*Details :* * SIP Call Direction: Outgoing Call-ID: [email protected] Our Codec Capability: 256 Non-Codec Capability: 1 Their Codec Capability: 256 Joint Codec Capability: 256 Format g729 Theoretical Address: 209.51.198.114:5060 Received Address: 209.51.198.114:5060 NAT Support: RFC3581 Audio IP: 59.XXX.XXX.XX (local) Our Tag: as0c9f4a40 Their Tag: 0909210916425544477827101 SIP User agent: Username: 18186223080 Peername: sip209 Original uri: sip:209.51.198.114:5060 Need Destroy: 0 Last Message: Tx: ACK Promiscuous Redir: No Route: sip:209.51.198.114:5060;transport=udp DTMF Mode: rfc2833 SIP Options: (none)
On Thu, Sep 10, 2009 at 2:06 AM, David @ULC <[email protected]> wrote: > > Local/718186223...@d 718186223...@default Up > Dial(SIP/18186223...@sip209||t > > > I see this in my Asterisk when I do > > show channels > > > > > On Thu, Sep 10, 2009 at 1:49 AM, David @ULC <[email protected]> wrote: > >> >> I don't know where is the problem. May be with VOIPSwitch OR may be with >> Asterisk.. >> >> Call getting stuck : My agent hang up the call but in Active calls , I see >> call connected and getting charged >> >> I use VOIP and NOT PSTN >> >> Didnt check the Asterisk CLI. Can I get any history of what asterisk >> REALLY had ? >> >> >> >> >> On Wed, Sep 9, 2009 at 11:41 PM, David @ULC <[email protected]> wrote: >> >>> I am using asterisk. >>> >>> I also have an access to VOIPSwitch ver 2 where I can see live calls. >>> >>> Many times I have seen that my calls are getting strucked and then it >>> gets disconneected after 59 mins ( as settings are done accordingly in >>> VOIPSwitch) >>> >>> What could be the reason ? >>> >> >> >
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
