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?
The issue was fixed by correcting the RTP packet size on the phone from 0.030 to 0.020.
lisavision,
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" folderPress Alt, click View, and then click Open FTP Site in Windows Explorer
Drag and drop files from PC into the Internet Explorer window
Thanks,
Matt
The issue was fixed by correcting the RTP packet size on the phone from 0.030 to 0.020.
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=10.10.20.1:5060 dst=10.10.20.2:5060
14:59:49.268 SIP.STACK MSG INVITE sip:2007@10.10.20.2:5060 SIP/2.0
14:59:49.268 SIP.STACK MSG From: "Joe Smith" <sip:2040@10.10.20.1:5060;transport=UDP>;tag=540fcb0-a0a1401-13c4-ac0ef-db27174c-ac0ef
14:59:49.268 SIP.STACK MSG To: "Site Supervisor" <sip:2007@10.10.20.1:5060>
14:59:49.268 SIP.STACK MSG Call-ID: 5467158-a0a1401-13c4-ac0ef-cb7e9108-ac0ef@10.10.20.1
14:59:49.268 SIP.STACK MSG CSeq: 1 INVITE
14:59:49.269 SIP.STACK MSG Via: SIP/2.0/UDP 10.10.20.1:5060;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:10.10.20.1:5060;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 10.10.20.3
14:59:49.271 SIP.STACK MSG s=SIP Call
14:59:49.271 SIP.STACK MSG c=IN IP4 10.10.20.3
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 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.
Thanks,
Matt