Author Topic: GXW4104 and Skype for Business Online Cloud Connector  (Read 94 times)

rfischman

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GXW4104 and Skype for Business Online Cloud Connector
« on: October 02, 2017, 05:52:16 PM »
I'm attempting to configure a GXW4104 to communicate with a mediation server in a skype for business online hybrid environment using the Cloud Connector Edition.
In my Cloud Connector config file, I specified TCP Port 5061 for the Gateway device.

I'm trying to figure out how to get the GXW4104 configured correctly...  I found lots of examples for Astreisk and FreePBX, all of which seem to have userID and Password listed in their configs.  The Cloud Connector configuration doesn't seem to provide this capability.  So how do I configure the GXW4104 to connect to the SIP trunk without specifying username and password?

here's what I've done so far:
On the Accounts Page, I set Account 1 active and set 2 and 3 NOT active.
SIP Server:  <IP Address of my Mediation Server>
SIP Settings: Registration is set to NO

Under Settings -> Channnels
Local Listen Port: ch1-4:5061++;

under VoIP:
User ID ch1-4:100;
SIP Server: ch1-4:p1;
SIP Destination port: ch1-4:5068;

FXO Lines
Dialing:
Stage Method: ch1-4:1;

5068 is the TCP port on the mediation server.

I'm seeing log files that look like this in my syslog:

10-02-2017   20:40:24   User.Debug   192.168.3.230   GS_LOG: [00:0B:82:B4:3A:2B][000][9660000323A][1.4.1.5] 1602 sip_transport.c Sess: 8 sent: SIP/2.0 200 OK  Via: SIP/2.0/TCP 192.168.3.212:53333;branch=z9hG4bKf3a7ded3  From: <sip:MedServer.rac.local:5068;transport=Tcp;ms-opaque=55d5e6f1e7cb2d5b>;epid=25F3C43641;tag=7bf89ea4ac  To: <sip:192.168.3.230>;tag=0053d676b1f53fc0  Call-ID: 9e3ee13fbffc46319e7e39ebf1d7fc6e  CSeq: 2493 OPTIONS  User-Agent: Grandstream GXW4104 (HW 2.3, Ch:8) 1.4.1.5  Contact: <sip:192.168.3.230:5061;transport=tcp>  Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK  Supported: replaces, timer, 100rel, path  Content-Length: 0   
10-02-2017   20:40:24   User.Debug   192.168.3.230   GS_LOG: [00:0B:82:B4:3A:2B][000][9660000323A][1.4.1.5] SIPTransmit_Go() using direct_sending_handle. Message number: 200
10-02-2017   20:40:24   User.Debug   192.168.3.230   GS_LOG: [00:0B:82:B4:3A:2B][000][9660000323A][1.4.1.5] TCP Handle Warning, a new handle accepted7
10-02-2017   20:40:24   User.Debug   192.168.3.230   GS_LOG: [00:0B:82:B4:3A:2B][000][9660000323A][1.4.1.5] 1324 sip.c Acct:0 h-port:5061 Received SIP message: OPTIONS sip:192.168.3.230 SIP/2.0  FROM: <sip:MedServer.rac.local:5068;transport=Tcp;ms-opaque=55d5e6f1e7cb2d5b>;epid=25F3C43641;tag=7bf89ea4ac  TO: <sip:192.168.3.230>  CSEQ: 2493 OPTIONS  CALL-ID: 9e3ee13fbffc46319e7e39ebf1d7fc6e  MAX-FORWARDS: 70  VIA: SIP/2.0/TCP 192.168.3.212:53333;branch=z9hG4bKf3a7ded3  CONTACT: <sip:MedServer.rac.local:5068;transport=Tcp;maddr=192.168.3.212>  CONTENT-LENGTH: 0  USER-AGENT: RTCC/6.0.0.0 CCE-MediationServer   

Unfortunately, calls don't seem to route outbound - my Skype clients report "user not found" errors, and when I try inbound calls, the GXW4101 never answers the call.

Any help is appreciated!