Any chance you can get a SIP trace of working vs. not working calls? Put them on <a href="http://pastebin.freeswitch.org">pastebin.freeswitch.org</a> and the community will take a look.<div><br></div><div>-MC<br><br><div class="gmail_quote">
On Fri, May 27, 2011 at 12:31 PM, Michael Gende <span dir="ltr"><<a href="mailto:mgende@gendesign.com">mgende@gendesign.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<span style="font-size:11pt;color:rgb(31, 73, 125)">Hey Folks,<br><br>Kind of strange issue here, tied to some other stuff I've posted about. Anyway, does anyone know how to set ptime or where that is configured? <br>
<br>Please forgive if this is staring me in the face somewhere, I've not found anything satisfactory just looking around. But, I am slow sometimes. <br><br>What's up? We have an FS that has two incoming numbers. One of them, just recently, started doing the following (from our VoIP ISP tech guy):<br>
<br>All calls that
work (to the second number) network originates the call with maxptime=20, we
respond with ptime=20 and everything works fine. The calls that do not
work (primary number) network originates the call with maxptime=30, we
respond with ptime=30 then immediately re-invite with ptime=20. I
suspect somewhere in this re-invite either your switch or our provider’s
switch is getting confused and not sending or processing RTP based on
the re-invite.<br><br>Result? The caller can't be heard by the call taker at our FS. Plus, the sound quality, even of the ring, is bad. <br><br>Sorry so long a post! Right now, we're forwarding the "bad" number to the "good". A band-aid. Any ptime experts here? We did not change this parameter, but did modify the call_timeout up to 19 from 15 trying to give 'em another ring. Then back down once this started to no avail. <br>
<br>Thanks for reading this far if you did. Any advice appreciated. <br><br><br></span>
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