no voice

[mckey415]

hi!

i have the following problem.
i'm using 3cx with stun server installed on the same machine. i'm trying to make a call through the internet from another lan sitting behind the nat. i'm using x-lite, the extension is amrked as external and bind to media server. i've got ports 5060, 3478, 7000-7500 opened. the call is establishing but none of the parties are hearing each other.

any suggestions on the problem?

 

[DaKhalli]

add a stune server to the xlite softphone configuration.

Solved the problems with my xlite.

 

[mckey415]

surely i added it.

 

[DaKhalli]

no result?

btw my exstension home aint marked external.

I use public stunserver in the 3cx box aswel asat home. Maybe its the fact your stunserver is locally on the same machine as 3cx.

For the xlite try to use a public stunserver

 

[SY]

Just search for "RFC STUN" in internet.
questions:
1. Why do you try to setup STUN server in local network?
2. Which kind of problems you did try to solve using such installation of STUN server?

Thanks,
Stepan

P.S. Hint: STUN server in local network causes no audio problem

 

[Borsoock]

Just search for "RFC STUN" in internet.
questions:
1. Why do you try to setup STUN server in local network?
2. Which kind of problems you did try to solve using such installation of STUN server?

Thanks,
Stepan

P.S. Hint: STUN server in local network causes no audio problem

Stephan- first post says his STUN server is for REMOTE clients sitting behind NAT . I've got exactly the same setup: 3cx and STUN on one machine and bunch of remote softphones, wifi phones, pap2t, spa1302 etc in several locations behind NATs. In my case everything works OK, I've got UDP 3479 opened too.

mckey415 - my suggestion is to make a test with firewall turned off to veryfi your install.

Thanks

Marcin

 

[SY]

Just search for "RFC STUN" in internet.
questions:
1. Why do you try to setup STUN server in local network?
2. Which kind of problems you did try to solve using such installation of STUN server?

Thanks,
Stepan

P.S. Hint: STUN server in local network causes no audio problem

Stephan- first post says his STUN server is for REMOTE clients sitting behind NAT . I've got exactly the same setup: 3cx and STUN on one machine and bunch of remote softphones, wifi phones, pap2t, spa1302 etc in several locations behind NATs. In my case everything works OK, I've got UDP 3479 opened too.

mckey415 - my suggestion is to make a test with firewall turned off to veryfi your install.

Thanks

Marcin

Sorry, probably I'm not right(about STUN)... (it is a real headache )

Other questions:
1. External extensions uses ports for external connections... Are they opened?
2. Is X-lite (external extension)configured to "Discover global address" with specified STUN server?

 

[Anonymous]

Sy, i do not think you far off actually.

STUN is good, and it is for SIP clients that are behind a NAT network. for this to work the STUN (I believe) needs to be on the edge of the NAT network. Hence you use a public STUN.

Having said that, there are people that have good results (i have not seen scientific facts) that installed a stun server on their local network and it appears that the SIP is working a lot better.

 

[mckey415]

thanks everyone for advices.

1. i use stun on the same machine as 3cx because 3cx is located in the dmz, wich is devided from my headquarters lan with isa server. so, to get softfones from headquarters lan to work i have to use stun on the same machine as 3cx server. on the other hand i have a plenty of remote offices, connected to the internet. there are small lans in every remote office, so they are sitting behind the nat. that's the reason of using stun. 3cx server has a real internet IP address, so there's no reason to use public stun for me.

2. yes, x-lite usus discover global address option.

3. actually i marked extension as external just to test it, i don't really understand what it means and what it changes, but i got exactly the same problem when this checkbox was cleared. can anyone please explain me what extension is external and bind to media server options mean?

 

[SY]

Hi mckey415,

There is a topic at the top:
Sticky: Include server status log, and equipment in use in posts!

Seems it is easier to post here log for problematic call, just copy/paste it from "Server Status" page. We will see what is correct and what is wrong.

Thanks,
Stepan

 

[SY]

Sy, i do not think you far off actually.

STUN is good, and it is for SIP clients that are behind a NAT network. for this to work the STUN (I believe) needs to be on the edge of the NAT network. Hence you use a public STUN.

Having said that, there are people that have good results (i have not seen scientific facts) that installed a stun server on their local network and it appears that the SIP is working a lot better.

Henk, I'm agree that STUN is in itself good... but please pay attention to how many people have problems related to STUN!

 

[Anonymous]

Yep, you not wrong mate.

I think STUN is a bit overrated and misunderstood in general.

Than again, a bit of good planning and network governance and you will find that most problems can be solved even without STUN.

 

[mckey415]

once again thanks for help
here's a piece of system status log file, showing the call from my extension to extension in the remote office

14:18:27.430|.\CallConf.cpp(95)|Log2|CallControl|CallConf:nIncoming:Call(C:46): Incoming call from Ext.0105 to sip:[email protected]
14:18:27.664|.\CallConf.cpp(152)|Log2|CallControl|CallConf:nProvisional:Call(C:46): got response from 3004
14:18:33.483|.\Call.cpp(260)|Log2|CallControl|CallLegImpl:nConnected:Call(C:46): Created audio channel for Ext.0105 (192.168.1.100:6270) with Media Server (x.x.x.x:7224)
14:18:33.483|.\CallStrategies.cpp(667)|Log2|CallManager|StratInOut:nConnected:Call(C:46): Setup completed for call from Ext.0105 to Ext.3004
14:18:33.499|.\Call.cpp(260)|Log2|CallControl|CallLegImpl:nConnected:Call(C:46): Created audio channel for Ext.3004 (x.x.x.x:61466) with Media Server (x.x.x.x:9000)
14:19:05.619|.\CallStrategies.cpp(262)|Log2|CallManager|StratLink:nHangUp:Call(C:46): Ext.0105 hung up call; reason: ???
14:19:05.619|.\SLServer.cpp(264)|Error1|MediaServer|MediaServerReporting::RTPReceiver:from 'sip:0/MediaServer':No RTP packets were received on 0000007A@:remoteAddr=y.y.y.y:61466,extAddr=x.x.x.x:9000,localAddr=0.0.0.0:9000

 

[SY]

Thanks for the log...

1. You are using build 1928, isn't it?
2. I cannot analyze correctness of IPs if it presented as x.x.x.x and y.y.y.y. Please specify at least first two parts of IPs.

You can send me log in private message, if you don't want to publish it here.

Thanks,
Stepan

 

[mujalli]

Hi friends,

I have same problem as mckey415 after instalation new version...
the call is establishing but none of the parties are hearing each other, also for a remote extension..that log always all the day as below:

16:15:30.069 ServRegs:nAdd [CM113002] Registered: Ext.105
16:15:28.210 ServRegs::checkExpiration [CM113000] Registration for sip:[email protected] has expired
16:14:33.741 ServRegs:nAdd [CM113002] Registered: Ext.105
16:14:32.069 ServRegs::checkExpiration [CM113000] Registration for sip:[email protected] has expired
16:13:37.600 ServRegs:nAdd [CM113002] Registered: Ext.105
16:13:36.194 ServRegs::checkExpiration [CM113000] Registration for sip:[email protected] has expired


When the extension is external and to use Bind to Media Server??
also more info about:
Which IP to use in 'Contact' field for registration: ???

below some of status log :
thanks for any help...



10:35:07.437 MediaServerReporting::RTPReceiver [MS105000] Call(16) Ext.105: No RTP packets were received on 0000001E@:remoteAddr=10.1.1.201:8000,extAddr=201.83.90.200:9000,localAddr=0.0.0.0:9000


16:01:29.694 CallLegImpl::initMSEndpoint [CM003001] Call(61): *!* Failed to create MediaServer endpoint for Ext.105

16:01:29.663 MediaServerReporting:ortMapper [MS006000] Call(61) Ext.102: No ports available to create EXTERNAL enpoint
16:01:25.725 MediaServerReporting:TMFhandler [MS211000] Call(60) Ext.102: DTMF (RTP) from 201.83.90.200:49188 arrived. in-band DTMF tone detection is turned off.

 

[nb]

Hi guys

The two options you can try to play about with in a problem like this is the "Support Replaces and Support Re-invite Options"

There are some phones which do not support these type of functions and therefore you can either not hear any audio at all or hear only 1 way audio the other end being dead when 2 such phones meet in a conversation.

Could you please tell me which phones are you using and the gateway in between?

So I would suggest that you try un ticking "support replaces" and keep bind to media server and support re-invite ticked.

If this does not work try unticking both options leaving bind to media server ticked only. Obviously if the phone is on an external network to the PBX you have to tick the external option.

 

[mujalli]

Thanks Nick...

I