No audio on external extension

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voipoint

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Hi all,

I got problem with Yealink IP phones (different models).
Firmware: 2.60.0.80 (newest from webpage)
All port on the router are forwarded:

5060 - SIP (UDP)
9000-9049 - RTP (UDP)

I configured phone using manual:
http://www.3cx.com/blog/voip-howto/remote-extensions/

PBX address - 188.121.18.xx
PBX is behind NAT with port forwarding. In the PBX we turn off STUN and set public address.

Trace from a call:
Code:
08:27:33.796  [MS105000] C:2.1: No RTP packets were received:remoteAddr=192.168.5.72:11780,extAddr=0.0.0.0:0,localAddr=188.121.18.xx:9002
08:27:32.781  [CM503008]: Call(2): Call is terminated
08:27:17.437  Session 155 of leg C:2.1 is confirmed
08:27:17.328  Currently active calls - 1: [2]
08:27:17.125  [CM503007]: Call(2): Device joined: sip:[email protected]:40600;rinstance=63ce595bc2c81674
08:27:17.125  [CM503007]: Call(2): Device joined: sip:[email protected]:5064
08:27:17.125  [MS210005] C:2.1:Answer provided. Connection(proxy mode):188.121.18.xx:9002(9003)
08:27:17.125  [MS210001] C:2.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40612(40613)
08:27:17.125  Remote SDP is set for legC:2.2
08:27:17.125  [CM505001]: Ext.99: Device info: Device Identified: [Man: 3CX Ltd.;Mod: Voice Mail Menu;Rev: 1] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Voice Mail Menu] PBX contact: [sip:[email protected]:5060]
08:27:17.125  [CM503002]: Call(2): Alerting sip:[email protected]:40600;rinstance=63ce595bc2c81674
08:27:16.968  [CM503025]: Call(2): Calling Ext:Ext.99@[Dev:sip:[email protected]:40600;rinstance=63ce595bc2c81674]
08:27:16.968  [MS210004] C:2.2:Offer provided. Connection(proxy mode): 127.0.0.1:7002(7003)
08:27:16.906  [CM503004]: Call(2): Route 1: Ext:Ext.99@[Dev:sip:[email protected]:40600;rinstance=63ce595bc2c81674]
08:27:16.906  [CM503010]: Making route(s) to <sip:[email protected]>
08:27:16.906  [MS210000] C:2.1:Offer received. RTP connection: 192.168.5.72:11780(11781)
08:27:16.906  Remote SDP is set for legC:2.1
08:27:16.906  [CM505001]: Ext.20: Device info: Device Identified: [Man: Yealink;Mod: T28;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, no-recvonly] UserAgent: [Yealink SIP-T28P 2.60.0.80] PBX contact: [sip:[email protected]:5060]
08:27:16.906  [CM503001]: Call(2): Incoming call from Ext.20 to <sip:[email protected]>
08:27:16.906  [CM500002]: Info on incoming INVITE:
  INVITE sip:[email protected] SIP/2.0
  Via: SIP/2.0/UDP 89.228.174.xx:5064;rport=5064;branch=z9hG4bK1375429119
  Max-Forwards: 70
  Contact: <sip:[email protected]:5064>
  To: <sip:[email protected]>
  From: "20"<sip:[email protected]>;tag=1517995330
  Call-ID: [email protected]
  CSeq: 2 INVITE
  Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
  Proxy-Authorization: Digest username="20",realm="3CXPhoneSystem",nonce="414d535c0377ead476:5709637020d6b99e648a26b53e45ada0",uri="sip:[email protected]",response="93dd385417830f15b2de08b9bc8fe299",algorithm=MD5
  Supported: replaces
  User-Agent: Yealink SIP-T28P 2.60.0.80
  Allow-Events: talk, hold, conference, refer, check-sync
  Content-Length: 0
We tried with 3CX Proxy Manger - its working but customers don't accept this solution.

Any help or clues are appreciated.

Thank you,
Tomasz

PS: if
 
is it a remote extension or is it an external caller that cannot get the audio?

It looks like a remote extension and if connecting directly (not through the tunnel) then you will need the "internal" rtp ports open which are in the 7000 udp range.
 
Hi Mark,

It's a remote extension connected directly to the pbx (the extension is login in to the PBX)
I just forward ports on router 7000-7500 UDP and still nothing:(

One more thing...
I use the instruction from release note SP5
http://www.3cx.com/blog/releases/3cx-version-9-service-pack-5/

Note (BETA): 3CX Users that want to register 3CXPhone for Android from an external location can modify a parameter in the custom parameters table of the 3CX Management console (Settings/Advanced/Custom Parameters) “ALLOWSOURCEASOUTBOUND”. Set this to 1 and you will receive 2 way audio upon successful registration. However this might have a negative effect on VoIP Providers so be careful to test this out properly on your environment before.

I changed “ALLOWSOURCEASOUTBOUND” to 1 and still nothing
 
Hi Tomasz

Can you post the screen shot (less full IPs and usernames) of the basic and advanced screens of a non working yealink.

The logs you post there show it is trying to send RTP to 192.168.5.xx and I suspect this is the lan on the remote extension not the 3cx lan.
 
Hi Mark,

You're right the ip address 192.168.5.xx is the ip in LAN of the external extension.

Basic conf
Link to image was deleted. Please attach it using "Upload Attachment"
Advanced conf
Link to image was deleted. Please attach it using "Upload Attachment"
 
I think it looks ok.

I have heard of problems with using the latest firmware 2.60.xxx can you downgrade to 2.50.0.50 (I think the new firmware sends the internal ip instead of the stun'd external ip) hence the RTP address is wrong.

Or you could try (and I am winging it here as I have no idea if this will work, but it is a 2 second change) to put the internal pbx ip in the Sip Server field, Enable the outbound Proxy Server and set the Outbound proxy server to the external IP of your pbx.

As I say, I have never tried it but we can see if it works.
 
If
Answer provided. Connection(proxy mode):188.121.18.xx:9002(9003)
is not blocked then ALLOWSOURCEASOUTBOUND should resolve this issue
If "border device" blocks traffic to PBX until PBX will send something to the address where from Yealink sends audio then packets from Yealink will not reach PBX and audio will not be available.

Please don't attach links to the pictures which are resided on third party servers and use Upload Attachment instead.

Thanks
 
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