Hi,
how make to connect a Goip to softswitch. Do you have a tutorial ?
flashmemory, did you manage to resolve your goip4 problem? could you please share how to distribute outbound calls to 4 different gsm channels in goip4?
Hi, this is my scenarioâŚi have a gsm gateway with a sim in it with a no say 0788xxxxx⌠I configured free pbx to work with sip phones and that part works wonderfullyâŚnowâŚwhen i want to connect my sip phones to gsm gateway to route the mobile calls through the gatewayâŚ
now, i understand that, i need to create a gsm trunkâŚ
q1. should i put the mobile number in the cid field.
q2. what details should i put in the trunk settingsâŚ
hostname
secret, MODE, IP ADDRESS,(ofcourse the gsm gateway address) and all that stuff⌠i need an explanation so that i really know what i am typing inâŚ
should i register this trunk in the registration fieldâŚif notâŚwhyâŚ
i need a point where i can actually see that the voip calls are hitting the gateway in the first placeâŚwhere can i find thatâŚ
thanks in advanceâŚ
âLEARN HONESTLY DEPLOY CONFIDENTLYâ
Can a Goip work behind a NAT?
I have a goip 8 , when a connect it with in single server mode donât work .
someone say me the probleme is my NAT .
do you have any idea?
Hola Msegura,
aplico tu setup pero no fonctiona. queria saber si tu GOIP esta detras un router o se conecta directamente in internet?
Hi everybody,
I have an Asterisk server + FreePBX in a data centre. I have a GoIP GSM gateway in my office. My ADSL has a Static IP but it comes to a switch then to GoIP.
I have attached a picture of all of the settings as bellow:
I canât terminate any call. The settings has been mentioned before. Could you please help me.
I suggest that it doesnât âcome to a switchâ but really comes instead to a router, you will need to configure that router to allow all necessary traffic to and from that host, start with UDP 5060 and UDP 10000-20000
Hi all,
I am new to goip with freepbx,I am trying for about some days to get goip on freepbx, I tried all tutorial and suggestion available on net, but could not figure out, what to do?
I have a vps with fix ip and the goip is on lan home network(dynamic ip).
If I configure goip as an extension of freebpx, I can make callout and callin both side.
I tried to configure it as a trunk at freepbx(goip in trunk gateway mode), the goip goes online(login status) but at freepbx the trunk is not registered nor online, so the situation is I can make call through incoming route from gsm to freepbx extension, but I cant make all from extension to gsm through outbound rout.
Please help me in this situation the setup is as follows:
freepbx trunk
Trunk Name= GoIP1
host=dynamic
port=5060
type=peer
context=from-internal
dtmfmode=rfc2833
insecure=very
qualify=yes
nat=force_rport,comedia
usernme=1001
secret=1234
context=from-internal
What is goip?
Lots of goip discussion at nerdvittls and piaf forum:
http://nerdvittles.com/?p=7581
http://pbxinaflash.com/community/index.php?threads/piaf-goip-gsm-gateway.13515/
I dont know about the type., but it is from hybertone model goip1.
hybertone or GOIP they are all the same. let me have my dinner first and I will get back at you.
This is per line configuration of GOIP:
In Freepbx:
Create a Trunk:
TUNK Name: whatever name
Peer Details:
allow=ulaw&alaw
host=ipaddress of GOIP or URL
port = 5060 or whatever the port address of GOIP
type=friend
nat=yes
qualify=yes
dtmfmode=rfc2833
canreinvite=no
Create an extension (not virtual)
example:
user extension = 6000
Display Name = 6000
secret = 6000GoIp
In GOIP under configuration:
Basic VOIP:
Config Mode = Config by Line
Line 1:
Phone number = 6000
Display Name = 6000
Authentication ID = 6000
Password = 6000GoIp
Gateway Prefix = 1 (for line 2,3,4 ⌠= 2, 3, 4 âŚ)
Sip Registrar Server = Server IP address
Advance VOIP:
SIP Listening Port Mode = Fixed
Port = 5060 or whatever port you assign here should be the same on the Trunk that you created.
DTMF signaling = Outband
Outband DTMF type = RFC 2883
Media:
RTP Port Range = 10000 - 20000 same as your freepbx
jitter buffer = adaptive
Media QOS = IP TOS
Tyoe of service = Routine - 0
Media NAT traversal = Port-forward/DMZ (if behind NAT)
Audio Codec Preferences = whatever codec you are using
CAll Out:
Line 1:
Call Out via GSM = enable
Dial Plan = 1:-1
restricted dial plan = disable
Try the above config for the outgoing calls. The only other thing that you need to create in Freepbx is the outbound routes.
Also on the extension that you created the sip port there should also reflect whatever the sip port you assign on your GOIP
If you still have problem let me know. We can go online together using VNC or temaviewer.
1 Like
the situation here is that I have goip1 gateway. It does not bring per line configuration and I want to configure it as trunk so any extension either from sonftphone become able through that gateway using outgoing route.
the goip is installed on lan(private network). I forworded my goip on port 80 & 5060. so it can be accessed on these ports by using my external ip. I tried all I know, but as I am new to goip it difficult for me to sort out.
Please help me in solving this problem. I can give you teamviewer, VNC even can give you access to the switch and goip, send me your e-mail and I will give you all accesses to get solved.
thanks again for your repy.
This is even easier. IN the Freepbx side the same configuration that I gave you, howeever you donât need the extension numbers, aside from the one that you use in the GOIP1 for the incoming calls. ON the GOIP configuration, you only need to enter the IP address of your server on the SIP Trunk gateway. Then on the VOIP configuration of GOIP1 it is similar to the one that I gave you. I canât remember how the GUI looks like on the GOIP1, I stop using GOIP1 since I have 4 and 8 GOIPâs. I will try to turn on one of my GOIP1 and get back at you. For the time being try the configuration that I mentioned here.
You can also download the manual of GOIP1.