I am trying to connect with MV-378 Gsm gateway with Freepbx. and while am calling time i can see recieved an unknown call

FreePBX Installation / Upgrade
1
  • Executing [2222@from-pstn:1] Set(“SIP/gsm1user-000000f6”, “__FROM_DID=2222”) in new stack
    – Executing [2222@from-pstn:2] NoOp(“SIP/gsm1user-000000f6”, “Received an unknown call with DID set to 2222”) in new stack
    – Executing [2222@from-pstn:3] Goto(“SIP/gsm1user-000000f6”, “s,a2”) in new stack
    – Goto (from-pstn,s,2)
    – Executing [s@from-pstn:2] Answer(“SIP/gsm1user-000000f6”, “”) in new stack
    – Executing [s@from-pstn:3] Wait(“SIP/gsm1user-000000f6”, “2”) in new stack
    – Executing [s@from-pstn:4] Playback(“SIP/gsm1user-000000f6”, “ss-noservice”) in new stack
    – Playing ‘ss-noservice.ulaw’ (language ‘en’)
    – Executing [s@from-pstn:5] SayAlpha(“SIP/gsm1user-000000f6”, “2222”) in new stack
    – Playing ‘digits/2.ulaw’ (language ‘en’)
    – Playing ‘digits/2.ulaw’ (language ‘en’)
    – Playing ‘digits/2.ulaw’ (language ‘en’)
    – Playing ‘digits/2.ulaw’ (language ‘en’)
    – Executing [s@from-pstn:6] Hangup(“SIP/gsm1user-000000f6”, “”) in new stack
    == Spawn extension (from-pstn, s, 6) exited non-zero on ‘SIP/gsm1user-000000f6’
    – Executing [h@from-pstn:1] Hangup(“SIP/gsm1user-000000f6”, “”) in new stack
    == Spawn extension (from-pstn, h, 1) exited non-zero on ‘SIP/gsm1user-000000f6’
2

You need to setup a trunk for the gateway. Use the from-internal context.

If the gateway is on a static IP it won’t need to register you can simply populate the trunk with the host IP and Asterisk will match on that peer.

3

I have a trunk account i created already in Sip trunk
can you help me in more details please

4

Impossible to help with the limited information you gave.

Have you read the Asterisk documentation on creating SIP peers? With the GSM gateway on a fixed IP you don’t need to authenticate to Asterisk just match on the IP, set the context to from-trunk and allow the CODEC’s you want.

5

Hi muneerbapu,
Maybe this can help:
http://samyantoun.50webs.com/asterisk/freepbx/portechmv370
You may need to change some settings to meet your needs (Gain, Band, IP’s and Codecs)

6

Such as what are your trunk settings ? (sanitized passwords and such…)

7

I create Sip Trunk Account with the following details

General Settings

Trunk Description: GSM Gateway
Maximum Channels: 2

Outgoing Dial Rules

Dial Rules: .

Outgoing Settings

Trunk Name:
PEER Details:
type=peer
username=712055
secret=passw0rd
host=192.168.1.8
context=from-internal
disallow=all
allow=ulaw&alaw
canreinvite=no
dtmfmode=rfc2833
insecure=very
nat=no
qualify=yes

Incoming Settings

USER Context:
USER Details:

username=712055
secret=pass159
type=peer
context=from-trunk
qualify=yes

Register String: 712055:[email protected]