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Multiple OpenVPN connections needed
Author Message
funchaser Offline
Junior Member

Posts: 2
Joined: Apr 2014
Reputation: 0
Post: #10
RE: Multiple OpenVPN connections needed
(08-01-2013 10:38 AM)Yealink Support Wrote:  
(07-26-2013 03:08 PM)davidywilson Wrote:  I'd vote for this feature too.

I don't actually need to have two vpns active but even with one vpn active it appears that using the other accounts that don't need the vpn wont work.

As soon as I activate the vpn I get no audio on the other accounts that don't need the vpn. So it seems that its either all accounts via vpn or none.

Hi Davidywilson,
It seems that your VPN server didn't configure client-to-client, could you check for this?

I have similar problem.
I use Yealink SIP-T32G fw ver
The first line of my phone connected to our internal office IP PBX (asterisk on network), the phone internal IP is
The second line of my phone connected to remote IP PBX over the OpenVPN. The second line works just fine!
How ever the first line has the same "no audio" problem.
CLI [sip show peer] for this line looks like this -

Quote: * Name : 200
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-internal
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup : 10
Pickupgroup : 10
MOH Suggest :
Mailbox : 200@default
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "200" <200>
MaxCallBR : 384 kbps
Expire : 3593
Insecure : port,invite
Force rport : Yes
ACL : Yes
DirectMedACL : No
T.38 support : Yes
T.38 EC mode : FEC
T.38 MaxDtgrm: -1
DirectMedia : Yes
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : Yes
Ign SDP ver : No
Trust RPID : Yes
Send RPID : Yes
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP :
Defaddr->IP : (null)
Prim.Transp. : TLS
Allowed.Trsp : TLS
Def. Username: 200
SIP Options : (none)
Codecs : 0x191f (g723|gsm|ulaw|alaw|g726|g729|g726aal2|g722)
Codec Order : (ulaw:20,alaw:20,gsm:20,g722:20,g723:30,g726:20,g729:20,g726aal2:20)
Auto-Framing : No
Status : OK (35 ms)
Useragent : Yealink SIP-T32G
Reg. Contact : sip:200@;transport=TLS
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : Yes

Obviously, the error is here - Reg. Contact : sip:200@;transport=TLS

the ip address of is incorrect, it is belong to the tun interface;
as a result all RTP traffic from PBX to my phone looks like this

Quote:sip*CLI> rtp set debug ip
RTP Debugging Enabled for address:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [*97@from-internal:1] Answer("SIP/200-0000247c", "") in new stack
-- Executing [*97@from-internal:2] Wait("SIP/200-0000247c", "1") in new stack
Sent RTP packet to (type 00, seq 035451, ts 000160, len 000170)
Sent RTP packet to (type 00, seq 035452, ts 000320, len 000170)

RTP packets are not reaching my phone, because PBX does not know anything about ip address.
The quick solution I use is adding this route on PBX server.
Quote: Destination Gateway Netmask Interface eth0
How ever, this is very inconvenient, because if ip addresses change, I will have to add new route on the server box!

Any other solution from Yealink???
(This post was last modified: 04-24-2014 08:11 PM by funchaser.)
04-24-2014 08:10 PM
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RE: Multiple OpenVPN connections needed - funchaser - 04-24-2014 08:10 PM

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