[asterisk-users] Call getting stucked !!

[David @ULC]

*Details :*

  * SIP Call
  Direction:              Outgoing
  Call-ID:                593722384f5174775f83837c30fd51b0 at 59.165.44.21
  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 <ucoms2001 at gmail.com> wrote:

>
> Local/718186223080 at d 718186223080 at default Up
> Dial(SIP/18186223080 at sip209||t
>
>
> I see this in my Asterisk when I do
>
> show channels
>
>
>
>
> On Thu, Sep 10, 2009 at 1:49 AM, David @ULC <ucoms2001 at gmail.com> 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 <ucoms2001 at gmail.com> 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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