<div dir="ltr"><div>Hi,</div><div><br></div><div>I have Asterisk 10.12.1. I can not figure out the solution. </div><div><br></div><div>Thank you for your help.</div><div><br></div><div>Best Regards</div></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Thu, Nov 21, 2013 at 7:07 PM, Alyed <span dir="ltr"><<a href="mailto:alyed@vivoxie.com" target="_blank">alyed@vivoxie.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><div><div>Which version of Asterisk are you using?<br><br></div>According to <a href="http://www.voip-info.org/wiki/view/Asterisk%20T.38" target="_blank">http://www.voip-info.org/wiki/view/Asterisk%20T.38</a> unless you are using Asterisk 10, there's quite some patching (or buying) you'll need to be doing.<br>
<br></div>Alyed<br><div><div><div><br><div class="gmail_extra"><br><div class="gmail_quote">2013/11/21 Bryant Zimmerman <span dir="ltr"><<a href="mailto:BryantZ@zktech.com" target="_blank">BryantZ@zktech.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><span style="font-family:Tahoma,Geneva,sans-serif;font-size:10pt">Can you funnel them through a specific inbound dial context. Then force a re-invite to g729?<br>
<br><div>Thanks<br>
<br>
Bryant Zimmerman (ZK Tech Inc.)<br>
616-855-1030 Ext. 2003</div><br><br><span style="font-family:tahoma,arial,sans-serif;font-size:10pt"><hr align="center" size="2" width="100%"><b>From</b>: "Damian Gonzalez" <<a href="mailto:dgonzalez@denwaip.com" target="_blank">dgonzalez@denwaip.com</a>><br>
<b>Sent</b>: Thursday, November 21, 2013 8:25 AM<br><b>To</b>: "Asterisk Users Mailing List - Non-Commercial Discussion" <<a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a>><br>
<b>Subject</b>: Re: [asterisk-users] Movistar sip Mexico</span><div><div><br><br><div dir="ltr">Any posible solution?</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner <span dir="ltr"><<a href="mailto:kris@kriskinc.com" target="_blank">kris@kriskinc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr">It is possible that Asterisk requires an rtpmap even for static payload types (I'm not sure about this). The INVITE from your provider omits rtpmap for payload type 18 (G729) which is perfectly valid.<br>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez <span dir="ltr"><<a href="mailto:dgonzalez@denwaip.com" target="_blank">dgonzalez@denwaip.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div><div>Hello,</div><div><br></div><div>Thanks for the quickly response. I have only G729 in the peer but I have t38pt_udptl= yes</div>
<div><br></div><div>If I put t38pt_udptl=no , asterisk reject the call with 488 code.</div>
<div><br></div><div>The problem is that Movistar send T38 codec in all calls and I need ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have only T38 I have to negociate a fax call.</div></div><div><br>
</div><div>Thanks.</div></div><div class="gmail_extra"><div><div><br><br><div class="gmail_quote">On Wed, Nov 20, 2013 at 4:46 PM, Alyed <span dir="ltr"><<a href="mailto:alyed@vivoxie.com" target="_blank">alyed@vivoxie.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div><div><div>Think you only need to make sure you have in your sip.conf file these configs:<br>
<br></div>
[your-device-name]<br></div>.....<br>.....<br>disallow=all<br></div>allow=g729<br><div><div><div>.....<br>
.....<br><div><div><br><br>Alyed<br><div><div class="gmail_extra"><br><div class="gmail_quote">2013/11/20 Damian Gonzalez <span dir="ltr"><<a href="mailto:dgonzalez@denwaip.com" target="_blank">dgonzalez@denwaip.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div><div>Hello,</div><div><br></div><div>I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call.</div>
<div><br></div><div>When a fax call is made Movistar send only T38 in the INVITE. </div><div><br></div><div>Invite example:</div></div><div><br></div><div><div>v=0</div><div>o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2</div>
<div>s=sip call</div><div>c=IN IP4 192.168.1.2</div><div>t=0 0</div><div>m=audio 6370 RTP/AVP 18 101</div><div>a=fmtp:18 annexb=yes</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-15</div><div>a=ptime:20</div>
<div>m=image 6372 udptl t38</div><div>a=T38FaxVersion:0</div><div>a=T38FaxMaxBuffer:1100</div><div>a=T38FaxMaxDatagram:612</div><div>a=T38MaxBitRate:14400</div><div>a=T38FaxRateManagement:transferredTCF</div><div>a=T38FaxUdpEC:t38UDPRedundancy</div>
</div><div><br></div><div><div>How can I ignore T38 and use only G729 for this call?.</div></div><div><br></div><div>Thanks for your help.</div><span><font color="#888888"><div><br></div><div>Damian</div>
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