[Freeswitch-users] incompatible destination

[Michael Collins]

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.

-MC

On Tue, Mar 8, 2011 at 10:00 PM, Sam <u2nsam at gmail.com> wrote:

> 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
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110309/56fec3d7/attachment.html