Dropped calls on remote extension

[stobby256]

Hi All...

Got a Yealink T20P working over the internet to my 3CX , via the usual stuff.. firewall.. public IP etc. Call quality is rather good, and latency is also quite acceptable.

The firewall checker is of great help with this, but it does not tell you it nakes no attempt to start the services when you finish with it - naughty, naughty!! :twisted:

However, calls drop after around 20-30s everytime. Nothing seems to occur beforehand, they just... STOP :shock: No other internet rraffic going on at the time, speed measured around 6mb/s at both ends.

Help - where should I be looking?

Thanks,
Steve

 

[leejor]

That time, is about the time after which a call drops (times out) if an ACK is not received. Check the 3CX logs, they may confirm that.

 

[complex1]

Hi,

Have you enabled "UDP Keep-alive Message" in the phone?
For more information how to configure external exensions: http://www.3cx.com/blog/voip-howto/remote-extensions/

 

[stobby256]

Indeed..

Keep alives are configured as per your email link.

There do not appear to be any errors stating as to why it has been terminated, other than "user request" which it wasn't

21-Nov-2013 14:38:44.736 L:15.1[Extn]: Terminating targets, reason:
21-Nov-2013 14:38:44.736 Leg L:15.1[Extn] is terminated: Cause: BYE from PBX
21-Nov-2013 14:38:44.736 L:15.1[Extn] Sending: OnSendReq Send Req BYE from 0.0.0.0:0 tid=3522bb3d9a41295d [email protected]:
BYE sip:[email protected]:18846;transport=TCP SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-d8754z-3522bb3d9a41295d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:10208;transport=TCP>
To: "Tyron Wain"<sip:[email protected]>;tag=1789602946
From: <sip:[email protected]>;tag=ea5d000d
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0
21-Nov-2013 14:38:44.734 L:15.1[Extn]: Terminating targets, reason: SIP ;cause=200 ;text="Call terminated on user request"
21-Nov-2013 14:38:44.733 L:15.1[Extn] Sending: OnSendResp Send 200/INVITE from 0.0.0.0:0 tid=1728365051 [email protected]:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 31.24.217.195:18846;rport=18846;branch=z9hG4bK1728365051
Contact: <sip:[email protected]:10208;transport=TCP>
To: <sip:[email protected]>;tag=ea5d000d
From: "Tyron Wain"<sip:[email protected]>;tag=1789602946
Call-ID: [email protected]
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 12.0.32816.397 (32731)
Content-Length: 257

v=0
o=3cxPS 411796766720 439831494657 IN IP4 213.123.227.120
s=3cxPS Audio call
c=IN IP4 213.123.227.120
t=0 0
m=audio 12287 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtcp:12288
21-Nov-2013 14:38:40.608 L:15.1[Extn] Sending: OnSendResp Send 200/INVITE from 0.0.0.0:0 tid=1728365051 [email protected]:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 31.24.217.195:18846;rport=18846;branch=z9hG4bK1728365051
Contact: <sip:[email protected]:10208;transport=TCP>
To: <sip:[email protected]>;tag=ea5d000d
From: "Tyron Wain"<sip:[email protected]>;tag=1789602946
Call-ID: [email protected]
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhoneSystem 12.0.32816.397 (32731)
Content-Length: 257

v=0
o=3cxPS 411796766720 439831494657 IN IP4 213.123.227.120
s=3cxPS Audio call
c=IN IP4 213.123.227.120
t=0 0
m=audio 12287 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtcp:12288
21-Nov-2013 14:38:36.555 L:15.1[Extn] Sending: OnSendResp Send 200/INVITE from 0.0.0.0:0 tid=1728365051 [email protected]:

 

[leejor]

It may be the set, that is not receiving an ACK and dropping the call.

Contact: <sip:[email protected]:10208;transport=TCP>

The port numbers being used are "interesting" and suggest that there may be port substitution happening by one of the routers. 3CX may be sending SIP messages that are not reaching the intended target. What type of routers are being used at each end? Some require changes to handle SIP correctly.

 

[stobby256]

At the 3CX end, I have a Zyxel router, with port forwarding, into BT Broadband.

At the Remote phone end, I have no control over it... It is part of a business park with some managed IT system. I can get hold of their IT guys to see if there's any port translation..

I did try this....

Blocked port 5060 at the 3CX end and Telnet'd into the 3CX server.. obviously no joy. Opened up 5060, and a successful Telnet took place .. may sound a bit whacky but at least I know 5060 works from end to end.

 

[GManNAtl]

If you have no control over the firewall at the remote end you may want to try the 3CX Proxy application. This could help with the NAT Traversal at the far end.

 

[jpillow]

under extensions--->other do you have the "pbx delivers auido" box checked?

 

[bardissi]

Is it just one phone on the remote side?

Did you try stun?

 

[ian.watts]

Curious.. have you configured the phone as a remote extension?
If so.. consider the far end may have SIP ALG on their firewall/router.. and don't configure it as a full "remote" extension but just employ the PBX address/host name without all the STUN, NAT, RPORT, etc.. I find that some router SIP ALGs work well and fine.. and some don't. Similar, some phones will traverse them okay.. others not so well.

Last one of my own was a Cisco SPA504G at home behind my Craptiontec the FIOS guy left me a year or two ago (never bothered to replace it..). I have configured a bazillion of these for remote extensions.. yet behind that router it would get one-way audio.. finally just turned it on and turned off the NAT-T and keepalive.. worked fine since.

This morning.. started fussing with a Yealink T32 which wouldn't provision at the office on Friday. Got it to provision.. forgot I left the extension to provision the LAN address, changed it on the config, registered and works fine. No NAT, no nothing.

Usually a 30-second drop does indicate an edge router/firewall problem, though. If they have that SIP ALG on.. it may be fighting with your remote config.. so perhaps a standard config with the remote PBX address will do the job.