<div dir="ltr"><div>Hey Philipp, I will try soon the new version and let you know. </div><div> </div><div>Currently my users are pointing to a PBX in my local-private network with no problems.</div><div> </div><div>When I use wireshark I see my internal peers trying to send the ACK packets 4 or 5 times until hangup, at the same time the PBX are requesting that very packet many times until it decides to hangup (as you can see in previous message).</div>
<div> </div><div>The funny thing happens when I restart my router, everything works fine, but 2 or 3 hours later calls start getting cut-offs again.</div><div>I'm not very used to routers but if someone have some tip on Cisco 2811 it will be great.</div>
<div> </div><div>Definitely it's a NAT issue, any help is welcome.</div><div> </div><div>Elder D. Arohuanca</div><div>Lima - Peru</div><div> </div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Sat, May 18, 2013 at 8:10 PM, Philipp von Klitzing <span dir="ltr"><<a href="mailto:philipp@vklitzing.com" target="_blank">philipp@vklitzing.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi!<br>
<div class="im"><br>
> I've suffering cut offs after 6 or 7 seconds a call is answered,<br>
> incoming calls are working fine, but outgoing ones show the gollowing<br>
> messages when are being dropped<br>
</div>> [...]<br>
<div class="im">> It seems the SIP ACK is not being received properly.<br>
<br>
</div>I can confirm this issue: In my case it happens with calls coming in from<br>
a patton ISDN gateway to Asterisk 1.8.20.1.<br>
<br>
The calls is processed and passed to a snom phone, audio flows fine for a<br>
few seconds, but then Asterisk terminates the call. Interestingly this<br>
never happens on internal calls (from snom to snom). Downgrading to<br>
Asterisk 1.4 makes the issue go away as well.<br>
<br>
Have you tried 1.8.22? I haven't yet, but it seems to come with a fix for<br>
a deadlock in the SIP channel which *might* solve the issue we are both<br>
experiencing (see ASTERISK-21389).<br>
<span class="HOEnZb"><font color="#888888"><br>
Philipp<br>
</font></span><div class="HOEnZb"><div class="h5"><br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br></div>