proxy-media mode is still in the media path.<br>bypass-media is the one where you are not in the media path.<br><br><br><div class="gmail_quote">On Thu, Nov 6, 2008 at 9:30 AM, shehzad p <span dir="ltr"><<a href="mailto:pmhshz@gmail.com">pmhshz@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
What if I need to stay in media path in some cases.<br>
Is there a way to prevent this issue?<br>
<div><div></div><div class="Wj3C7c"><br>
<br>
Anthony Minessale-2 wrote:<br>
><br>
> You could use proxy_mode to avoid messing with the SDP at all.<br>
><br>
> set the option inbound-proxy-media to true in the profile and FS will not<br>
> get involved in the codecs<br>
><br>
> On Thu, Nov 6, 2008 at 3:57 AM, shehzad p <<a href="mailto:pmhshz@gmail.com">pmhshz@gmail.com</a>> wrote:<br>
><br>
>><br>
>> In the following trace,<br>
>> aaa.aaa.aaa.aaa = originator<br>
>> bbb.bbb.bbb.bbb = freeswitch<br>
>> ccc.ccc.ccc.ccc = terminator<br>
>><br>
>> ====================================================================<br>
>><br>
>> #<br>
>> U 2008/11/06 03:26:22.208150 aaa.aaa.aaa.aaa:5060 -> bbb.bbb.bbb.bbb:5060<br>
>> INVITE sip:XXXXXXXXXXX@bbb.bbb.bbb.bbb SIP/2.0.<br>
>> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK50b7762d;rport.<br>
>> From: "1598" <sip:1598@aaa.aaa.aaa.aaa>;tag=as0178868b.<br>
>> To: <sip:XXXXXXXXXXX@bbb.bbb.bbb.bbb>.<br>
>> Contact: <sip:1598@aaa.aaa.aaa.aaa>.<br>
>> Call-ID: 651149c41338f2801f8059825e048eaa@aaa.aaa.aaa.aaa.<br>
>> CSeq: 102 INVITE.<br>
>> User-Agent: Asterisk PBX.<br>
>> Max-Forwards: 70.<br>
>> Date: Thu, 06 Nov 2008 09:41:55 GMT.<br>
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<br>
>> Supported: replaces.<br>
>> Content-Type: application/sdp.<br>
>> Content-Length: 261.<br>
>> .<br>
>> v=0.<br>
>> o=root 2055 2055 IN IP4 aaa.aaa.aaa.aaa.<br>
>> s=session.<br>
>> c=IN IP4 aaa.aaa.aaa.aaa.<br>
>> t=0 0.<br>
>> m=audio 13262 RTP/AVP 18 101.<br>
>> a=rtpmap:18 G729/8000.<br>
>> a=fmtp:18 annexb=no.<br>
>> a=rtpmap:101 telephone-event/8000.<br>
>> a=fmtp:101 0-16.<br>
>> a=silenceSupp:off - - - -.<br>
>> a=ptime:20.<br>
>> a=sendrecv.<br>
>><br>
>> #<br>
>> U 2008/11/06 03:26:22.208352 bbb.bbb.bbb.bbb:5060 -> aaa.aaa.aaa.aaa:5060<br>
>> SIP/2.0 100 Trying.<br>
>> Via: SIP/2.0/UDP<br>
>><br>
>> aaa.aaa.aaa.aaa:5060;branch=z9hG4bK50b7762d;rport=5060;received=aaa.aaa.aaa.aaa.<br>
>> From: "1598" <sip:1598@aaa.aaa.aaa.aaa>;tag=as0178868b.<br>
>> To: <sip:XXXXXXXXXXX@bbb.bbb.bbb.bbb>.<br>
>> Call-ID: 651149c41338f2801f8059825e048eaa@aaa.aaa.aaa.aaa.<br>
>> CSeq: 102 INVITE.<br>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported.<br>
>> Content-Length: 0.<br>
>> .<br>
>><br>
>> #<br>
>> U 2008/11/06 03:26:22.233253 bbb.bbb.bbb.bbb:5080 -> ccc.ccc.ccc.ccc:5060<br>
>> INVITE sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc SIP/2.0.<br>
>> Via: SIP/2.0/UDP bbb.bbb.bbb.bbb:5080;rport;branch=z9hG4bK10QUmt8DjFDmF.<br>
>> Max-Forwards: 69.<br>
>> From: "1598" <sip:1598@bbb.bbb.bbb.bbb>;tag=4eyQv6ZZZNBBK.<br>
>> To: <sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc>.<br>
>> Call-ID: 6f5f1baf-267f-122c-1b8c-003048911494.<br>
>> CSeq: 106861655 INVITE.<br>
>> Contact: <sip:mod_sofia@bbb.bbb.bbb.bbb:5080>.<br>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported.<br>
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,<br>
>> NOTIFY,<br>
>> REFER, UPDATE, REGISTER, INFO.<br>
>> Supported: 100rel, timer, precondition, path, replaces.<br>
>> Allow-Events: talk.<br>
>> Min-SE: 120.<br>
>> Content-Type: application/sdp.<br>
>> Content-Disposition: session.<br>
>> Content-Length: 279.<br>
>> Remote-Party-ID: "1598"<br>
>> <sip:1598@bbb.bbb.bbb.bbb>;screen=yes;privacy=off.<br>
>> .<br>
>> v=0.<br>
>> o=FreeSWITCH 4182297201107985931 4664015740918096257 IN IP4<br>
>> bbb.bbb.bbb.bbb.<br>
>> s=FreeSWITCH.<br>
>> c=IN IP4 bbb.bbb.bbb.bbb.<br>
>> t=0 0.<br>
>> a=sendrecv.<br>
>> m=audio 28302 RTP/AVP 18 101.<br>
>> a=rtpmap:18 G729/8000.<br>
>> a=rtpmap:101 telephone-event/8000.<br>
>> a=fmtp:101 0-16.<br>
>> a=silenceSupp:off - - - -.<br>
>> a=ptime:20.<br>
>><br>
>><br>
>> #<br>
>> U 2008/11/06 03:26:22.420544 ccc.ccc.ccc.ccc:5060 -> bbb.bbb.bbb.bbb:5080<br>
>> SIP/2.0 100 Trying.<br>
>> Via: SIP/2.0/UDP bbb.bbb.bbb.bbb:5080;rport;branch=z9hG4bK10QUmt8DjFDmF.<br>
>> From: "1598" <sip:1598@bbb.bbb.bbb.bbb>;tag=4eyQv6ZZZNBBK.<br>
>> To: <sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc>.<br>
>> Call-ID: 6f5f1baf-267f-122c-1b8c-003048911494.<br>
>> CSeq: 106861655 INVITE.<br>
>> Content-Length: 0.<br>
>> .<br>
>><br>
>><br>
>> #<br>
>> U 2008/11/06 03:26:22.433039 ccc.ccc.ccc.ccc:5060 -> bbb.bbb.bbb.bbb:5080<br>
>> SIP/2.0 488 Not Acceptable Here.<br>
>> Supported: 100rel.<br>
>> Via: SIP/2.0/UDP bbb.bbb.bbb.bbb:5080;rport;branch=z9hG4bK10QUmt8DjFDmF.<br>
>> To: <sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc>;tag=3434953230-169789.<br>
>> From: "1598" <sip:1598@bbb.bbb.bbb.bbb>;tag=4eyQv6ZZZNBBK.<br>
>> Remote-Party-Id: "1598"<br>
>> <sip:1598@bbb.bbb.bbb.bbb>;screen=yes;privacy=off.<br>
>> Call-ID: 6f5f1baf-267f-122c-1b8c-003048911494.<br>
>> CSeq: 106861655 INVITE.<br>
>> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,<br>
>> SUBSCRIBE, PRACK.<br>
>> Contact: <sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc:5060>.<br>
>> Call-Info:<br>
>> <sip:ccc.ccc.ccc.ccc>;method="NOTIFY;Event=telephone-event;Duration=1000".<br>
>> Content-Length: 0.<br>
>> .<br>
>><br>
>> #<br>
>> U 2008/11/06 03:26:22.433137 bbb.bbb.bbb.bbb:5080 -> ccc.ccc.ccc.ccc:5060<br>
>> ACK sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc SIP/2.0.<br>
>> Via: SIP/2.0/UDP bbb.bbb.bbb.bbb:5080;rport;branch=z9hG4bK10QUmt8DjFDmF.<br>
>> Max-Forwards: 69.<br>
>> From: "1598" <sip:1598@bbb.bbb.bbb.bbb>;tag=4eyQv6ZZZNBBK.<br>
>> To: <sip:5651XXXXXXXXXXX@ccc.ccc.ccc.ccc>;tag=3434953230-169789.<br>
>> Call-ID: 6f5f1baf-267f-122c-1b8c-003048911494.<br>
>> CSeq: 106861655 ACK.<br>
>> Content-Length: 0.<br>
>> .<br>
>><br>
>> #<br>
>> U 2008/11/06 03:26:22.454386 bbb.bbb.bbb.bbb:5060 -> aaa.aaa.aaa.aaa:5060<br>
>> SIP/2.0 488 Not Acceptable Here.<br>
>> Via: SIP/2.0/UDP<br>
>><br>
>> aaa.aaa.aaa.aaa:5060;branch=z9hG4bK50b7762d;rport=5060;received=aaa.aaa.aaa.aaa.<br>
>> From: "1598" <sip:1598@aaa.aaa.aaa.aaa>;tag=as0178868b.<br>
>> To: <sip:XXXXXXXXXXX@bbb.bbb.bbb.bbb>;tag=vmZD3BgNZQ22c.<br>
>> Call-ID: 651149c41338f2801f8059825e048eaa@aaa.aaa.aaa.aaa.<br>
>> CSeq: 102 INVITE.<br>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported.<br>
>> Accept: application/sdp.<br>
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,<br>
>> NOTIFY,<br>
>> REFER, UPDATE, REGISTER, INFO, PUBLISH.<br>
>> Supported: 100rel, timer, precondition, path, replaces.<br>
>> Allow-Events: talk, presence, dialog, call-info, sla,<br>
>> include-session-description, presence.winfo, message-summary.<br>
>> Reason: Q.850;cause=16;text="NORMAL_CLEARING".<br>
>> Content-Length: 0.<br>
>> .<br>
>><br>
>><br>
>><br>
>> #<br>
>> U 2008/11/06 03:26:22.813849 aaa.aaa.aaa.aaa:5060 -> bbb.bbb.bbb.bbb:5060<br>
>> ACK sip:XXXXXXXXXXX@bbb.bbb.bbb.bbb SIP/2.0.<br>
>> Via: SIP/2.0/UDP aaa.aaa.aaa.aaa:5060;branch=z9hG4bK50b7762d;rport.<br>
>> From: "1598" <sip:1598@aaa.aaa.aaa.aaa>;tag=as0178868b.<br>
>> To: <sip:XXXXXXXXXXX@bbb.bbb.bbb.bbb>;tag=vmZD3BgNZQ22c.<br>
>> Contact: <sip:1598@aaa.aaa.aaa.aaa>.<br>
>> Call-ID: 651149c41338f2801f8059825e048eaa@aaa.aaa.aaa.aaa.<br>
>> CSeq: 102 ACK.<br>
>> User-Agent: Asterisk PBX.<br>
>> Max-Forwards: 70.<br>
>> Content-Length: 0.<br>
>><br>
>> ====================================================================<br>
>><br>
>> Check the above trace.<br>
>> "annexb=no" comes from originator to freeswitch, but while sending to<br>
>> terminator removes that.<br>
>><br>
>> Please Refer the trace and help me for the problem.<br>
>><br>
>><br>
>> Brian West-3 wrote:<br>
>> ><br>
>> > I would need to see a sip trace of this taking place. If you're using<br>
>> > the passthru codec we do pass the fmtp options thru when we receive<br>
>> > them.<br>
>> ><br>
>> > /b<br>
>> ><br>
>> > On Nov 5, 2008, at 8:26 AM, shehzad p wrote:<br>
>> ><br>
>> >><br>
>> >><br>
>> >> I have to route the inbound calls of G729A codec.<br>
>> >> Calls comes to my freeswitch with codec G729A (As "annexb=no" is set)<br>
>> >><br>
>> >> But when i route calls to termination gateway, calls are dropped<br>
>> >> (because<br>
>> >> of "annexb=no " is not set)<br>
>> >><br>
>> >> Why "annexb=no" is removed while i route the calls?<br>
>> >> How can I set "annexb=no'? (I am using javascript for routing the<br>
>> >> calls)<br>
>> >><br>
>> >> Does following SDP variables can help me in solving above problem?<br>
>> >> How to<br>
>> >> use those variables?<br>
>> >> <a href="http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation" target="_blank">http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation</a><br>
>> >><br>
>> >> Warm thanks in advance...<br>
>> >> MSP<br>
>> >> --<br>
>> >> View this message in context:<br>
>> >><br>
>> <a href="http://www.nabble.com/Problem-in-Routing-G729A-Calls-tp20342694p20342694.html" target="_blank">http://www.nabble.com/Problem-in-Routing-G729A-Calls-tp20342694p20342694.html</a><br>
>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.<br>
>> >><br>
>> >><br>
>> ><br>
>> > _______________________________________________<br>
>> > Freeswitch-users mailing list<br>
>> > <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>
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>> ><br>
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>> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>> ><br>
>> ><br>
>><br>
>> --<br>
>> View this message in context:<br>
>> <a href="http://www.nabble.com/Problem-in-Routing-G729A-Calls-tp20342694p20358106.html" target="_blank">http://www.nabble.com/Problem-in-Routing-G729A-Calls-tp20342694p20358106.html</a><br>
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.<br>
>><br>
>><br>
>> _______________________________________________<br>
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>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>><br>
><br>
><br>
><br>
> --<br>
> Anthony Minessale II<br>
><br>
> FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
> ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
><br>
> AIM: anthm<br>
</div></div>> <a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a> <<a href="mailto:MSN%253Aanthony_minessale@hotmail.com">MSN%3Aanthony_minessale@hotmail.com</a>><br>
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><br>
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</div>> <a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a> <<a href="mailto:sip%253A888@conference.freeswitch.org">sip%3A888@conference.freeswitch.org</a>><br>
<div class="Ih2E3d">> <a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
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> pstn:213-799-1400<br>
><br>
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<br>
--<br>
</div>View this message in context: <a href="http://www.nabble.com/Problem-in-Routing-G729A-Calls-tp20342694p20363233.html" target="_blank">http://www.nabble.com/Problem-in-Routing-G729A-Calls-tp20342694p20363233.html</a><br>
<div><div></div><div class="Wj3C7c">Sent from the Freeswitch-users mailing list archive at Nabble.com.<br>
<br>
<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
<br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
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