<br><br><div class="gmail_quote">On Sun, Mar 6, 2011 at 10:20 PM, Sam <span dir="ltr"><<a href="mailto:u2nsam@gmail.com">u2nsam@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Hello,<br><br><br>The extension 7006 was working few days ago and now it not working,rest all the phone are working properly.<br></blockquote><div><br></div><div>Sam,</div><div><br></div><div>Is 192.168.2.190 your FreeSWITCH box? If so then the issue is with the phone not accepting the PCMA codec. Go into your Polycom config and make sure that PCMA is enabled on the phone itself. The other option would be to let the Polycom speak in G722 and let FS do transcoding. In any case, if I were you I would find another working Polycom phone and mimic the configs on this non-working Poly 335.</div>
<div><br></div><div>-MC</div><div><br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><br><br>U <a href="http://192.168.2.190:5060" target="_blank">192.168.2.190:5060</a> -> <a href="http://192.168.2.14:5060" target="_blank">192.168.2.14:5060</a><br>
INVITE <a href="http://sip:7006@192.168.2.14:5060" target="_blank">sip:7006@192.168.2.14:5060</a> SIP/2.0..Via: SIP/2.0/UDP 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..Max-Forwards: 70..From: "" <<a href="mailto:sip%3A9322273640@192.168.2.190" target="_blank">sip:9322273640@192.168.2.190</a>>;tag=4QggZ1XcNZD3F..To: <<a href="http://sip:7006@192.168.2.14:5060" target="_blank">sip:7006@192.168.2.14:5060</a>>..Call-ID: 7bf<br>
ae496-c324-122e-bc85-0026b976aef7..CSeq: 9403059 INVITE..Contact: <<a href="http://sip:mod_sofia@192.168.2.190:5060" target="_blank">sip:mod_sofia@192.168.2.190:5060</a>>..User-Agent: NOVANET..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.<br>
.Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Content-Type: application/sdp..Content-Dis<br>
position: session..Content-Length: 203..X-FS-Support: update_display..Remote-Party-ID: <<a href="mailto:sip%3A9322273640@192.168.2.190" target="_blank">sip:9322273640@192.168.2.190</a>>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1299446773 1299446774 IN IP4 192.168.2.190..s=FreeSWITCH<br>
..c=IN IP4 192.168.2.190..t=0 0..m=audio 31474 RTP/AVP 8 101 13..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ptime:20..<br><br>U <a href="http://192.168.2.14:5060" target="_blank">192.168.2.14:5060</a> -> <a href="http://192.168.2.190:5060" target="_blank">192.168.2.190:5060</a><br>
SIP/2.0 488 Not Acceptable Here..Via: SIP/2.0/UDP 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..From: <<a href="mailto:sip%3A9322273640@192.168.2.190" target="_blank">sip:9322273640@192.168.2.190</a>>;tag=4QggZ1XcNZD3F..To: "7006" <<a href="http://sip:7006@192.168.2.14:5060" target="_blank">sip:7006@192.168.2.14:5060</a>>;tag=983B5A59-EB953B22..CSeq: 9403059<br>
INVITE..Call-ID: 7bfae496-c324-122e-bc85-0026b976aef7..Contact: <<a href="http://sip:7006@192.168.2.14:5060" target="_blank">sip:7006@192.168.2.14:5060</a>>..User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.2.2.0477..Accept-Language: en..Warning: 488 SDP "NotAcceptableHere"..Content-Length: 0....<br>
<br><br><br>I tried all the possibilities with codec priorities on U/A .<br><br><a href="http://pastebin.freeswitch.org/15577" target="_blank">http://pastebin.freeswitch.org/15577</a><br><br>Any help.<br><br>Regards<br><font color="#888888">Sam<br>
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