Incoming gateway problem -- Help Plz?

Hi,

I have a Grandstream GXW4104 which is working outbound fine. Problem is NO incoming calls.

The Grandstream device is set up for Unconditional Call Forward to VOIP: ch1-4:101; using p1 = profile 1
Profile 1 is 101 and password is say 1234

In FreePBX the SIP Trunk is set up as following: Outgoing settings which is working fine=

Trunk Name: 4041231234

PEER Details:

host=192.168.1.205 <------ this is the Grandstream GXW4104 external gateway
secret=1234
type=peer
username=101

The incoming settings are: Not working as set

USER Context: 101
allow=all
context=incoming
host=192.168.1.205
secret=1234
type=friend
username=101

Registration String is blank.

I’ve tried various settings and have set up and extension to 101 to route but the pbx CLI doesn’t show any activity on incoming calls. Actually the Channel LED on the Grandstream device doesn’t come on for incoming calls but does for outgoing.

Does anyone have any ideas? I know most people here have problems with outgoing but that’s working fine.
Any ideas for suggestions will be greatly appreciated!

you allow anonymous sip connections? you can find the setting in the general settings page. needs to be yes for sip trunk.

I appreciate any advice, but I didn’t see in the trunk setting or on the Grandstream anything about anonymous sip connections.

But, I have found something new. I have two lines connected to this Grandstream external gateway. BOTH lines work fine for all outgoing calls but not incoming. When I call the first line nothing appears in the CLI. But for the second line this is the output:

Connected to Asterisk 1.4.18.1 currently running on pbx (pid = 2779)
Verbosity is at least 3
– Executing [101@from-sip-external:1] NoOp(“SIP/192.168.1.200-092be4d8”, “Received incoming SIP connection from unknown peer to 101”) in new stack
– Executing [101@from-sip-external:2] Set(“SIP/192.168.1.200-092be4d8”, “DID=101”) in new stack
– Executing [101@from-sip-external:3] Goto(“SIP/192.168.1.200-092be4d8”, “s|1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/192.168.1.200-092be4d8”, “0?from-trunk|101|1”) in new stack
– Executing [s@from-sip-external:2] Set(“SIP/192.168.1.200-092be4d8”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2008-10-16 19:26:53 UTC.
– Executing [s@from-sip-external:3] Answer(“SIP/192.168.1.200-092be4d8”, “”) in new stack
– Executing [s@from-sip-external:4] Wait(“SIP/192.168.1.200-092be4d8”, “2”) in new stack
– Executing [s@from-sip-external:5] Playback(“SIP/192.168.1.200-092be4d8”, “ss-noservice”) in new stack
– <SIP/192.168.1.200-092be4d8> Playing ‘ss-noservice’ (language ‘en’)
== Spawn extension (from-sip-external, s, 5) exited non-zero on ‘SIP/192.168.1.200-092be4d8’
– Executing [h@from-sip-external:1] NoOp(“SIP/192.168.1.200-092be4d8”, “Hangup”) in new stack
– Executing [h@from-sip-external:2] Set(“SIP/192.168.1.200-092be4d8”, “DID=s”) in new stack
– Executing [h@from-sip-external:3] Goto(“SIP/192.168.1.200-092be4d8”, “s|1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/192.168.1.200-092be4d8”, “0?from-trunk|s|1”) in new stack
– Executing [s@from-sip-external:2] Set(“SIP/192.168.1.200-092be4d8”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2008-10-16 19:26:59 UTC.
– Executing [s@from-sip-external:3] Answer(“SIP/192.168.1.200-092be4d8”, “”) in new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on ‘SIP/192.168.1.200-092be4d8’

Of course, it plays the no service message but I’m now wondering WHY does port 2 answer but not port 1. I would really like to see the config files for someone that has this working properly. I’m wondering if it’s my Grandstream gateway that might be bad. I know there is a newer firmware but I’ve heard horror stories on problems flashing. I also notice Grandstream’s site has update pending for this device.
But I would think that if the outgoing part is working fine the incoming calls should also. HELP!

ok
change type=peer to type=friend

I appreciate any advice, but I didn’t see in the trunk setting or on the Grandstream anything about anonymous sip connections.

Try looking in the General Settings tab in freepbx, scroll down to the bottom, and make sure Allow Anonymous SIP calls is set to Yes