One way audio with PPTP VPN after 3-6 mins

Asterisk Asterisk Support
1

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]
2

Do a “sip set debug on” in Asterisk CLI, make a call and copy/paste the output of the command here.

3

Hey
How r u
I dont have a solution for your techy question but i can suggest to contact PureVPN customer support team. I am damm sure that they will gonna help you.
Thanx