[asterisk-users] Cut offs on outgoing SIP calls

[Daniel - Asterisk]

Hey Philipp, I will try soon the new version and let you know.

Currently my users are pointing to a PBX in my local-private network with
no problems.

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).

The funny thing happens when I restart my router, everything works fine,
but 2 or 3 hours later calls start getting  cut-offs again.
I'm not very used to routers but if someone have some tip on Cisco 2811 it
will be great.

Definitely it's a NAT issue, any help is welcome.

Elder D. Arohuanca
Lima - Peru



On Sat, May 18, 2013 at 8:10 PM, Philipp von Klitzing <philipp at vklitzing.com
> wrote:

> Hi!
>
> > I've suffering cut offs after 6 or 7 seconds a call is answered,
> > incoming calls are working fine, but outgoing ones show the gollowing
> > messages when are being dropped
> > [...]
> > It seems the SIP ACK is not being received properly.
>
> I can confirm this issue: In my case it happens with calls coming in from
> a patton ISDN gateway to Asterisk 1.8.20.1.
>
> The calls is processed and passed to a snom phone, audio flows fine for a
> few seconds, but then Asterisk terminates the call. Interestingly this
> never happens on internal calls (from snom to snom). Downgrading to
> Asterisk 1.4 makes the issue go away as well.
>
> Have you tried 1.8.22? I haven't yet, but it seems to come with a fix for
> a deadlock in the SIP channel which *might* solve the issue we are both
> experiencing (see ASTERISK-21389).
>
> Philipp
>
>
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