Ckleea & bubblestar C-hing have tested the USB stick for voice&SMS. You might ask them for more info in the delay issue. I think it quite reasonable that ATCOM's GSM module has better performance.

yiucsw

issue is that the only working model is GSM call forward to PSTN.
GOIP(GSM) -> GOIP(VOIP) -> OBI(VOIP) -> OBI(PSTN) is long path.
GSM Box -> Phone port -> PSTN(SPA(3102)).  is much shorter path. but delay inside SPA is still long.
so i guess ATCOM (GSM)-> ATCOM (PSTN) will be the shortest path. ?
do you know anyone has test the environment?

Qnewbie

I have to fine tuned IP01's FXO port. Maybe you need to do that too.

I think "ASK before BUY" is the best you can get right now.

I still need to buy one GOIP. But US$199 is ......

GSM box+SPA3000/SPA3102 has less delay than GOIP, which is the test result from YHChing.

yiucsw

IP01 PSTN port is not working, and the OBI cannot have local access function as SPA3102, so i use ET263 call forward function. It may add additional delay.
I can clear hear the echo after i say "1 2 3 4".
That is why i plan to buy the ATCOM GSM/FXS/FXO device. what is local route.
What is your suggestion. ATCOM GSM gateway or build an ASTerisk server with GSM and PSTN module?

ckleea

回復 29# yiucsw


Do you have problem in the network set up? It seems 2 sec is too long.

yiucsw

I try the configuration again but no luck.
GOIP (ET263) -> call forward -> OBI (ET263) -> call forward to landline. the delay is 2 sec. All in same lan segment. So you are luck for your config.

May be buy the ATCOM GSM gateway and try again

Qnewbie

I am not sure.

If you are using GOIP from DBL. There is remote control. It might help.

"Call Diversion lets you divert your calls to almost any phone, including your mobile. Call Diversion can also divert calls while your phone line is in use." Call forward is just transfer your call, not matter what it is. Actually I don't see any difference for GOIP.

yiucsw

i already install a very high DB antenna. 12 DB. for GSM connection.
Will try the local call forward function again.
Do you know if I can login to GOIP (trunk mode) and use OBIHAI as client?
or
what is the different of call forward mode in call setting and call divert.

Qnewbie

The internet connection is a must for VOIP. If the bandwidth cannot be guaranteed, all other attempts are in vain.

GOIP needs a better antenna to improve the voice quality(GSM problem?).

You might to replace GOIP with GSM box+SPA3000. Still, GOIP(anttena)+VPN would be my dream...

yiucsw

want to set up GOIP again. to forward call from beijing.
setup is fast. call forward to OBI SP2, then call forward to landline.
The sound quality is so so, need fine tune. (OBI SPX need time to stable the sound quality)

Then ask myself, why not directly call forward from GOIP to PSTN. change the call forward # et263 to 0+ china number. seem not convince and troublesome.

So ask ET263 helpdesk for set call forward number permanent for me.
XXXX3 is to call forward to my macau phone number. ...
so I only need to login goip and switch et263 number and incoming call will forward to different number.

But a plan is a plan, just after implement, i found the sound quality is bad.
Do not have energy to find out way. Any suggestion.
(The network quality may be bad, i may bring the GOIP to china for another test)

Qnewbie

Sounds deployment of company branches.

If I were you, I prefer place DD-WRT/OPENWRT routers with openvpn for connections. Each one could possible with asterisk, or connect to asterisk at HK(or place with redundant internet connections). With this deployment, it might solve the delay and voice quality problem.

yiucsw

you are lucky,
I just try the following call forward in this weekend.
1) BJ mobile call CN mobile (bj) ->(call forward)-> to landline (gz) -> obi(et263) -> hk(smv)-macau(smv). the quality is unacceptable
2) BJ mobile call CN mobile (bj) ->(call forward)-> to landline (gz) -> obi(et263) -> macau(smv). the quality is unacceptable at first 5 min, then better later.
3) BJ mobile call CN mobile (bj) -> (call forward) -> Macau (CDMA) the quality is good for first 5 min, bad for 10 sec, and repeat this frequency.

so i plan to buy an "GSM call forward to PSTN platform". Any suggestion?
Mobile (BJ mobile) -> platform -> Macau landline -> macau mobile.
which i can bypass all the route. and improve the sound quality.

Qnewbie

回復 21# yiucsw


    During the office hour, you might use callback function(it would be convenient that Asterisk is implemented with PC).

For me, the CID is "known"(restricted by trust list in GOIP). Hence, I prefer this config

There is CID call forwarding function in GOIP, I haven't tested it. If you use GOIP+IP01(direct connection), you might test it.