(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 32.70.0.130
The first line of my phone connected to our internal office IP PBX (asterisk on 192.168.182.0/24 network), the phone internal IP is 192.168.182.192
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 : 192.168.182.192
Addr->IP : 192.168.182.192:14926
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 32.70.0.130
Reg. Contact : sip:200@10.8.3.197:14926;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@10.8.3.197:14926;transport=TLS
the ip address of 10.8.3.197 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 10.8.3.197
RTP Debugging Enabled for address: 10.8.3.197:0
== 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 10.8.3.197:11782 (type 00, seq 035451, ts 000160, len 000170)
Sent RTP packet to 10.8.3.197:11782 (type 00, seq 035452, ts 000320, len 000170)
RTP packets are not reaching my phone, because PBX does not know anything about 10.8.3.197 ip address.
The quick solution I use is adding this route on PBX server.
Quote: Destination Gateway Netmask Interface
10.8.3.197 192.168.182.192 255.255.255.255 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???
