I see this : /etc/asterisk/logger.conf
[logfiles] console => notice,warning,error messages => notice,warning,error,debug,verbose On Thu, Sep 10, 2009 at 2:11 AM, David @ULC <[email protected]> wrote: > *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 ? >>>> >>> >>> >> >
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