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7 posts

Wannabe Geek


Topic # 66125 13-Aug-2010 00:53
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Hi guys recently I bought a GoIP gsm gateway but until now Im struggling how to make it work. I use a vodafone simcard and italk line.
Maybe someone here was using GoIP and has a working system. Could you guys shade me some light on this.
Thanks

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26927 posts

Uber Geek
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  Reply # 367004 13-Aug-2010 06:29
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It's hard for anybody to offer advice when you make no mention of all what is wrong or what you are trying to do.



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Ultimate Geek
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  Reply # 367026 13-Aug-2010 08:39
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Here's a few things to think about telling us:
1) What OS you're running (Elastix / Trixbox?)
2) Have you done any setup at all?
3) What does asterisk show if you run: asterisk -rx 'sip show peers'
4) What documentation have you read already?



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Wannabe Geek


  Reply # 367404 13-Aug-2010 23:06
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sorry guys here is the details.

I don't use asterisk basically I just have an linksys PAP2 ATA.
What I want is to connect the GoIp into my ATA directly just to test if I could get a dialtone.
As what I understand GoIP acts as a trunkline so basically I can plug it directly to an ATA device 
My problem the configuration how to register it into the ATA.
here is the link of goip manual..
http://www.hybertone.com/en/PDF/User%20Manual%20GoIP_EN.pdf

Thanks.

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  Reply # 367467 14-Aug-2010 09:25
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A PAP2 is an FXS device. It's an adapter for analogue phones, hence the reason it's an ATA (analogue telephone adapter). It is not for hooking up analogue lines which is presumably what you want to do.

You need a device with an FXO port such as a SPA3102 to hook up a phoneline.

I also completely fail to understand from your post what you want to do. If you're not running a PBX of any sort I don't see why you would want or need any sort of VoIP adapter.






7 posts

Wannabe Geek


  Reply # 368982 17-Aug-2010 23:20
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Hi guys I decided to install the asterisk and test the goip gsm gateway and successfully registered. I test the outbound and it was working. My problem now was the inbound. Everytime I call my simcard it says "we're sorry your call cannot be completed at this time pls hangup and try to call later". I checked the cli and it doesn't show anything that the incoming call was coming through.

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  Reply # 369019 18-Aug-2010 06:33
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chriserasmo: Hi guys I decided to install the asterisk and test the goip gsm gateway and successfully registered. I test the outbound and it was working. My problem now was the inbound. Everytime I call my simcard it says "we're sorry your call cannot be completed at this time pls hangup and try to call later". I checked the cli and it doesn't show anything that the incoming call was coming through.


Once again this could be for a number of reasons and the above comment means very little.



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Wannabe Geek


  Reply # 369810 19-Aug-2010 14:19
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This is the complete details:


This is my elastix config

Peer Details:

------------------
[gsm1]
allow=g729
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
fromuser=gsm1
host=192.168.1.101
insecure=very
secret=gsmtest
type=friend
username=gsm1

Registration String:

----------------------
gsm1:gsmtest@192.168.1.101/gsm1

Extensions.conf

-----------------------------
; dialout using default OUT trunk – no prefix
[macro-dialout-default]
exten => s,1,Macro(user-callerid,SKIPTTL)
exten => s,2,Macro(record-enable,${CALLERID(number)},OUT)
exten => s,3,Macro(outbound-callerid,${ARG1})
exten => s,4,Dial(${OUT}/${ARG1})
exten => s,5,Playtones(congestion)
exten => s,6,Congestion(5)
exten => s,105,Macro(outisbusy)

[from-sip-external]
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($${EXTEN:1:2}"="?s:${EXTEN})})
exten => _.,n,Goto(s,1)
exten => s,1,GotoIf($${ALLOW_SIP_ANON}"="yes?from-trunk,${DID},1)
exten => s,n,Set(TIMEOUT(absolute)=15)
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,Playtones(congestion)
exten => s,n,Congestion(5)
exten => h,1,NoOp(Hangup)
exten => i,1,NoOp(Invalid)
exten => t,1,NoOp(Timeout)

GOIP Configuration/Settings..

GoIp IP Address: 192.168.1.101
GoIP Call Settings

--------------------
Endpoint: SIP Phone
Config Mode: Single Server Mode
Phone Number: gsm1
Display Name: gsm1
SIP Proxy: 192.168.1.101
SIP Registrar Server: 192.168.1.101
Register Expiry: 3600
Outbound Proxy:
Home Domain:
Authenication ID: gsm1
Password: gsmtest
Call Forward Type: Not Forward
Call Forward Number:

Call Divert

----------------------
Forward to VOIP: Enable
Forward Number(PSTN to VOIP): 3000



7 posts

Wannabe Geek


  Reply # 370867 22-Aug-2010 00:41
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Hi guys, my problem was solved it was the GoIP configuration also with my codec on asterisk..
Thanks..

alx

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Wannabe Geek


  Reply # 371303 23-Aug-2010 01:37
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How is the quality of GoIP? Is it stable or does it need several reboots a day? I have also been considering to buy of this unit from ebay but Im afraid the quality is poor. Can someone confirm the quality of this unit?

RE,
ALX 



7 posts

Wannabe Geek


  Reply # 384699 26-Sep-2010 23:12
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in terms of quality it was perfect. For me it was stable

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