For the record, it seems to be a SIP-ALG issue. It's fixed now. Vieri
--- On Wed, 6/8/11, Vieri <[email protected]> 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 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar > every Thurs: > > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
