[Freeswitch-users] incompatible destination

[Sam]

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 at freeswitch.org> wrote:

>
>
> On Sun, Mar 6, 2011 at 10:20 PM, Sam <u2nsam at 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 at 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 at 192.168.2.190>;tag=4QggZ1XcNZD3F..To: <
>> sip:7006 at 192.168.2.14:5060>..Call-ID: 7bf
>>   ae496-c324-122e-bc85-0026b976aef7..CSeq: 9403059 INVITE..Contact: <
>> sip:mod_sofia at 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 at 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 at 192.168.2.190>;tag=4QggZ1XcNZD3F..To: "7006" <
>> sip:7006 at 192.168.2.14:5060>;tag=983B5A59-EB953B22..CSeq: 9403059
>>   INVITE..Call-ID: 7bfae496-c324-122e-bc85-0026b976aef7..Contact: <
>> sip:7006 at 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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>
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