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<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">And you also sending with T.38.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">I dont know what Asterisk Versions you are using but check this:<o:p></o:p></span></p>
<ul type="disc">
<li class="MsoNormal" style="color:windowtext;mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo1">
Asterisk <b>1.2</b> has no support for T.38.<o:p></o:p></li><li class="MsoNormal" style="color:windowtext;mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo1">
Asterisk <b>1.4</b> supports only T.38 fax pass through; there is however a <a href="http://www.voip-info.org/wiki/view/T38modem+configuration+with+Asterisk" title="T38modem configuration with Asterisk">
third party way</a> using HylaFax and OPAL to send and receive fax through Asterisk 1.4. See also rejected
<a href="http://bugs.digium.com/view.php?id=12931" target="_blank">patch 12931</a> that includes a T.38 gateway.
<a href="http://www.attractel.com/t38.html" target="_blank">Attractel</a> offers a commercial T.38 gateway solution for Asterisk.<o:p></o:p></li><li class="MsoNormal" style="color:windowtext;mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo1">
In Asterisk <b>1.6</b> also origination and termination features will be added (with gateway functionality still missing)<o:p></o:p></li><li class="MsoNormal" style="color:windowtext;mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo1">
NEW: In 2009 Digium introduced the commercial "Fax for Asterisk" with T.38 support (free licence for 1 channel)<o:p></o:p></li><li class="MsoNormal" style="color:windowtext;mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo1">
For Asterisk <b>1.8</b> exist T.38 Gateway patch http://bugs.digium.com/view.php?id=13405<o:p></o:p></li><li class="MsoNormal" style="color:windowtext;mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0 level1 lfo1">
In Asterisk <b>10</b> is <a href="http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway" title="Asterisk T.38 Gateway">
Asterisk T.38 Gateway</a> integrated<o:p></o:p></li></ul>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">Regards,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black">Max<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:black"><o:p> </o:p></span></p>
<div>
<div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm">
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext">Von:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext"> spce-user-bounces@lists.sipwise.com [mailto:spce-user-bounces@lists.sipwise.com]
<b>Im Auftrag von </b>Mario Contreras<br>
<b>Gesendet:</b> Montag, 13. Mai 2013 17:21<br>
<b>An:</b> Imanol Pardavila<br>
<b>Cc:</b> spce-user@lists.sipwise.com<br>
<b>Betreff:</b> Re: [Spce-user] Remove SDP media-level attribute - direction<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Hi all.<br>
<br>
Sorry for reopen this old thread. I can't remove that line from body. I have two asterisks, which I use for sending and receiving fax, but it's not working, and I'm not sure why. This is the caller leg conversation:<br>
<img border="0" width="199" height="422" id="Bild_x0020_1" src="cid:image001.png@01CE4FFF.3B32E8D0" alt="cid:part1.04020200.05000100@innovasur.es"><br>
And the log in asterisk(the one sending 488), show this:<br>
<br>
<tt><span style="font-size:10.0pt">[May 13 17:02:32] DEBUG[1911] chan_sip.c: Initializing initreq for method INVITE - callid 713319c15c67f95e34adee3c59efc3bf@domain</span></tt><span style="font-size:10.0pt;font-family:"Courier New""><br>
<tt>[May 13 17:02:32] DEBUG[1911] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for IP_SPCE:5060</tt><br>
<tt>[May 13 17:02:32] VERBOSE[1911] app_dial.c: -- Called sip/911111111@fax</tt><br>
<tt>[May 13 17:02:32] DEBUG[1911] channel.c: Set channel SIP/fax-00000008 to read format slin</tt><br>
<tt>[May 13 17:02:32] DEBUG[1911] channel.c: Set channel SIP/fax-00000008 to write format slin</tt><br>
<tt>[May 13 17:02:32] DEBUG[31087] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '713319c15c67f95e34adee3c59efc3bf@domain</tt><br>
<tt>...skipping...</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: Processing session-level SDP c=IN IP4 46.39.192.6... OK.</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.</tt><br>
<tt>[May 13 17:02:36] VERBOSE[31087] netsock.c: == Using UDPTL CoS mark 5</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: Setting NAT on UDPTL to Off</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: FaxVersion: 0</tt></span><br>
<tt><span style="font-size:10.0pt">[May 13 17:02:36] DEBUG[31087] chan_sip.c: Processing media-level (image) SDP a=T38FaxVersion:0... OK.</span></tt><span style="font-size:10.0pt;font-family:"Courier New""><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: T38MaxBitRate: 14400</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: Processing media-level (image) SDP a=T38MaxBitRate:14400... OK.</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: RateManagement: transferredTCF</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: Processing media-level (image) SDP a=T38FaxRateManagement:transferredTCF... OK.</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: FaxMaxDatagram: 849</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: Processing media-level (image) SDP a=T38FaxMaxDatagram:849... OK.</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: UDP EC: t38UDPFEC</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: Processing media-level (image) SDP a=T38FaxUdpEC:t38UDPFEC... OK.</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: Processing media-level (image) SDP a=direction:active... UNSUPPORTED.</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: Processing media-level (image) SDP a=nortpproxy:yes... UNSUPPORTED.</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: T38 state changed to 2 on channel SIP/fax-00000008</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: Have T.38 but no audio, accepting offer anyway</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: SIP/fax-00000008: This call is UP....</tt><br>
<tt>[May 13 17:02:36] DEBUG[31087] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for IP_SPCE:5060</tt><br>
<tt>[May 13 17:02:36] DEBUG[1910] chan_sip.c: T38 state changed to 4 on channel SIP/fax-00000008</tt><br>
<tt>[May 13 17:02:36] DEBUG[1910] chan_sip.c: Trying to put 'SIP/2.0 488' onto UDP socket destined for IP_SPCE:5060</tt></span><br>
<br>
I think it's caused by these lines:<o:p></o:p></p>
<p class="MsoNormal">direction:active<br>
nortpproxy:yes<o:p></o:p></p>
<p class="MsoNormal" style="margin-bottom:12.0pt">That attributes are created in spce, not in asterisk. Anyway, like I've said, I've tried the modification made by Imanol, but although in kamailio-proxy log appears "Rewriting T.38 'UDPTL' to 'udptl'", the line
is still in body. <br>
<br>
Any ideas? Thanks!<br>
<br>
<o:p></o:p></p>
<div>
<p class="MsoNormal">El 19/11/2012 18:35, Imanol Pardavila escribió:<o:p></o:p></p>
</div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal" style="margin-bottom:12.0pt">Hi everybody,<br>
<br>
Finally it works!. <br>
If somebody has the same problem, I inserted the modification in two places:<br>
<br>
/etc/ngcp-config/templates/etc/kamailio/proxy/<b>proxy.cfg.customtt.tt2</b><br>
<br>
########################################################################<br>
# Reply route 'base-nat-reply'<br>
########################################################################<br>
onreply_route[REPLY_ROUTE_NAT]<br>
{<br>
xlog("L_INFO", "NAT-Reply - [% logres_init -%]\n");<br>
force_rport();<br>
<br>
if(has_body("application/sdp") && ($avp(s:from_faxserver) == 1 || $avp(s:to_faxserver) == 1))<br>
{<br>
# fix for T.38 to patton faxserver gateway, which chokes on upper-case<br>
xlog("L_INFO", "Rewriting T.38 'UDPTL' to 'udptl' - [% logres -%]\n");<br>
replace_body_all(" UDPTL ", " udptl ");<br>
}<br>
<br>
<b>if(has_body("application/sdp"))<br>
{<br>
# fix SDP direction attribute for Sonus SBC<br>
xlog("L_INFO", "Removing direction attribute from SDP - [% logreq -%]\n");<br>
replace_body_all("a=direction:active\r?\n", "");<br>
}</b><br>
<br>
if(status=~"(180)|(183)|2[0-9][0-9]") <br>
...<br>
...<br>
<br>
########################################################################<br>
# Branch route 'cli-rtp'<br>
########################################################################<br>
branch_route[BRANCH_ROUTE_CLI_RTP]<br>
...<br>
...<br>
if($avp(s:callee_uuid) != $null && !is_present_hf("P-Callee-UUID"))<br>
{<br>
xlog("L_INFO", "Setting P-Callee-UUID to '$avp(s:callee_uuid)' - [% logreq -%]\n");<br>
append_hf("P-Callee-UUID: $avp(s:callee_uuid)\r\n");<br>
}<br>
<br>
<b>if(has_body("application/sdp"))<br>
{<br>
# fix SDP direction attribute for Sonus SBC<br>
xlog("L_INFO", "Removing direction attribute from SDP - [% logreq -%]\n");<br>
replace_body_all("a=direction:active\r?\n", "");<br>
}</b><br>
<br>
xlog("L_INFO", "Request leaving server, D-URI='$du' - [% logreq -%]\n");<br>
}<br>
<br>
I hope it would be helpful for somebody.<br>
<br>
Thanks Andreas<br>
Best regards<br>
Imanol<o:p></o:p></p>
<div>
<p class="MsoNormal">El 14/11/2012 10:20, Andreas Granig escribió:<o:p></o:p></p>
</div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<pre>Hi,<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>You might want to remove the newline as well, like this:<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>replace_body_all("a=directmedia:active\r?\n", "");<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>Andreas<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>On 11/14/2012 10:12 AM, Imanol Pardavila wrote:<o:p></o:p></pre>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<pre>Hi,<o:p></o:p></pre>
<pre>I was able to remove the directionline, but the problem is that it<o:p></o:p></pre>
<pre>leaves an empty line between ptime and oldmediaip and SBC returns "400<o:p></o:p></pre>
<pre>Bad Request"due to malformed INVITE:<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>a=sendrecv<o:p></o:p></pre>
<pre>a=ptime:20<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>a=oldmediaip:10.34.240.60<o:p></o:p></pre>
<pre>a=nortpproxy:yes<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>I tried both:<o:p></o:p></pre>
<pre>replace_body_all("a=direction:active" ,"");<o:p></o:p></pre>
<pre>replace_body_all("a=direction:active" ,"");<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>with same result.<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>How could I remove the empty line?<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>Thanks<o:p></o:p></pre>
<pre>Best regards<o:p></o:p></pre>
<pre>Imanol<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>El 09/11/2012 14:05, Imanol Pardavila escribió:<o:p></o:p></pre>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<pre>Hi Andreas,<o:p></o:p></pre>
<pre>"kamailio.cfg.tt2" file has the following main structure:<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>route<o:p></o:p></pre>
<pre>...<o:p></o:p></pre>
<pre>route[ROUTE_RELAY]<o:p></o:p></pre>
<pre>...<o:p></o:p></pre>
<pre>route[ROUTE_REQUEST]<o:p></o:p></pre>
<pre>...<o:p></o:p></pre>
<pre>onreply_route<o:p></o:p></pre>
<pre>...<o:p></o:p></pre>
<pre>onsend_route<o:p></o:p></pre>
<pre>...<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>I guess that I have to insert something like this:<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>if(has_body("application/sdp")<o:p></o:p></pre>
<pre>{<o:p></o:p></pre>
<pre> # fix for Sonus SBC - remove directmedia attribute<o:p></o:p></pre>
<pre> xlog("L_INFO", "Removing directmedia attribute - [% logreq -%]\n");<o:p></o:p></pre>
<pre> replace_body_all(" a=directmedia:active ", ""); <o:p></o:p></pre>
<pre>}<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>But I'm not sure about where to insert it. Could you help me?<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>Thanks<o:p></o:p></pre>
<pre>Best regards<o:p></o:p></pre>
<pre>Imanol<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>El 09/11/2012 12:45, Andreas Granig escribió:<o:p></o:p></pre>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<pre>Hi Imanol,<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>Oh, right, it's added by the proxy then after mediaproxy is engaged. So<o:p></o:p></pre>
<pre>the best place would be the load-balancer config at<o:p></o:p></pre>
<pre>/etc/ngcp-config/templates/etc/kamailio/lb/kamailio.cfg.customtt.tt2.<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>Andreas<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>On 11/09/2012 11:49 AM, Imanol Pardavila wrote:<o:p></o:p></pre>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<pre>Hi Andreas,<o:p></o:p></pre>
<pre>Looking at my traces I'm sure that endpoints aren't sending this<o:p></o:p></pre>
<pre>attribute; I've just filtered the first packet with this line on SDP<o:p></o:p></pre>
<pre>body and it's been sent by CE.<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre> * SDP sent from our SBC to CE:<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre> Session Description Protocol<o:p></o:p></pre>
<pre> Session Description Protocol Version (v): 0<o:p></o:p></pre>
<pre> Owner/Creator, Session Id (o): Sonus_UAC IN IP4 10.34.241.67<o:p></o:p></pre>
<pre> Session Name (s): SIP Media Capabilities<o:p></o:p></pre>
<pre> Connection Information (c): IN IP4 10.34.240.60<o:p></o:p></pre>
<pre> Time Description, active time (t): 0 0<o:p></o:p></pre>
<pre> Media Description, name and address (m): audio 1552 RTP/AVP<o:p></o:p></pre>
<pre>0 8 101<o:p></o:p></pre>
<pre> Media Attribute (a): rtpmap:0 PCMU/8000<o:p></o:p></pre>
<pre> Media Attribute (a): rtpmap:8 PCMA/8000<o:p></o:p></pre>
<pre> Media Attribute (a): rtpmap:101 telephone-event/8000<o:p></o:p></pre>
<pre> Media Attribute (a): fmtp:101 0-15<o:p></o:p></pre>
<pre> Media Attribute (a): sendrecv<o:p></o:p></pre>
<pre> Media Attribute (a): ptime:20<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre> * SDP sent from CE to our SBC (following packet in the sequence):<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre> Session Description Protocol<o:p></o:p></pre>
<pre> Session Description Protocol Version (v): 0<o:p></o:p></pre>
<pre> Owner/Creator, Session Id (o): Sonus_UAC IN IP4 10.34.242.11<o:p></o:p></pre>
<pre> Session Name (s): SIP Media Capabilities<o:p></o:p></pre>
<pre> Connection Information (c): IN IP4 10.34.242.11<o:p></o:p></pre>
<pre> Time Description, active time (t): 0 0<o:p></o:p></pre>
<pre> Media Description, name and address (m): audio 30132 RTP/AVP<o:p></o:p></pre>
<pre>0 8 101<o:p></o:p></pre>
<pre> Media Attribute (a): rtpmap:0 PCMU/8000<o:p></o:p></pre>
<pre> Media Attribute (a): rtpmap:8 PCMA/8000<o:p></o:p></pre>
<pre> Media Attribute (a): rtpmap:101 telephone-event/8000<o:p></o:p></pre>
<pre> Media Attribute (a): fmtp:101 0-15<o:p></o:p></pre>
<pre> Media Attribute (a): sendrecv<o:p></o:p></pre>
<pre> Media Attribute (a): ptime:20<o:p></o:p></pre>
<pre> *Media Attribute (a): direction:active**<o:p></o:p></pre>
<pre>** Media Attribute (a): oldmediaip:10.34.240.60**<o:p></o:p></pre>
<pre>** Media Attribute (a): nortpproxy:yes*<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>Flow diagram:<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>|endpoint A| ------------- |SBC| ------------- |CE| ------------- |SBC|<o:p></o:p></pre>
<pre>------------- |endpoint B|<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>SBC: 10.34.240.60, 10.34.241.67<o:p></o:p></pre>
<pre>CE: 10.34.242.11<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>Any ideas?<o:p></o:p></pre>
<pre>Thanks<o:p></o:p></pre>
<pre>Best regards<o:p></o:p></pre>
<pre>Imanol<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>El 08/11/2012 20:22, Andreas Granig escribió:<o:p></o:p></pre>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<pre>Hi,<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>On 11/08/2012 05:25 PM, Imanol Pardavila wrote:<o:p></o:p></pre>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<pre>I'm not sure about who builds SDP? lb, proxy, sems? I guess that the<o:p></o:p></pre>
</blockquote>
<pre>The SDP is created by the endpoints, not by the CE.<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<pre>idea is create my customtt.tt2 file and use replace_body_all () function<o:p></o:p></pre>
<pre>something like:<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre># remove direction attribute<o:p></o:p></pre>
<pre>replace_body_all(" a=directmedia:active ", "");<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>Is it correct?<o:p></o:p></pre>
</blockquote>
<pre>Not sure about the whitespaces, you might need to leave them out I guess.<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>Andreas<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre><o:p> </o:p></pre>
<pre><o:p> </o:p></pre>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>Spce-user mailing list<o:p></o:p></pre>
<pre><a href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a><o:p></o:p></pre>
<pre><a href="http://lists.sipwise.com/listinfo/spce-user">http://lists.sipwise.com/listinfo/spce-user</a><o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre><o:p> </o:p></pre>
<pre>-----<o:p></o:p></pre>
<pre>No se encontraron virus en este mensaje.<o:p></o:p></pre>
<pre>Comprobado por AVG - <a href="http://www.avg.com">www.avg.com</a><o:p></o:p></pre>
<pre>Version: 2012.0.2221 / Base de datos de virus: 2441/5382 - Fecha de publicacion: 11/08/12<o:p></o:p></pre>
</blockquote>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>Spce-user mailing list<o:p></o:p></pre>
<pre><a href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a><o:p></o:p></pre>
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<pre><o:p> </o:p></pre>
</blockquote>
<pre><o:p> </o:p></pre>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>Spce-user mailing list<o:p></o:p></pre>
<pre><a href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a><o:p></o:p></pre>
<pre><a href="http://lists.sipwise.com/listinfo/spce-user">http://lists.sipwise.com/listinfo/spce-user</a><o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre><o:p> </o:p></pre>
<pre>-----<o:p></o:p></pre>
<pre>No se encontraron virus en este mensaje.<o:p></o:p></pre>
<pre>Comprobado por AVG - <a href="http://www.avg.com">www.avg.com</a><o:p></o:p></pre>
<pre>Version: 2012.0.2221 / Base de datos de virus: 2441/5383 - Fecha de publicacion: 11/08/12<o:p></o:p></pre>
</blockquote>
</blockquote>
<pre><o:p> </o:p></pre>
<pre><o:p> </o:p></pre>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>Spce-user mailing list<o:p></o:p></pre>
<pre><a href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a><o:p></o:p></pre>
<pre><a href="http://lists.sipwise.com/listinfo/spce-user">http://lists.sipwise.com/listinfo/spce-user</a><o:p></o:p></pre>
<pre><o:p> </o:p></pre>
</blockquote>
<p class="MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>Spce-user mailing list<o:p></o:p></pre>
<pre><a href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a><o:p></o:p></pre>
<pre><a href="http://lists.sipwise.com/listinfo/spce-user">http://lists.sipwise.com/listinfo/spce-user</a><o:p></o:p></pre>
<p class="MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
<pre>-----<o:p></o:p></pre>
<pre>No se encontraron virus en este mensaje.<o:p></o:p></pre>
<pre>Comprobado por AVG - <a href="http://www.avg.com">www.avg.com</a><o:p></o:p></pre>
<pre>Version: 2012.0.2221 / Base de datos de virus: 2441/5393 - Fecha de publicacion: 11/13/12<o:p></o:p></pre>
</blockquote>
<p class="MsoNormal"><o:p> </o:p></p>
</blockquote>
<p class="MsoNormal"><o:p> </o:p></p>
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