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

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

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.

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

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.

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)

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

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]