I am experiencing one way audio after 3-6 minutes of established call.
Here is the network settings
The NAT setting of Asterisk is disabled.
The network of SIP client connect to the network of PBX thru PPTP VPN. The subset of the client network is 192.168.6.x and the PBX side is 192.168.10.x
The client side is using Grandstream 1450
As you see, extension 102 called extension 100. The “Theoretical Address” and “Received Address” are different, does that causing trouble?
[code]trixboxCLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message
192.168.10.216 (None) 61e183a46e2488f 0x0 (nothing) No Init: OPTIONS
192.168.10.216 100 6b5175f144b0cca 0x4 (ulaw) No Tx: ACK
192.168.10.216 102 1888224767-5060 0x4 (ulaw) No Rx: ACK
3 active SIP dialogs
trixboxCLI> sip show channel 6b5175f144b0cca
trixbox*CLI>
- SIP Call
Curr. trans. direction: Outgoing
Call-ID: 6b5175f144b0cca90bfba7516381151a@192.168.10.101
Owner channel ID: SIP/100-000034b6
Our Codec Capability: 2621452
Non-Codec Capability (DTMF): 1
Their Codec Capability: 12
Joint Codec Capability: 12
Format: 0x4 (ulaw)
T.38 support No
Video support Yes
MaxCallBR: 384 kbps
Theoretical Address: 192.168.6.10:55926
Received Address: 192.168.10.216:55926
SIP Transfer mode: open
NAT Support: Always
Audio IP: 192.168.10.101 (local)
Our Tag: as770098fd
Their Tag: 341372295
SIP User agent: Grandstream GXP1450 1.0.5.24
Username: 100
Peername: 100
Original uri: sip:100@192.168.6.10:55926
Need Destroy: No
Last Message: Tx: ACK
Promiscuous Redir: No
Route: sip:100@192.168.6.10:55926
DTMF Mode: rfc2833
SIP Options: (none)
Session-Timer: Inactive
trixbox*CLI>
trixboxCLI> sip show channel 1888224767-5060
trixboxCLI>
- SIP Call
Curr. trans. direction: Incoming
Call-ID: 1888224767-5060-5@BJC.BGI.G.CAC
Owner channel ID: SIP/102-000034b5
Our Codec Capability: 2621452
Non-Codec Capability (DTMF): 1
Their Codec Capability: 7437
Joint Codec Capability: 12
Format: 0x4 (ulaw)
T.38 support No
Video support Yes
MaxCallBR: 384 kbps
Theoretical Address: 192.168.6.202:56118
Received Address: 192.168.10.216:56118
SIP Transfer mode: open
NAT Support: Always
Audio IP: 192.168.10.101 (local)
Our Tag: as41b22aad
Their Tag: 666293171
SIP User agent: Grandstream GXP1450 1.0.5.24
Username: 102
Peername: 102
Original uri: sip:102@192.168.6.202:56118
Caller-ID: 102
Need Destroy: No
Last Message: Rx: ACK
Promiscuous Redir: No
Route: sip:102@192.168.6.202:56118
DTMF Mode: rfc2833
SIP Options: replaces replace timer path
Session-Tim[/code]