[OpenSIPS-Users] Sip invite sent, not reaching dest from certain phones

[Sam Govind]

Looking at the INVITEs I bet the packet is reaching the asterisk server(I
second Abdullah) - try run a tcpdump of whole interface at asterisk server
and see if you're getting anything from OpenSIPs.

Also I found that "rport=32857" is different on the second one so replies
may get lost on that port !

Regards,
-Sammy

On Thu, Sep 22, 2011 at 3:40 AM, Schneur Rosenberg <rosenberg11219 at gmail.com
> wrote:

> both systems are on the open internet, I have no firewalls etc on any
> of the systems, I will try another 2 systems with same configurations
> and see what happens.
>
> On Thu, Sep 22, 2011 at 1:24 AM, Vallimamod ABDULLAH
> <vallimamod.abdullah at imtelecom.fr> wrote:
> > Then you have any intermediate device (known or unknown) that does
> filtering or mangling in some way…
> > Try to trace the sip packet on every hop between the 2 servers to see how
> far it goes.
> >
> > Regards,
> > - vma
> > .
> >
> > On Sep 22, 2011, at 12:07 AM, Schneur Rosenberg wrote:
> >
> >> The packet does not reach asterisk, I did a ngrep on the asterisk
> >> server and not a single packet arrives from the opensips when using
> >> the Aastra phone, therefore its not sending back anything, the
> >> asterisk CLI is also quiet nothing whatsoever :-(
> >>
> >> On Thu, Sep 22, 2011 at 1:03 AM, Vallimamod ABDULLAH
> >> <vallimamod.abdullah at imtelecom.fr> wrote:
> >>> Hi Schneur,
> >>>
> >>> What do you mean precisely by never hitting the asterisk server ?
> >>> As your ngrep trace shows, both packets are sent over the wire to the
> exact same address (68.233.222.9:5060) so they should both reach Asterisk.
> But it's possible that the latter doesn't treat them the same way, depending
> on nat issues most of the time (Asterisk send replies to the contact header
> URI by default if I recall correctly...)
> >>>
> >>> Try to make a ngrep trace on the iface attached to 68.233.222.9 and
> check if Asterisk does not send the answer to a different IP. Also enable
> the debug log on the asterisk console to spot any error / warning messages
> or sip retransmissions.
> >>>
> >>> Hope this would help.
> >
> >
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> >
>
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