<div dir="ltr"><div>Looks like you have an issue with packet fragmentation.</div><div>I had a similar issue with grandstream a few years back due to ISP improperly handling fragmentation.</div><div>
<pre class="gmail-aLF-aPX-K0-aPE">send 1509 bytes to udp/[34.210.91.114]:5060 at 15:59:12.585438:
------------------------------------------------------------------------
</pre></div><div>Your packet size is 1509 bytes which is more than standard 1500, thus it may be an fragmentation issue @ your router or further down the line.</div><div>That's why you don't get replies to the invite - flowroute don't get a complete request and just ignores it.</div><div>I would disable these to prove the point (will make a packet < 1500 for sure):</div><div>
<pre class="gmail-aLF-aPX-K0-aPE">disable P-Access-Network-Info
disable P-Emergency-Info
<br></pre><pre class="gmail-aLF-aPX-K0-aPE">Also, I would disable alert-info because I've seen some carriers reject the invite with this header.<br>
</pre>

</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Wed, Apr 14, 2021 at 9:20 PM Steven Schoch <<a href="mailto:schoch%2Bfreeswitch.org@xwin32.com">schoch+freeswitch.org@xwin32.com</a>> wrote:<br></div><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">The sip trace is attached.<div>It seems to show that it sends INVITE messages, but never gets a response.</div><div>However, when it sends an OPTIONS message, it does get a response.</div><div>When calling from a different extension (using a Polycom instead of a Grandstream ATA), the INVITE gets answered and the call proceeds.</div><div><br></div><div>It seems that there is something "wrong" with this INVITE that makes Flowroute ignore it. What could that be, and how do I fix it?</div><div><br></div><div>I may have enough data here to ask Flowroute directly, so I'm going to give that a try as well.</div><div><br></div><div>-- </div><div>Steve</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Wed, Apr 14, 2021 at 12:43 PM Brian West <<a href="mailto:brian@freeswitch.com" target="_blank">brian@freeswitch.com</a>> wrote:<br></div><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">That's the FMTP for OPUS.  Chances are that invite breaks the Polycom, what does the SDP look like coming back from that invite? I'll be you, it's broken.<div><br></div><div>/b</div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Wed, Apr 14, 2021 at 1:51 PM Bote Man <<a href="mailto:botelist@gmail.com" target="_blank">botelist@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div lang="EN-US"><div><p class="MsoNormal">I am absolutely no expert on SDP, but that SDP line that begins<u></u><u></u></p><p class="MsoNormal">a=fmtp:102 useinbandfec=1…<u></u><u></u></p><p class="MsoNormal">looks to me like it’s trying to set up a video call. I saw this behavior with the newer Polycom VVX501 before I beat those eager beavers into submission.<u></u><u></u></p><p class="MsoNormal"><br>It also looks like the Grandstream is offering a lot more codecs which you might prefer to trim down to only those necessary to get the job done. Sometimes additional codecs or codecs listed in the “wrong” sequence can cause mystery problems.<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">You might have to resort to siptrace logging between FS and your carrier.<u></u><u></u></p><p class="MsoNormal">sofia profile external siptrace on <span style="font-family:Wingdings">ç</span> or whatever profile handles your provider; or maybe internal to snoop what’s going on between FS and your Grandstream.<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">Hope this helps.<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><span style="font-size:10pt;color:rgb(127,127,127)"> --- <u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:10pt;color:rgb(127,127,127)">John Boteler <u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:10pt;color:rgb(127,127,127)">BnC Group U.S.A. <u></u><u></u></span></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><u></u> <u></u></p><div style="border-color:rgb(225,225,225) currentcolor currentcolor;border-style:solid none none;border-width:1pt medium medium;padding:3pt 0in 0in"><p class="MsoNormal"><b>From:</b> FreeSWITCH-users <<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>> <b>On Behalf Of </b>Steven Schoch<br><b>Sent:</b> Tuesday, 13 April, 2021 20:47<br><b>To:</b> freeswitch-users <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br><b>Subject:</b> [Freeswitch-users] Why doesn't this call get answered?<u></u><u></u></p></div><p class="MsoNormal"><u></u> <u></u></p><div><div><p class="MsoNormal">This office has a bunch of Polycom SoundPoint IP 320 phones, and a single Grandstream HT801 (connected to a FAX machine for outgoing FAXes).<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I can make a call from the phones to outside numbers.<u></u><u></u></p></div><div><p class="MsoNormal">I can make a call from the HT801 to local phones.<u></u><u></u></p></div><div><p class="MsoNormal">But I can't call from the HT801 to outside numbers.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">The last important thing that happens in the failed call is this:<u></u><u></u></p></div><div><p class="MsoNormal">2021-04-13 17:28:15.260844 [DEBUG] sofia.c:7406 Channel sofia/external/<number> entering state [calling][0]<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">The difference between the work and not work seems to be this: When I call from a phone to an outside number, it does this:<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Local SDP:<br>v=0<br>o=FreeSWITCH 1618335260 1618335261 IN IP4 <external-IP><br>s=FreeSWITCH<br>c=IN IP4 <external-IP><br>t=0 0<br>m=audio 25104 RTP/AVP 0 8 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=ptime:20<br>a=sendrecv<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">When it tries to call from the HT801 to an outside number, it does this:<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Local SDP:<u></u><u></u></p></div><div><p class="MsoNormal">v=0<br>o=FreeSWITCH 1618327543 1618327544 IN IP4 <external-IP><br>s=FreeSWITCH<br>c=IN IP4 <external-IP><br>t=0 0<br>m=audio 32552 RTP/AVP 0 8 102 9 101 103<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:102 opus/48000/2<br>a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40; stereo=1<br>a=rtpmap:9 G722/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=rtpmap:103 telephone-event/48000<br>a=fmtp:103 0-16<br>a=ptime:20<br>a=sendrecv<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Is that why it doesn't answer? If so, how do I change it?<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I should mention that when I tried this at home, it worked, but when I attempted to install it here at the bookstore, it didn't. The Comcast router at my home is a little different; they use a Netgear router here; and I may have upgraded Freeswitch between the time it worked and now.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">-- <u></u><u></u></p></div><div><p class="MsoNormal">Steve<u></u><u></u></p></div></div></div></div>_________________________________________________________________________<br>
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