<div dir="ltr">I am stumped so far. What is most interesting to me is that Asterisk is actually sending two BYE transactions for the same dialog, at basically the same time. I am still going through your traces again, but maybe someone else has suggestions on how to add more debug to the res_pjsip logging that would prove useful.</div><br><div class="gmail_quote"><div dir="ltr">On Thu, Feb 11, 2016 at 1:33 AM Dmitriy Serov <<a href="mailto:serov.d.p@gmail.com">serov.d.p@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div>The call initiated from internal
extension.<br>
<br>
I have made two test call:<br>
Successful call from device on res_pjsip via endpoint on chan_sip:
<a href="http://pastebin.com/LWeDYstj" target="_blank">http://pastebin.com/LWeDYstj</a><br>
Unsuccessful call from device on res_pjsip via endpoint on
res_pjsip: <a href="http://pastebin.com/hepVb6Nu" target="_blank">http://pastebin.com/hepVb6Nu</a><br>
<br>
And ones again i don't see anything that would make asterisk send
BYE.<br>
<br>
I would be grateful for any ideas.<br>
<br>
11.02.2016 1:47, Trey Hilyard пишет:<br>
</div></div><div text="#000000" bgcolor="#FFFFFF">
<blockquote type="cite">
<p dir="ltr">How are you initiating the call out to that server?
Are you dialing from an internal phone or doing it from the CLI?
It looks like it is from an internal extension, if I were
guessing, but that side of the call isn't in your log.</p>
<p dir="ltr">If it is from an internal extension, I think a SIP
trace on that side would help.</p>
<br>
<div class="gmail_quote">
<div dir="ltr">On Wed, Feb 10, 2016, 3:20 PM Dmitriy Serov <<a href="mailto:serov.d.p@gmail.com" target="_blank"><a href="mailto:serov.d.p@gmail.com" target="_blank">serov.d.p@gmail.com</a></a>>
wrote:<br>
</div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> Please help find the
cause of strange behavior res_pjsip. <br>
<br>
Making outgoint call to other sip server (CommuniGatePro),
my asterisk suddenly sends BYE after picking up! <br>
Partial log of an outgoing call with full debug is attached
and on web: <a href="http://pastebin.com/tLNCpx4d" target="_blank">http://pastebin.com/tLNCpx4d</a><br>
<br>
No diagnostic messages why asterisk suddenly decided to
hangup i don't found :(<br>
<br>
There are suggestions or strong belief about the reasons of
such behavior?<br>
<br>
Thanks.<br>
<br>
Dmitriy.<br>
<span style="margin:0px;padding:0px;outline:0px;color:rgb(34,34,34);font-family:Arial,Helvetica,sans-serif;font-size:20px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:24px;text-align:start;text-indent:0px;text-transform:none;white-space:pre-wrap;word-spacing:0px;background-color:rgb(255,255,255)"></span>
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