Asterisk GSM Gateway

Asterisk Asterisk Support
1

Does anyone know how to configure Portech MV-370/MV-372 GSM Gateway with asterisk. So far i have:

sip.conf

[103] type=friend username=103 fromuser=103 regexten=103 secret=mypassword context=gateway dtmfmode=inband call-limit=1 callerid=GSM Gateway <103> host=dynamic nat=no canreinvite=no insecure=very qualify=yes disallow=all allow=ulaw allow=alaw

extensions.conf

[code] [gateway]
exten => 103,1,Answer()
exten => 103,2,Set(TIMEOUT(digit)=3)
exten => 103,3,Set(TIMEOUT(response)=5)
exten => 103,4,DISA(no-password,outgoing)

[outgoing]
exten => 888,1,SetCallerID("SIMNUMBER")
exten => 888,2,Dial(SIP/${EXTEN}@103,60,r)
exten => 888,3,Hangup()[/code]

However, I get a busy message when I dial the SIMNUMBER.
Any thoughts?

2

On the gateway set up * route in “Mobile to LAN” and “LAN to Mobile” part of the gateway.

On Asterisk just set up a SIP peer with no authentication. Don’t complicate things with registrations.

3

Hello all I have one error but i don’t understand why.
The gateway GSM return with Busy.

I see my CLI
== Using SIP RTP CoS mark 5
– Executing [96xxxxx@default:1] Set(“SIP/10004-00000000”, “10004=96xxxxxx”) in new stack
– Executing [96xxxxxx@default:2] Dial(“SIP/10004-00000000”, “SIP/10001/96xxxxxx,300,tTg”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/10001/961366207
– SIP/10001-00000001 is ringing
– Got SIP response 486 “Busy Here” back from 192.168.0.100:5060
– SIP/10001-00000001 is busy
== Everyone is busy/congested at this time (1:1/0/0)
– Executing [9xxxxx@default:3] Hangup(“SIP/10004-00000000”, “”) in new stack
== Spawn extension (default, 96xxxxxx, 3) exited non-zero on ‘SIP/10004-00000000’

SIP set Debug on

<— SIP read from UDP:192.168.0.100:5060 —>
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.0.42:5060;branch=z9hG4bK6611bfec
From: “10004” sip:10004@192.168.0.42;tag=as61221aa4
To: sip:96xxxxxx@192.168.0.100:5060;tag=548e4cbb
Call-ID: 2a34c5c864f3a2430a75faac3df85ff0@192.168.0.42:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
– Got SIP response 486 “Busy Here” back from 192.168.0.100:5060
set_destination: Parsing sip:10001@192.168.0.100:5060 for address/port to send to
set_destination: set destination to 192.168.0.100:5060
Transmitting (no NAT) to 192.168.0.100:5060:
ACK sip:10001@192.168.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.42:5060;branch=z9hG4bK6611bfec
Max-Forwards: 70
From: “10004” sip:10004@192.168.0.42;tag=as61221aa4
To: sip:96xxxx@192.168.0.100:5060;tag=548e4cbb
Contact: sip:10004@192.168.0.42:5060
Call-ID: 2a34c5c864f3a2430a75faac3df85ff0@192.168.0.42:5060
CSeq: 102 ACK
User-Agent: Agora VOIP
Content-Length: 0

-- SIP/10001-00000003 is busy

== Everyone is busy/congested at this time (1:1/0/0)
– Executing [96xxxx@default:3] Hangup(“SIP/10004-00000002”, “”) in new stack
== Spawn extension (default, 96xxxx, 3) exited non-zero on 'SIP/10004-00000002’
Scheduling destruction of SIP dialog ‘WT-rjZfgDLZz5VDZYIrO2A…’ in 32000 ms (Method: INVITE)