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標題: [Help]Deployment Preparation [打印本頁]

作者: Qnewbie 時間: 2011-5-22 18:26 標題: [Help]Deployment Preparation

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:

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

TIA!

圖片附件: [Deployment 1] Deployment1.png (2011-5-22 18:19, 64.1 KB) / 下載次數 1634
http://telecom-cafe.com/forum/attachment.php?aid=751&k=4a9ebe80591642b19a92dc898d75926a&t=1761813088&sid=tz5T53

作者: ckleea 時間: 2011-5-22 20:49

Can your GOIP connect to the asterisk server and work well?

I understand ET263 helps a lot in quality but for redundancy,you may consider to add a sip connection to asterisk server directly.

作者: ckleea 時間: 2011-5-22 20:49

For Asterisk server, will you also consider to add the Obiapps to it to increase the flexibility

作者: Qnewbie 時間: 2011-5-22 21:05

回復 2# ckleea

The direct connection with GOIP and Asterisk is the VOIP traffic blocking in mainland China. I can only put GOIP there and cannot add one more router.

作者: Qnewbie 時間: 2011-5-22 21:10

回復 3# ckleea

It would if the Asterisk is built with PC. Any recommendation for the passive cooling ATOM D525 PC?

作者: ckleea 時間: 2011-5-22 21:16

Buy our model or the one from Magic -pro. It all depends if you need to install a PCI TDM 400P card. If you do not need, go for a slim D525 from Magic-pro. Also get a SSD will be excellent for the operation.

For me, I use 2.5" HD, it does generate some heat but mine have been on for weeks.

作者: ckleea 時間: 2011-5-22 21:17

回復 4# Qnewbie

Can you elaborate a bit for GOIP? Is it a ippbx similar to IP01?

作者: Qnewbie 時間: 2011-5-22 21:34

本帖最後由 Qnewbie 於 2011-5-22 21:35 編輯

No, it likes like combo of GSM box and single FXO gateway.

http://www.hybertone.com/en/pro_detail.asp?proid=10

作者: 角色 時間: 2011-5-22 22:18

Have you considered the trunk group of OBi device?

YH

作者: Qnewbie 時間: 2011-5-22 22:26

回復 9# 角色

I am considering which VSP can be used(better if verified) as trunk group: pennytel, iptel, sip2sip, and Betamax.

作者: ckleea 時間: 2011-5-22 22:36

回復 8# Qnewbie

I see what GOIP likes. Combo H323 + SIP. It limits your connection. I think if you have an IPO1 in china will help you a lot.

作者: Qnewbie 時間: 2011-5-22 22:56

The option of adding router or other stuff is limited. That's why I choose GOIP(mainly used as hotline).

Another route from mainland China goes thru callback with 角色 C-hing's help(not implemented fully until I visit mainland China).

作者: 角色 時間: 2011-5-28 12:22

You may use ET263 on your GOIP. Whenever your relative in Mainland China calls your mobile number in GOIP, it formware to call another ET263 account to improve the voice quality for over 15 minutes chat.

YH

作者: yiucsw 時間: 2011-5-31 02:40

I have nearly the same configuration as you but finally i give up.
I have test GOIP with use ET263 and/or IP01. and find the following problem mostly is introduced by GOIP.
1) GOIP introduce a lot of delay.
2) GOIP also do not identify itself in roaming/local call condition.
3) GOIP is not stable and will hang(need to add a timer to reboot)
4) GOIP will discharge all SMS except GOIP SMS control message.
5) GOIP cannot call forward to multiple number.

The way i handle it is two fold 1) outgoing and 2) incoming
1) outgoing - use ET263 as outgoing and apply your mobile # is CID. RMB$0.08 per minute with excellence sound quality is much better than mobile call.
2) incoming -
i) the best solution is call forward # (一号通). but now it cannot forward to HK # now. otherwise it is the best way. (1.5 per minute to call hk/macau)
ii) add a fix line and obi/spa, then mobile number is call forward to fix line. 20/month for fix line. but you loss the CID number
iii) buy an hk/mainland SIM card and call forward your mobile phone to this new HK issued sim. then in HK call forward to your HK fix line. 0.69/min + 0.2-0.5 call forward price. + monthly charge.
iv) activate the secretary service. and call back when you receive SMS message.

作者: Qnewbie 時間: 2011-5-31 05:16

本帖最後由 Qnewbie 於 2011-5-31 05:26 編輯

Thanks yiucsw C-hing!

Currently I use GOIP+IP01. The combination does work as GOIP and IP01 in the same LAN. I don't experience the delay problem(maybe the delay between GOIP&IP01 is small enough?) I use GOIP as DID, just for inbound calls.

The GOIP, as planed, is to used as hotline(飛綫). As incoming calls (only trusted numbers) will be forwarded to OBi110(HK), which in turn, either to phone@HK or to IP01(call route@Obi110). The outgoing calls are placed@IP01 with second stage dialing(GOIP => Obi110@HK=> IP01 => PSTN). Delay is the worst case. Better to launch callback@IP01 in stead

The call from HK/Aboard is just forward to a mobile number.

In my case, it is a more simple task for GOIP:
1) GOIP introduce a lot of delay. <= It would be severe!
2) GOIP also do not identify itself in roaming/local call condition. <= Hotline&fixed place, I think it would be OK.
3) GOIP is not stable and will hang(need to add a timer to reboot) <= It can be rebooted once a day with its own function.
4) GOIP will discharge all SMS except GOIP SMS control message. <= Pure voice applications.
5) GOIP cannot call forward to multiple number. <= It forwards only to one number.

I think the GOIP is not so bad for me(I might only place 1 piece of hardware in Mainland).

作者: ckleea 時間: 2011-5-31 06:33

回復 15# Qnewbie

Very interesting discussion. Could you elaborate on your dialplan to make things work? It will be useful for us to learn.

作者: Qnewbie 時間: 2011-5-31 16:07

回復 16# ckleea

I don't get Obi110 in my hand. So the detailed call route@HK would be "Help"

There is not fancy stuff for the dialplan. It should be a simple AA like "to call HK, press 1, to call other destinattion, press 2" and hopefully, ET263 and Obi110 could handle this.

On IP01, the dialplan is quit simple. Just treat OBi110@HK as a trunk. Inbound calls are directed to IVR. Outbound call is placed to OBi110.

作者: yiucsw 時間: 2011-5-31 22:08

Your config is my dream world.
Anyway, for incoming call forward (china->HK) GOIP has one call forward function, which you can call forward to OBIHAI's et263 SIP client. There is a multiple trunk mode, i have not success to enable it.
Pls let me know if you can. So without trunk mode, you cannot call forward to multiple location (hk and abroad).
OBHHAI (HK) should able call forward to (LI).

Do not understand your outgoing statement, you mean you want to implement two way call forward?
you mean dial to IP01(PTSN), call forward to GOIP(IP01)?
or you just need to call OBI110(HK) then call forward to GOIP(phone), then from GOIP(call forward to you china number)?

Anyway,
have not test call forward in OBIHAI. you really remind me that it is a very simple way to call forward from china to HK>

作者: Qnewbie 時間: 2011-5-31 23:00

回復 18# yiucsw

Thanks again!

I am not sure that this config would work.

The two way call forward is not a general one. Call is just forward and backward to a fixed mobile number:
Mobile 1380013800 <=> GOIP <=> ET263 <=> Obi <=> IP01 <=> Other destination

The only thing that works is GOIP acts as DID for mobile phone. The GOIP register to IP01 within the same LAN and second stage dialing works. But I am not sure that the second stage dialing works from mobile to IP01. The outgoing call is mobile to other destination THRU ALL the way.

作者: Qnewbie 時間: 2011-5-31 23:13

In GOIP, there is hotline function.

You can forward to ET263 number(call initiates from mobile) or to mobile number(call from ET263 number).

作者: yiucsw 時間: 2011-6-1 16:03

To me, i need the CID in order to know who is calling
And cannot accept the delay with pstn->VOIP->VOIP->pstn, during office hours.

but i plan to use obihai call forward function after office hours.

作者: Qnewbie 時間: 2011-6-1 17:25

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.

作者: yiucsw 時間: 2011-6-6 00:24

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.

作者: Qnewbie 時間: 2011-6-6 05:27

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.

作者: yiucsw 時間: 2011-6-10 00:37

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)

作者: Qnewbie 時間: 2011-6-10 02:13

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...

作者: yiucsw 時間: 2011-6-10 03:00

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.

作者: Qnewbie 時間: 2011-6-10 03:36

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.

作者: yiucsw 時間: 2011-6-22 17:52

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

作者: ckleea 時間: 2011-6-22 18:02

回復 29# yiucsw

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

作者: yiucsw 時間: 2011-6-23 01:32

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?

作者: Qnewbie 時間: 2011-6-23 03:02

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.

作者: yiucsw 時間: 2011-6-23 14:16

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?

作者: Qnewbie 時間: 2011-6-23 16:09

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.

作者: ckleea 時間: 2011-6-23 17:20

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.

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