3 Replies Latest reply on Oct 16, 2012 1:16 PM by matt

    Soundstation 2w Wireless Conference phone

    lisavision New Member

      I have a Soundstation 2w that when a call is made from this phone dialing to a Adtran 706 or 712, the call goes directly to voicemail. If I dial from the Soundstation 2w to a Polycom 601 it works just fine. Any suggestions where I should look?

        • Re: Soundstation 2w Wireless Conference phone



          I am assuming all of these phones are off of the same unit.  If that is not correct please let me know.   If it is a NetVanta 7100 (or other AOS voice product)  I would need to see the current configuration of the unit and the output from a debug sip stack messages and debug voice verbose while the problem is recreated.  If it is on a UC server I would need to see a packet capture from the server and possibly the swlog.txt file that corresponds.  If you do not have any private sensitive information (public IPs, passwords, etc) you can attach this information in a reply.  If you do have sensitive information in the configurations and debugs you can either strip it out before attaching it or you can upload it to our FTP server from the instructions below.  I would need to know the exact file name(s) if you upload them to the FTP server.


          Open Internet Explorer web browser
          Type the following URL:  ftp://ftp.adtran.com
          Double-click the "Incoming" folder

          Press Alt, click View, and then click Open FTP Site in Windows Explorer

          Drag and drop files from PC into the Internet Explorer window




            • Re: Soundstation 2w Wireless Conference phone
              lisavision New Member

              The issue was fixed by correcting the RTP packet size on the phone from 0.030 to 0.020.

                • Re: Soundstation 2w Wireless Conference phone

                  Thanks for posting the resolution.  Just to give some additional information to anyone else who comes across this post, this is related to the ptime sent in the SDP.  It would look like the portion highlighted in red below in a debug sip stack messages:


                  14:59:49.268 SIP.STACK MSG     Tx: UDP src= dst=

                  14:59:49.268 SIP.STACK MSG         INVITE sip:2007@ SIP/2.0

                  14:59:49.268 SIP.STACK MSG         From: "Joe Smith" <sip:2040@;transport=UDP>;tag=540fcb0-a0a1401-13c4-ac0ef-db27174c-ac0ef

                  14:59:49.268 SIP.STACK MSG         To: "Site Supervisor" <sip:2007@>

                  14:59:49.268 SIP.STACK MSG         Call-ID: 5467158-a0a1401-13c4-ac0ef-cb7e9108-ac0ef@

                  14:59:49.268 SIP.STACK MSG         CSeq: 1 INVITE

                  14:59:49.269 SIP.STACK MSG         Via: SIP/2.0/UDP;branch=z9hG4bK-ac0ef-2a01a7de-cac560e

                  14:59:49.269 SIP.STACK MSG         Max-Forwards: 70

                  14:59:49.269 SIP.STACK MSG         Supported: 100rel,replaces

                  14:59:49.269 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

                  14:59:49.269 SIP.STACK MSG         User-Agent: ADTRAN_NetVanta_7100/R10.3.1.E

                  14:59:49.269 SIP.STACK MSG         Contact: <sip:;transport=UDP>

                  14:59:49.270 SIP.STACK MSG         Content-Type: application/sdp

                  14:59:49.270 SIP.STACK MSG         Content-Length: 284

                  14:59:49.270 SIP.STACK MSG

                  14:59:49.270 SIP.STACK MSG         v=0

                  14:59:49.270 SIP.STACK MSG         o=MxSIP 0 643433883 IN IP4

                  14:59:49.271 SIP.STACK MSG         s=SIP Call

                  14:59:49.271 SIP.STACK MSG         c=IN IP4

                  14:59:49.271 SIP.STACK MSG         t=0 0

                  14:59:49.271 SIP.STACK MSG         m=audio 3000 RTP/AVP 0 18 101

                  14:59:49.271 SIP.STACK MSG         a=ptime:20

                  14:59:49.271 SIP.STACK MSG         a=sendrecv

                  14:59:49.271 SIP.STACK MSG         a=silenceSupp:off - - - -

                  14:59:49.272 SIP.STACK MSG         a=rtpmap:0 PCMU/8000

                  14:59:49.272 SIP.STACK MSG         a=rtpmap:18 G729/8000

                  14:59:49.272 SIP.STACK MSG         a=fmtp:18 annexb=no

                  14:59:49.272 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000

                  14:59:49.272 SIP.STACK MSG         a=fmtp:101 0-15


                  ADTRAN IP 700 phones do not support anything other than a ptime of 20 ms.  This is denoted in the IP 700 Series 2.2.0 (2.2.0-T3) Release Notes as follows:


                  Calls with packetization periods other than 20ms will be disconnected by the phone with a BYE response, as only 20ms packetization periods are currently supported.