Thanks,
Yes 192.168.2.190 is FS and I only have 1 polycom phone and rest are cisco 7960s .
I tried with g711alaw in third priority and its working, but i want to have the codec preference on polycom like g722,g711ulaw,g711alaw.
I have 2 debugs when i removed G711alaw and just kept G722 ,G711,G729 .
Made a call from polycom to cisco the call disconnected after pick up. http://pastebin.freeswitch.org/15621
Made an incoming call to polycom and it did not transcode. http://pastebin.freeswitch.org/15622
I am having this settings in profile:
<param name="inbound-codec-negotiation" value="generous"/>
<param name="disable-transcoding" value="true"/>
also tried :-
<param name="inbound-codec-negotiation" value="generous"/>
<param name="disable-transcoding" value="false"/>
Am i correct with those , if not do correct me.
Regards
Sam
On Wed, Mar 9, 2011 at 10:36 AM, Michael Collins <msc@freeswitch.org> wrote:
On Sun, Mar 6, 2011 at 10:20 PM, Sam <u2nsam@gmail.com> wrote:
Hello,
The extension 7006 was working few days ago and now it not working,rest all the phone are working properly.Sam,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.-MC_______________________________________________
U 192.168.2.190:5060 -> 192.168.2.14:5060
INVITE sip:7006@192.168.2.14:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..Max-Forwards: 70..From: "" <sip:9322273640@192.168.2.190>;tag=4QggZ1XcNZD3F..To: <sip:7006@192.168.2.14:5060>..Call-ID: 7bf
ae496-c324-122e-bc85-0026b976aef7..CSeq: 9403059 INVITE..Contact: <sip:mod_sofia@192.168.2.190:5060>..User-Agent: NOVANET..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
.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
position: session..Content-Length: 203..X-FS-Support: update_display..Remote-Party-ID: <sip:9322273640@192.168.2.190>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1299446773 1299446774 IN IP4 192.168.2.190..s=FreeSWITCH
..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..
U 192.168.2.14:5060 -> 192.168.2.190:5060
SIP/2.0 488 Not Acceptable Here..Via: SIP/2.0/UDP 192.168.2.190;rport;branch=z9hG4bK8mmBB5e8Z2mKr..From: <sip:9322273640@192.168.2.190>;tag=4QggZ1XcNZD3F..To: "7006" <sip:7006@192.168.2.14:5060>;tag=983B5A59-EB953B22..CSeq: 9403059
INVITE..Call-ID: 7bfae496-c324-122e-bc85-0026b976aef7..Contact: <sip:7006@192.168.2.14:5060>..User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.2.2.0477..Accept-Language: en..Warning: 488 SDP "NotAcceptableHere"..Content-Length: 0....
I tried all the possibilities with codec priorities on U/A .
http://pastebin.freeswitch.org/15577
Any help.
Regards
Sam
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