Without seeing the whole config or the SIP trace I can't really say much about what is happening. If I were you I would backup my configs and start from scratch. My guess is that you changed something that you didn't mean to change. If you restart from a known I'll bet you can get it working again.<div>
<br></div><div>-MC<br><br><div class="gmail_quote">On Tue, Mar 8, 2011 at 10:00 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;">
Thanks,<br><br>Yes 192.168.2.190 is FS and I only have 1 polycom phone and rest are cisco 7960s .<br><br>I tried with g711alaw in third priority and its working, but i want to have the codec preference on polycom like g722,g711ulaw,g711alaw.<br>
<br>I have 2 debugs when i removed G711alaw and just kept G722 ,G711,G729 .<br><br>Made a call from polycom to cisco the call disconnected after pick up. <a href="http://pastebin.freeswitch.org/15621" target="_blank">http://pastebin.freeswitch.org/15621</a><br>
<br>Made an incoming call to polycom and it did not transcode. <a href="http://pastebin.freeswitch.org/15622" target="_blank">http://pastebin.freeswitch.org/15622</a><br><br>I am having this settings in profile:<br><param name="inbound-codec-negotiation" value="generous"/><br>
<param name="disable-transcoding" value="true"/><br>also tried :-<br><param name="inbound-codec-negotiation" value="generous"/><br>
<param name="disable-transcoding" value="false"/><br><br>Am i correct with those , if not do correct me.<br><br>Regards<br><font color="#888888">Sam</font><div><div></div><div class="h5"><br>
<br><br><br><br><div class="gmail_quote">On Wed, Mar 9, 2011 at 10:36 AM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex"><br><br><div class="gmail_quote"><div>On Sun, Mar 6, 2011 at 10:20 PM, Sam <span dir="ltr"><<a href="mailto:u2nsam@gmail.com" target="_blank">u2nsam@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);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><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:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex"><div><div></div><div><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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