<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">This looks like you have disable-transcoding set to true.<div><br><div><div apple-content-edited="true"> <span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 14px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 14px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><font class="Apple-style-span" face="'Helvetica Neue'"><font class="Apple-style-span" color="#1C00FF">David Ponzone </font><font class="Apple-style-span" color="#000000" size="3"><span class="Apple-style-span" style="font-size: 12px; ">Direction Technique</span></font></font></div><div><font class="Apple-style-span" face="'Helvetica Neue'"><font class="Apple-style-span" size="3"><span class="Apple-style-span" style="font-size: 13px; ">email: <a href="mailto:david.ponzone@ipeva.fr">david.ponzone@ipeva.fr</a></span></font></font></div><div><font class="Apple-style-span" face="'Helvetica Neue'"><font class="Apple-style-span" size="3"><span class="Apple-style-span" style="font-size: 13px; ">tel: 01 74 03 18 97</span></font></font></div><div><font class="Apple-style-span" face="'Helvetica Neue'"><font class="Apple-style-span" size="3"><span class="Apple-style-span" style="font-size: 13px; ">gsm: 06 66 98 76 34</span></font></font></div><div><font class="Apple-style-span" face="'Helvetica Neue'"><br></font></div><div><font class="Apple-style-span" color="#1C00FF" face="'Helvetica Neue'">Service Client<span class="Apple-converted-space"> </span></font><font class="Apple-style-span" face="'Helvetica Neue'"><font class="Apple-style-span" color="#FF0000">IP</font></font><font class="Apple-style-span" color="#1C00FF" face="'Helvetica Neue'">eva</font></div><div><font class="Apple-style-span" color="#1C00FF" face="'Helvetica Neue'"><span class="Apple-style-span" style="color: rgb(0, 0, 0); font-family: Helvetica; "><div><font class="Apple-style-span" face="'Helvetica Neue'"><font class="Apple-style-span" size="3"><span class="Apple-style-span" style="font-size: 13px; ">tel: 0811 46 26 26</span></font></font></div><div><font class="Apple-style-span" face="'Helvetica Neue'" size="3"><span class="Apple-style-span" style="font-size: 13px; "><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Arial; color: rgb(0, 34, 243); "><span style="text-decoration: underline; "><a href="BLOCKED::http://www.ipeva.fr/">www.ipeva.fr</a></span><span style="color: rgb(101, 104, 149); "> - <span style="color: rgb(0, 34, 243); text-decoration: underline; "><a href="BLOCKED::http://www.ipeva-studio.com/">www.ipeva-studio.com</a></span></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Arial; color: rgb(0, 34, 243); "><span class="Apple-style-span" style="text-decoration: underline; "><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Arial; color: rgb(0, 34, 243); "><span class="Apple-style-span"><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; text-align: justify; font: normal normal normal 10px/normal Arial; color: rgb(192, 192, 192); "><i>Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. </i><b><i>IPeva</i></b><i> décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.</i></div><div style="text-decoration: underline; text-align: justify; "><font class="Apple-style-span" color="#C0C0C0"><i><br></i></font></div></span></div></span></font></div></span></font></div></div></span><br class="Apple-interchange-newline"></div></span><br class="Apple-interchange-newline"> </div><br><div><div>Le 30/06/2010 à 15:07, Mark Campbell-Smith a écrit :</div><br class="Apple-interchange-newline"><blockquote type="cite"><div>Updating the configuration did not help.<br><br>I'm not telling FS to use any other codecs other than what I have<br>specified below, so I'm not sure what I have done wrong. Below is the<br>full trace:<br><br> ------------------------------------------------------------------------<br> INVITE <a href="sip:1020@192.168.1.120">sip:1020@192.168.1.120</a> SIP/2.0<br> Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae<br> From: 1000 <<a href="sip:1000@192.168.1.120">sip:1000@192.168.1.120</a>>;tag=2c7a518d12f9370eo0<br> To: <<a href="sip:1020@192.168.1.120">sip:1020@192.168.1.120</a>><br> Call-ID: <a href="mailto:316156b9-f6b413d2@192.168.1.121">316156b9-f6b413d2@192.168.1.121</a><br> CSeq: 102 INVITE<br> Max-Forwards: 70<br> Proxy-Authorization: Digest<br>username="1000",realm="192.168.1.120",nonce="77836a7a-8447-11df-93a6-d9ad5b204ca2",uri="<a href="sip:1020@192.168.1.120">sip:1020@192.168.1.120</a>",algorithm=MD5,response="c4a5e5cb60676366b8bcfb1329f3fc08",qop=auth,nc=00000001,cnonce="6a0220d0"<br> Contact: 1000 <<a href="sip:1000@192.168.1.121:5060">sip:1000@192.168.1.121:5060</a>><br> Expires: 240<br> User-Agent: Linksys/PAP2T-5.1.6(LS)<br> Content-Length: 451<br> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER<br> Supported: x-sipura, replaces<br> Content-Type: application/sdp<br><br> v=0<br> o=- 17789112 17789112 IN IP4 192.168.1.121<br> s=-<br> c=IN IP4 192.168.1.121<br> t=0 0<br> m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101<br> a=rtpmap:0 PCMU/8000<br> a=rtpmap:102 G726-32/8000<br> a=rtpmap:4 G723/8000<br> a=rtpmap:8 PCMA/8000<br> a=rtpmap:18 G729/8000<br> a=rtpmap:96 G726-40/8000<br> a=rtpmap:97 G726-24/8000<br> a=rtpmap:98 G726-16/8000<br> a=rtpmap:100 NSE/8000<br> a=fmtp:100 192-193<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-15<br> a=ptime:30<br> a=sendrecv<br> ------------------------------------------------------------------------<br>send 343 bytes to udp/[192.168.1.121]:5060 at 13:00:32.882684:<br> ------------------------------------------------------------------------<br> SIP/2.0 100 Trying<br> Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae<br> From: 1000 <<a href="sip:1000@192.168.1.120">sip:1000@192.168.1.120</a>>;tag=2c7a518d12f9370eo0<br> To: <<a href="sip:1020@192.168.1.120">sip:1020@192.168.1.120</a>><br> Call-ID: <a href="mailto:316156b9-f6b413d2@192.168.1.121">316156b9-f6b413d2@192.168.1.121</a><br> CSeq: 102 INVITE<br> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19<br>14-49-15 -0500<br> Content-Length: 0<br><br> ------------------------------------------------------------------------<br>2010-06-30 23:00:32.899423 [DEBUG] sofia.c:5975 IP 192.168.1.121<br>Rejected by acl "domains". Falling back to Digest auth.<br>2010-06-30 23:00:32.914917 [NOTICE] switch_channel.c:776 New Channel<br><a href="mailto:sofia/internal/1000@192.168.1.120">sofia/internal/1000@192.168.1.120</a><br>[77a3ab82-8447-11df-93a7-d9ad5b204ca2]<br>2010-06-30 23:00:32.943135 [DEBUG] switch_core_state_machine.c:314<br>(sofia/internal/1000@192.168.1.120) Running State Change CS_NEW<br>2010-06-30 23:00:32.945232 [DEBUG] switch_core_state_machine.c:320<br>(sofia/internal/1000@192.168.1.120) State NEW<br>2010-06-30 23:00:32.989778 [DEBUG] sofia.c:4293 Channel<br>sofia/internal/1000@192.168.1.120 entering state [received][100]<br>2010-06-30 23:00:32.997500 [DEBUG] sofia.c:4304 Remote SDP:<br>v=0<br>o=- 17789112 17789112 IN IP4 192.168.1.121<br>s=-<br>c=IN IP4 192.168.1.121<br>t=0 0<br>m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:102 G726-32/8000<br>a=rtpmap:4 G723/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729/8000<br>a=rtpmap:96 G726-40/8000<br>a=rtpmap:97 G726-24/8000<br>a=rtpmap:98 G726-16/8000<br>a=rtpmap:100 NSE/8000<br>a=fmtp:100 192-193<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=ptime:30<br><br>2010-06-30 23:00:33.002369 [DEBUG] sofia_glue.c:3877 Audio Codec<br>Compare [PCMU:0:8000:30]/[G729:18:8000:20]<br>2010-06-30 23:00:33.005967 [DEBUG] sofia_glue.c:3877 Audio Codec<br>Compare [PCMU:0:8000:30]/[PCMU:0:8000:20]<br>2010-06-30 23:00:33.008248 [DEBUG] sofia_glue.c:3877 Audio Codec<br>Compare [PCMU:0:8000:30]/[GSM:3:8000:20]<br>2010-06-30 23:00:33.010721 [DEBUG] sofia_glue.c:3924 Substituting<br>codec PCMU@30i@8000h<br>2010-06-30 23:00:33.019002 [DEBUG] sofia_glue.c:2462 Set Codec<br>sofia/internal/1000@192.168.1.120 PCMU/8000 30 ms 240 samples<br>2010-06-30 23:00:33.038714 [DEBUG] sofia_glue.c:3816 Set 2833 dtmf<br>send/recv payload to 101<br>2010-06-30 23:00:33.040992 [DEBUG] sofia.c:4451<br>(sofia/internal/1000@192.168.1.120) State Change CS_NEW -> CS_INIT<br>2010-06-30 23:00:33.043182 [DEBUG] switch_core_session.c:1027 Send<br>signal sofia/internal/1000@192.168.1.120 [BREAK]<br>2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:314<br>(sofia/internal/1000@192.168.1.120) Running State Change CS_INIT<br>2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:338<br>(sofia/internal/1000@192.168.1.120) State INIT<br>2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:83<br>sofia/internal/1000@192.168.1.120 SOFIA INIT<br>2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:117<br>(sofia/internal/1000@192.168.1.120) State Change CS_INIT -> CS_ROUTING<br>2010-06-30 23:00:33.048752 [DEBUG] switch_core_session.c:1027 Send<br>signal sofia/internal/1000@192.168.1.120 [BREAK]<br>2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:338<br>(sofia/internal/1000@192.168.1.120) State INIT going to sleep<br>2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:314<br>(sofia/internal/1000@192.168.1.120) Running State Change CS_ROUTING<br>2010-06-30 23:00:33.048752 [DEBUG] switch_channel.c:1474<br>(sofia/internal/1000@192.168.1.120) Callstate Change DOWN -> RINGING<br>2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:341<br>(sofia/internal/1000@192.168.1.120) State ROUTING<br>2010-06-30 23:00:33.055456 [DEBUG] switch_channel.c:1333<br>(sofia/internal/1000@192.168.1.120) Callstate Change RINGING -> ACTIVE<br>2010-06-30 23:00:33.055456 [DEBUG] mod_sofia.c:140<br>sofia/internal/1000@192.168.1.120 SOFIA ROUTING<br>2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:77<br>sofia/internal/1000@192.168.1.120 Standard ROUTING<br>2010-06-30 23:00:33.055456 [INFO] mod_dialplan_xml.c:331 Processing<br>1000->1020 in context default<br>Dialplan: sofia/internal/1000@192.168.1.120 parsing<br>[default->Local_1000_1019] continue=false<br>Dialplan: sofia/internal/1000@192.168.1.120 Regex (FAIL)<br>[Local_1000_1019] destination_number(1020) =~ /^(10[01][0-9])$/<br>break=on-false<br>Dialplan: sofia/internal/1000@192.168.1.120 parsing<br>[default->Mobile_1020s] continue=false<br>Dialplan: sofia/internal/1000@192.168.1.120 Regex (PASS)<br>[Mobile_1020s] destination_number(1020) =~ /^(102[0-9])$/<br>break=on-false<br>Dialplan: sofia/internal/1000@192.168.1.120 Action set(dialed_extension=1020)<br>Dialplan: sofia/internal/1000@192.168.1.120 Action export(codec_string=GSM)<br>Dialplan: sofia/internal/1000@192.168.1.120 Action set(codec_string=GSM)<br>Dialplan: sofia/internal/1000@192.168.1.120 Action<br>bridge(user/${dialed_extension}@${domain})<br>Dialplan: sofia/internal/1000@192.168.1.120 Action set_user(1000@${domain})<br>Dialplan: sofia/internal/1000@192.168.1.120 Action answer()<br>Dialplan: sofia/internal/1000@192.168.1.120 Action sleep(1000)<br>Dialplan: sofia/internal/1000@192.168.1.120 Action<br>system(/usr/local/freeswitch/scripts/sms.pl ${smsaccount}<br>${smspassword} ${smsnumber} 'You have one new voicemail from ${effec$<br> <action application=)<br>2010-06-30 23:00:33.110926 [DEBUG] switch_core_state_machine.c:119<br>(sofia/internal/1000@192.168.1.120) State Change CS_ROUTING -><br>CS_EXECUTE<br>2010-06-30 23:00:33.114917 [DEBUG] switch_core_session.c:1027 Send<br>signal sofia/internal/1000@192.168.1.120 [BREAK]<br>2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:341<br>(sofia/internal/1000@192.168.1.120) State ROUTING going to sleep<br>2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:314<br>(sofia/internal/1000@192.168.1.120) Running State Change CS_EXECUTE<br>2010-06-30 23:00:33.128360 [DEBUG] switch_core_state_machine.c:348<br>(sofia/internal/1000@192.168.1.120) State EXECUTE<br>2010-06-30 23:00:33.129661 [DEBUG] mod_sofia.c:233<br>sofia/internal/1000@192.168.1.120 SOFIA EXECUTE<br>2010-06-30 23:00:33.134566 [DEBUG] switch_core_state_machine.c:157<br>sofia/internal/1000@192.168.1.120 Standard EXECUTE<br>EXECUTE sofia/internal/1000@192.168.1.120 set(dialed_extension=1020)<br>2010-06-30 23:00:33.157698 [DEBUG] mod_dptools.c:843<br>sofia/internal/1000@192.168.1.120 SET [dialed_extension]=[1020]<br>EXECUTE sofia/internal/1000@192.168.1.120 bridge(user/1020@mydns.dyndns.org)<br>2010-06-30 23:00:33.248458 [DEBUG] switch_ivr_originate.c:1956<br>variable string 0 = [presence_id=1020@mydns.dyndns.org]<br>2010-06-30 23:00:33.260573 [NOTICE] switch_channel.c:776 New Channel<br>sofia/internal/sip:1020@192.168.1.123:5060<br>[77d7e3b6-8447-11df-93a8-d9ad5b204ca2]<br>2010-06-30 23:00:33.303487 [DEBUG] mod_sofia.c:3883<br>(sofia/internal/sip:1020@192.168.1.123:5060) State Change CS_NEW -><br>CS_INIT<br>2010-06-30 23:00:33.304564 [DEBUG] switch_core_session.c:1027 Send<br>signal sofia/internal/sip:1020@192.168.1.123:5060 [BREAK]<br>2010-06-30 23:00:33.327507 [DEBUG] switch_core_state_machine.c:314<br>(sofia/internal/sip:1020@192.168.1.123:5060) Running State Change<br>CS_INIT<br>2010-06-30 23:00:33.332010 [DEBUG] switch_core_state_machine.c:338<br>(sofia/internal/sip:1020@192.168.1.123:5060) State INIT<br>2010-06-30 23:00:33.334621 [DEBUG] mod_sofia.c:83<br>sofia/internal/sip:1020@192.168.1.123:5060 SOFIA INIT<br>send 1183 bytes to udp/[192.168.1.123]:5060 at 13:00:33.351944:<br> ------------------------------------------------------------------------<br> INVITE sip:1020@192.168.1.123:5060 SIP/2.0<br> Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bKFDpKe5v6j2QcQ<br> Max-Forwards: 69<br> From: "1000" <sip:1000@192.168.1.120>;tag=Z51Qve55SUHta<br> To: <sip:1020@192.168.1.123:5060><br> Call-ID: 4f442091-feea-122d-448a-00e04c0312e9<br> CSeq: 132833080 INVITE<br> Contact: <sip:mod_sofia@192.168.1.120:5060><br> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19<br>14-49-15 -0500<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,<br>REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE<br> Supported: timer, precondition, path, replaces<br> Allow-Events: talk, hold, presence, dialog, line-seize, call-info,<br>sla, include-session-description, presence.winfo, message-summary,<br>refer<br> Content-Type: application/sdp<br> Content-Disposition: session<br> Content-Length: 246<br> X-FS-Support: update_display<br> Remote-Party-ID: "1000"<br><sip:1000@192.168.1.120>;party=calling;screen=yes;privacy=off<br><br> v=0<br> o=FreeSWITCH 1277875447 1277875448 IN IP4 192.168.1.120<br> s=FreeSWITCH<br> c=IN IP4 192.168.1.120<br> t=0 0<br> m=audio 27386 RTP/AVP 0 101 13<br> a=rtpmap:0 PCMU/8000<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-16<br> a=rtpmap:13 CN/8000<br> a=ptime:30<br><br>It fails after this with Not Acceptable Here / INCOMPATIBLE_DESTINATION<br><br>On Mon, Jun 28, 2010 at 5:15 PM, David Ponzone <david.ponzone@gmail.com> wrote:<br><blockquote type="cite">Please, retry with a genuine config (the default one would be a wise<br></blockquote><blockquote type="cite">choice).<br></blockquote><blockquote type="cite">David Ponzone Direction Technique<br></blockquote><blockquote type="cite">email: david.ponzone@ipeva.fr<br></blockquote><blockquote type="cite">tel: 01 74 03 18 97<br></blockquote><blockquote type="cite">gsm: 06 66 98 76 34<br></blockquote><blockquote type="cite">Service Client IPeva<br></blockquote><blockquote type="cite">tel: 0811 46 26 26<br></blockquote><blockquote type="cite">www.ipeva.fr - www.ipeva-studio.com<br></blockquote><blockquote type="cite">Ce message et toutes les pièces jointes sont confidentiels et établis à<br></blockquote><blockquote type="cite">l'intention exclusive de ses destinataires. Toute utilisation ou diffusion<br></blockquote><blockquote type="cite">non autorisée est interdite. Tout message électronique est susceptible<br></blockquote><blockquote type="cite">d'altération. IPeva décline toute responsabilité au titre de ce message s'il<br></blockquote><blockquote type="cite">a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce<br></blockquote><blockquote type="cite">message, merci de le détruire immédiatement et d'avertir l'expéditeur.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Le 28/06/2010 à 07:37, Mark Campbell-Smith a écrit :<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Hi All,<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">I'm not really sure if I got a firm answer for this one. Is the only<br></blockquote><blockquote type="cite">way to ensure that transcoding is performed is by using<br></blockquote><blockquote type="cite">late-negotiation? Why aren't all my codecs sent in the INVITE message<br></blockquote><blockquote type="cite">to the B-leg (extension 1020)?<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Thanks!<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">On Thu, Jun 24, 2010 at 9:32 AM, Mark Campbell-Smith<br></blockquote><blockquote type="cite"><mcampbellsmith@gmail.com> wrote:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">FS version and codecs are shown below, but my config file are probably<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">quite old. But I guess they should still work?<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">All codecs are loaded, and the call works if late negotiation is set<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">on profile internal.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">As I wrote above:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">The call setup is extension 1000 calls extension 1020<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">FS as the A-leg codec<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">2. Extension 1020 only supports GSM codec. The call fails with Not<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Acceptable Here.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">I forgot to write that Extension 1000 does not support GSM (I want to<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">force transcoding). Is that why FS is filtering out GSM on the b-leg?<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">freeswitch@internal> version<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500)<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">freeswitch@internal> show codecs<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">type,name,ikey<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,ADPCM (IMA),mod_voipcodecs<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,G.711 alaw,CORE_PCM_MODULE<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,G.711 ulaw,CORE_PCM_MODULE<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,G.722,mod_voipcodecs<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,G.723.1 6.3k,mod_g723_1<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,G.726 16k,mod_voipcodecs<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,G.726 16k (AAL2),mod_voipcodecs<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,G.726 24k,mod_voipcodecs<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,G.726 24k (AAL2),mod_voipcodecs<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,G.726 32k,mod_voipcodecs<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,G.726 32k (AAL2),mod_voipcodecs<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,G.726 40k,mod_voipcodecs<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,G.726 40k (AAL2),mod_voipcodecs<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,G.729,mod_com_g729<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,GSM,mod_voipcodecs<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,H.261 Video (passthru),mod_h26x<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,H.263 Video (passthru),mod_h26x<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,H.263+ Video (passthru),mod_h26x<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,H.263++ Video (passthru),mod_h26x<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,H.264 Video (passthru),mod_h26x<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,LPC-10,mod_voipcodecs<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,PROXY PASS-THROUGH,CORE_PCM_MODULE<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec,Speex,mod_speex<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">25 total.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone <david.ponzone@gmail.com><br></blockquote><blockquote type="cite">wrote:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Mark,<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">I confirm that, as I wrote that wiki page (the early negotiation part) :)<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Can you really confirm your FS version ?<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">The parameter you showed is old.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codec-prefs has been replaced in SIP profiles by:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">David Ponzone Direction Technique<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">email: david.ponzone@ipeva.fr<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">tel: 01 74 03 18 97<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">gsm: 06 66 98 76 34<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Service Client IPeva<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">tel: 0811 46 26 26<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">www.ipeva.fr - www.ipeva-studio.com<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Ce message et toutes les pièces jointes sont confidentiels et établis à<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">l'intention exclusive de ses destinataires. Toute utilisation ou diffusion<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">non autorisée est interdite. Tout message électronique est susceptible<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">d'altération. IPeva décline toute responsabilité au titre de ce message s'il<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">message, merci de le détruire immédiatement et d'avertir l'expéditeur.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Le 23/06/2010 à 14:43, Mark Campbell-Smith a écrit :<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Check this good wiki page for how FS negotiates codecs (early<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">negotiation default):<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">http://wiki.freeswitch.org/wiki/Codec_Negotiation<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">I have this set in my internal profile:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> <param name="codec-prefs" value="$${global_codec_prefs}"/><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">and as stated before vars.xml:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Setting late negotiation works (thanks Sergey), but reading the wiki<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">page, I see the following sentence, which I interpret that GSM should<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">still be sent:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">When FS calls leg B, the list of codecs in outbound-codec-prefs for<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">the SIP profile is reorganized by pushing the codec negotiated above<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">for leg A at the top . If B does not accept any of the codecs, the<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">calls fails, obviously.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><tgraziano@myitdepartment.net> wrote:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">I'm a newb to fs, but doesn't codec get neogtiated by the endpoints?<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Wouldn't fs only get involved when its media server is referred to?<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">If the "other endpoint" will only accept G729, doesn't that mean you<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">need to change that endpoint to also accept G711 or also license G729<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">in FS?<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">On 6/23/10, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Test Setup:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">vars.xml:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">The call setup is extension 1000 calls extension 1020<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">FS as the A-leg codec<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">2. Extension 1020 only supports GSM codec. The call fails with Not<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Acceptable Here.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">FS only offers G729 and PCMU to 1020. How do I change the number of<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codecs that are offered to an extension? I know I can change the<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">order in the codec_prefs, but would prefer FS to offer all three<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">codecs to an extension.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> m=audio 23662 RTP/AVP 0 18 101 13<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> a=rtpmap:0 PCMU/8000<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> a=rtpmap:18 G729/8000<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> a=rtpmap:101 telephone-event/8000<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> a=fmtp:101 0-16<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> a=rtpmap:13 CN/8000<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> a=ptime:30<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Thanks<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">_______________________________________________<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">FreeSWITCH-users mailing list<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">FreeSWITCH-users@lists.freeswitch.org<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">http://www.freeswitch.org<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">--<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Sent from my mobile device<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">======================<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Tony Graziano, Manager<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Telephone: 434.984.8430<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">sip: tgraziano@voice.myitdepartment.net<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Fax: 434.984.8431<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Email: tgraziano@myitdepartment.net<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">LAN/Telephony/Security and Control Systems Helpdesk:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Telephone: 434.984.8426<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">sip: helpdesk@voice.myitdepartment.net<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Fax: 434.984.8427<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Helpdesk Contract Customers:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">http://www.myitdepartment.net/gethelp/<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Why do mathematicians always confuse Halloween and Christmas?<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Because 31 Oct = 25 Dec.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">_______________________________________________<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">FreeSWITCH-users mailing list<br></blockquote><blockquote 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