For the record, it seems to be a SIP-ALG issue. It's fixed now.
Vieri
--- On Wed, 6/8/11, Vieri <rentorbuy at yahoo.com> wrote:
> Hi,
>
> I'm having an issue with all my calls going out my SIP
> provider. I'm using
> a softphone registering to a local Asterisk PBX (I'm using
> Jitsi by the way - it's great and actively growing).
>
> I register as extension 4053 to asterisk server at
> 10.215.147.115 (alias IP -
> real IP addr. is 10.215.147.111) and dial a phone number
> that is routed via
> an Internet SIP provider.
> The call is correctly established and conversation is OK.
> If the local softphone user
> hangs up first, the remote end is also disconnected
> immediately.
> However, if the remote party hangs up first, the local
> caller is not
> immediately disconnected.
> That, of course, is undesirable.
>
> I'd like to understand why the call isn't automatically
> hung up and fix it.
>
> I'm supposing that Jitsi isn't receiving a BYE as expected
> in a correct SIP
> transaction (or BYE is arriving very late).
> I don't know why though.
>
> Here's my network setup:
>
> Softphone asterisk extension 4053 at 10.215.144.48
> Asterisk eth0: 10.215.147.111 but softphone registers to
> the alias/floating IP
> for failover setup 10.215.147.115
> Asterisk eth1: 192.168.103.111
> Asterisk default gateway: 192.168.103.1
> -> Asterisk accesses Internet via eth1 (192.168.103.1 is
> a DSL modem/router)
>
> I did a tcpdump on the asterisk server while calling from
> the local softphone as so:
> tcpdump -s0 -X -n -w asterisk.cap -i eth0 host
> 10.215.144.48
>
> It's here:
> http://213.96.91.201/temp/jitsi_via_asterisk.cap.gz
>
> Here's the full session (softphone waits 2 minutes until it
> finally hangs up):
> http://213.96.91.201/temp/jitsi_via_asterisk_full_session.cap.gz
>
> Asterisk seems to send BYE to the softphone after 120
> seconds since the remote party actually hung up...
>
> A packet dump on eth1 during the call also shows the BYE
> message coming in from the SIP provider:
>
> http://213.96.91.201/temp/asterisk_eth1.txt
>
> I'm almost certain the remote SIP provider sends BYE in
> time because earlier
> today I tested by connecting the softphone directly to the
> SIP provider and going out
> the same DSL line (thus removing Asterisk from the
> equation). ie. I placed a laptop with Jitsi in the same
> subnet
> 192.168.103.0 and used the default gateway 192.168.103.1
> (just like
> Asterisk). All went well.
> I also setup my Jitsi laptop within the 10.215.0.0 subnet
> (just like my
> Asterisk client setup) but connected directly to the SIP
> provider (without
> going through Asterisk). In this case the call ended as
> expected (OK).
> So I guess that something's wrong with my Asterisk
> configuration. Both my softphone and network configuration
> *should* be OK.
>
> However, it may have something to do with my Asterisk
> eth0/eth1 setup but I don't see what.
>
> Any ideas/suggestions?
>
> Thanks,
>
> Vieri
>
>
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