[asterisk-users] Call getting stucked !!

David @ULC ucoms2001 at gmail.com
Wed Sep 9 15:45:11 CDT 2009


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 <ucoms2001 at gmail.com> wrote:

> *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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