One way audio on HT-503 grandstream and 3CX VoIP Client

[dnwklin]

One way audio on HT-486 grandstream. When I use Grandstream/X-Lite to place a call, only the other party can hear me. I cannot hear any sound? What is the problem? Thanks
{UPDATED}

log: (Called by HT-503 & Close the STUN)

12:40:28.640 [CM503008]: Call(9): Call is terminated
12:40:28.640 [CM503008]: Call(9): Call is terminated

12:40:20.203 Session 214 of leg C:9.1 is confirmed

12:40:19.593 [CM503007]: Call(9): Device joined: sip:[email protected]:5060

12:40:19.593 [CM503007]: Call(9): Device joined: sip:[email protected]:5060;user=phone

12:40:19.578 [MS210001] C:9.2:Answer received. RTP connection: 204.11.192.36:52460(52461)

12:40:19.578 Remote SDP is set for legC:9.2

12:40:09.828 [MS210003] C:9.1:Answer provided. Connection(transcoding mode):221.195.40.28:9002(9003)

12:40:09.828 [MS210001] C:9.2:Answer received. RTP connection: 204.11.192.36:52460(52461)

12:40:09.828 [CM505003]: Provider:[CC] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:221.195.40.28:5060]

12:40:09.812 Remote SDP is set for legC:9.2

12:40:09.812 [CM503002]: Call(9): Alerting sip:[email protected]:5060

12:40:06.984 [MS210002] C:9.2:Offer provided. Connection(transcoding mode): 221.195.40.28:9004(9005)

12:40:06.968 [CM503004]: Call(9): Calling: VoIPline:15186890550@(Ln.10004@CC)@[Dev:sip:[email protected]:5060]

12:40:06.921 [CM503010]: Making route(s) to <sip:[email protected];user=phone>

12:40:06.921 [MS210000] C:9.1:Offer received. RTP connection: 192.168.1.114:5004(5005)

12:40:06.921 Remote SDP is set for legC:9.1

12:40:06.921 [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream HT-503 1.0.0.5] Transport: [sip:221.195.40.28:5060]

12:40:06.921 [CM503001]: Call(9): Incoming call from Ext.100 to <sip:[email protected];user=phone>
12:40:06.906 [CM500002]: Info on incoming INVITE:

INVITE sip:[email protected];user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK369511425;rport=63977;received=166.217.173.232

Max-Forwards: 70

Contact: <sip:[email protected]:5060;user=phone>

To: <sip:[email protected];user=phone>

From: <sip:[email protected];user=phone>;tag=354981241

Call-ID: [email protected]

CSeq: 21 INVITE

Accept: application/sdp, application/dtmf-relay

Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE

Proxy-Authorization: Digest username="100",realm="3CXPhoneSystem",nonce="12878368806:aa21f5229114316d362d0d1c9c095be0",uri="sip:[email protected];user=phone",response="d86feff853759b3c49bb5bb0fc2c060e",algorithm=MD5

Supported: replaces, path

User-Agent: Grandstream HT-503 1.0.0.5

Content-Length: 0

 

[William400]

Re: One way audio on HT-486 grandstream

Hi

Can you please give more information on your setup. Is XLite on the same LAN as the 3CX server or on a separate network?

If on the same network please ensure that STUN is disabled. STUN needs to be enabled only if the phone is external to the 3CX server network. see http://www.3cx.com/sip-phones/x-lite.html

 

[dnwklin]

Re: One way audio on HT-486 grandstream

I had already disable the stun and all firewalls. The HT-486 and x-lite are on the same LAN. I just started to use the system, I am not sure what kinds of information should I also posted here. Could someone give me more specific instructions? Thank you

 

[Vali_3CX]

Re: One way audio on HT-486 grandstream

Hi
From the phone(s) which has no incoming audio, try first dial 999 and see if you hear it (sounds nice "see" when about "hearing" :mrgreen: )

 

[SY]

Re: One way audio on HT-486 grandstream

I had already disable the stun and all firewalls. The HT-486 and x-lite are on the same LAN. I just started to use the system, I am not sure what kinds of information should I also posted here. Could someone give me more specific instructions? Thank you

"Forum rules - Read to get answers" (It is the second line in the "TOPICS" part)
direct link is http://www.3cx.com/forums/forum-rules-read-to-get-answers-93.html
Picture attached.

Thanks

P.S. Verbose level version of 3CXMediaServer.trace.log may highlight problem.

 

[dnwklin]

Re: One way audio on HT-486 grandstream

The following is the log.

12:14:05.625 [CM503008]: Call(2): Call is terminated

12:14:05.625 [CM503008]: Call(2): Call is terminated

12:14:04.468 [MS210002] C:2.1:Offer provided. Connection(transcoding mode): 221.195.40.28:9002(9003)

12:14:04.468 [MS210000] C:2.2:Offer received. RTP connection: 127.0.0.1:40610(40611)

12:14:04.468 Remote SDP is set for legC:2.2

12:13:52.781 Session 28 of leg C:2.1 is confirmed

12:13:52.109 [CM503007]: Call(2): Device joined: sip:[email protected]:40600;rinstance=8fc012626ce79c33

12:13:52.109 [CM503007]: Call(2): Device joined: sip:[email protected]:5060;user=phone

12:13:52.109 [MS210005] C:2.1:Answer provided. Connection(proxy mode):221.195.40.28:9002(9003)

12:13:52.109 [MS210001] C:2.2:Answer received. RTP connection: 127.0.0.1:40610(40611)

12:13:52.093 Remote SDP is set for legC:2.2

12:13:52.093 [CM505001]: Ext.800: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] Transport: [sip:127.0.0.1:5060]

12:13:52.093 [CM503002]: Call(2): Alerting sip:[email protected]:40600;rinstance=8fc012626ce79c33

12:13:51.656 [MS210004] C:2.2:Offer provided. Connection(proxy mode): 127.0.0.1:7000(7001)

12:13:51.656 [CM503004]: Call(2): Calling: Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=8fc012626ce79c33]

12:13:51.656 [MS210000] C:2.1:Offer received. RTP connection: 192.168.1.114:5004(5005)

12:13:51.656 [CM503010]: Making route(s) to <sip:[email protected];user=phone>

12:13:51.640 Remote SDP is set for legC:2.1

12:13:51.640 [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream HT-503 1.0.0.5] Transport: [sip:221.195.40.28:5060]

12:13:51.640 [CM503001]: Call(2): Incoming call from Ext.100 to <sip:[email protected];user=phone>

12:13:51.343 [CM500002]: Info on incoming INVITE:

INVITE sip:[email protected];user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK1789676156;rport=63603;received=166.217.173.232

Max-Forwards: 70

Contact: <sip:[email protected]:5060;user=phone>

To: <sip:[email protected];user=phone>

From: <sip:[email protected];user=phone>;tag=550178675

Call-ID: [email protected]

CSeq: 51 INVITE

Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE

Proxy-Authorization: Digest username="100",realm="3CXPhoneSystem",nonce="12878367229:a519a64b199595357ef4305225775912",uri="sip:[email protected];user=phone",response="7510da9a7eaec49791c3622ec5536083",algorithm=MD5

Supported: replaces, path

User-Agent: Grandstream HT-503 1.0.0.5

Content-Length: 0