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N720 with Gigaset SL750h handsets having audio issues after SP6 update

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Tony Aerts

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We have a N720 DM with 3 Base stations installed at one of our customers and until SP6
we didn't have any issues with audio. Everything was working fine till then. This weekend
the update to SP6 was done and since monday issues with audio on the Handsets.

We haven't changed a single thing in the configuration. The handsets have PBX delivers audio on.
No Stun, no SBC or any of the know limitations specified on https://www.3cx.com/sip-phones/gigaset-n720/

We can't seem to pinpoint the problem. A Server reboot resolves the problem temporary,
but we'll have the customer back on the line in a couple of hours. Anyone can help or point us in the right direction. Here is the verbose log of one of the calls made:

09/26/2018 11:56:27 AM - L:1243.2[Extn:22] got Terminated Recv 487/INVITE from 81.82.217.75:5060 tid=433b68027c7e5044 Call-ID=fb3B3FFbZND-ZBBVL3je1w..: SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.110.68:5060;branch=z9hG4bK-524287-1---433b68027c7e5044;rport=5060;received=84.241.191.236 Contact: <sip:[email protected]:5060> To: <sip:[email protected]>;tag=1399415395 From: "Balie:0485763413" <sip:[email protected]:5060;nf=q>;tag=d76f0524 Call-ID: fb3B3FFbZND-ZBBVL3je1w.. CSeq: 1 INVITE User-Agent: Gigaset N720 DM PRO/70.111.00.000.000 Content-Length: 0
09/26/2018 11:56:27 AM - L:1243.2[Extn:22] got Failure: Failure Recv 487/INVITE from 81.82.217.75:5060 tid=433b68027c7e5044 Call-ID=fb3B3FFbZND-ZBBVL3je1w..: SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.110.68:5060;branch=z9hG4bK-524287-1---433b68027c7e5044;rport=5060;received=84.241.191.236 Contact: <sip:[email protected]:5060> To: <sip:[email protected]>;tag=1399415395 From: "Balie:0485763413" <sip:[email protected]:5060;nf=q>;tag=d76f0524 Call-ID: fb3B3FFbZND-ZBBVL3je1w.. CSeq: 1 INVITE User-Agent: Gigaset N720 DM PRO/70.111.00.000.000 Content-Length: 0
09/26/2018 11:56:27 AM - Session 87115 has failed in leg L:1243.2[Extn:22] ; Cause: 487 Request Cancelled/INVITE from 81.82.217.75:5060
09/26/2018 11:56:26 AM - Stop call record for leg L:1243.2[Extn:22]
09/26/2018 11:56:26 AM - Removing leg L:1243.2[Extn:22]
09/26/2018 11:56:26 AM - L:1243.2[Extn:22]: Terminating targets, reason: SIP ;cause=200 ;text="Call terminated on user request"
09/26/2018 11:56:26 AM - Call(C:1243), Extn:22 on exit: DlgInfo(1243-3522/Terminated / R)
09/26/2018 11:56:26 AM - Call(C:1243), Extn:22 on entry: DlgInfo(1243-3522/Early / R)
09/26/2018 11:56:26 AM - Notify dialog-info: Extn:22: sip:[email protected]:5060, Call(C:1243)
09/26/2018 11:56:20 AM - Provisional response arrived for session 87115 of Leg L:1243.2[Extn:22]
09/26/2018 11:56:20 AM - L:1243.2[Extn:22] got Provisional Recv 180/INVITE from 81.82.217.75:5060 tid=433b68027c7e5044 Call-ID=fb3B3FFbZND-ZBBVL3je1w..: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.110.68:5060;branch=z9hG4bK-524287-1---433b68027c7e5044;rport=5060;received=84.241.191.236 Contact: <sip:[email protected]:5060> To: <sip:[email protected]>;tag=1399415395 From: "Balie:0485763413" <sip:[email protected]:5060;nf=q>;tag=d76f0524 Call-ID: fb3B3FFbZND-ZBBVL3je1w.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE User-Agent: Gigaset N720 DM PRO/70.111.00.000.000 Allow-Events: message-summary, refer, ua-profile, check-sync Content-Length: 0
09/26/2018 11:56:20 AM - [CM505001]: Endpoint Extn:22: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Gigaset N720 DM PRO/70.111.00.000.000] PBX contact: [sip:[email protected]:5060]
09/26/2018 11:56:20 AM - [CM503002]: Call(C:1243): Alerting Extn:22 by contact <sip:[email protected]:5060>
09/26/2018 11:56:20 AM - Call(C:1243), Extn:22 on exit: DlgInfo(1243-3522/Early / R)
09/26/2018 11:56:20 AM - Call(C:1243), Extn:22 on entry: DlgInfo(1243-3522/Initial / R)
09/26/2018 11:56:20 AM - Notify dialog-info: Extn:22: sip:[email protected]:5060, Call(C:1243)
09/26/2018 11:56:20 AM - UacSession 87115 has formed leg L:1243.2[Extn:22]
09/26/2018 11:56:19 AM - [CM503025]: Call(C:1243): Calling T:Extn:22@[Dev:sip:[email protected]:5060] for L:1243.1[Queue:85]
09/26/2018 11:56:19 AM - Route to L:1243.2[Extn:22] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=753ec80b52398234 Call-ID=fb3B3FFbZND-ZBBVL3je1w..: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---753ec80b52398234;rport Max-Forwards: 70 Contact: <sip:[email protected]:5060> To: <sip:[email protected]> From: "Balie:0485763413"<sip:[email protected]:5060;nf=q>;tag=d76f0524 Call-ID: fb3B3FFbZND-ZBBVL3je1w.. CSeq: 1 INVITE Alert-Info: <http://www.notused.invalidtld>;info=queue Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer Content-Length: 420 v=0 o=3cxPS 360626257920 89422561281 IN IP4 84.241.191.236 s=3cxPS Audio call c=IN IP4 84.241.191.236 t=0 0 m=audio 9064 RTP/AVP 3 18 9 0 8 112 101 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 opus/48000/2 a=fmtp:112 maxplaybackrate=16000;sprop-maxcapturerate=16000 a=rtpmap:101 telephone-event/8000 a=sendrecv
09/26/2018 11:56:19 AM - L:1243.2[Extn:22]: SLA slot is acquired: 22#0 'idle'
09/26/2018 11:56:19 AM - Added leg L:1243.2[Extn:22]
09/26/2018 11:56:19 AM - [Flow] Call(C:1243): making call from L:1243.1[Queue:85] to T:Extn:22@[Dev:sip:[email protected]:5060]
 
What exactly are the "audio issues" you are having? No Audio, choppy audio, audio quality?

Also I just feel that this may be related to the audio port expansion in V15.5 SP6, FYI, the External Ports of 3CX have expanded to 9000-10999, and the internal port range used by 3CX is now 7000-8499.
So what I am trying to say, run the firewall checker to make sure it still passes, and if not make sure the new ports are opened on the firewall.
 
It's on inbound and outbound calls that the people can't hear eachother. I did the Firewall check and indead didn't get passed on some of the new ports, so edited the firewall rules and now the Firewall check is passed.

I'll keep this topic updated if that doesn't resolve the issue.
 
OK, is the problem only with external calls, or do you have the same problem on calls extension-to-extension?

If it is only external calls, then expanding the ports on your firewall will probably do it.
 
It's on both internal and external calls.
 
By the time you have "PBX Delivers Audio" enabled for all extensions like you said initially, all audio traffic should be going via the 3CX Server. Would you happen to have the 3CX Server on a different VLAN as the Extensions? Or over a VPN?
If yes and you have enabled certain ports only to be able to communicate over the VLAN, you may have to expand those as well. As I said, internal ports also changed from 7000-7499 to 7000-8499.
 
Not using any VLAN.
 
Then the only other thing I can think to do is to start a packet capture on the 3CX Server, then make a call, then open the capture file in Wireshark and see where the audio is going.
 
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