List Info

Thread: Disconnect Request Problem




Disconnect Request Problem
country flaguser name
Japan
2007-10-16 22:26:48
Hi,

I still have problem with Disconnect Request, when
doing call setup.
Here are my extension.conf snippet

[from-zaptel]
exten => s,1,h324m_gw(callsipphone)

[sipphone]
exten => call,1,h324m_gw_answer()
exten => call,n,Dial(SIP/1005openser)

[sipout]
exten => _X.,1,Set(GLOBAL(number)=$)
exten => _X.,2,h324m_call(_X.threegvideo)

[threegvideo]
exten => _X.,1,Set(CHANNEL(transfercapability)=VIDEO)
exten => _X.,n,Dial(Zap/g0/$)

I have tested against asterisk 1.4.11 and 1.4.13.
The calls sometime disconnected in the middle of call
setup, for call-in after invoked
Dial(SIP/1005openser).
And as for call-out, the call disconnected after
CONNECTION ACKNOWLEDGED state.

Also for the call-in against asterisk 1.4.13 i still
can't make it. The calls always disconnected after
Dial(SIP/1005openser), only call-out works.
Below some logs from 1.4.13:
----------------------------
Received out of order SRP_NSRP_RESPONSE [5]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate
Active, peerstate Active
q931.c:2798 q931_disconnect: call 101 on channel 23
enters state 11 (Disconnect Request)
> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 101/0x65) (Terminator)
> Message type: DISCONNECT (69)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU)
standard (0)  Spare: 0  Location: Private network
serving the local user (1)
>                  Ext: 1  Cause: Normal Clearing
(16), class = Normal Event (1) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate
Active, peerstate Active
q931.c:2798 q931_disconnect: call 101 on channel 23
enters state 11 (Disconnect Request)
> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 101/0x65) (Terminator)


Let me know if you need more info.

Regards,
Reza


     
____________________________________________________________
________________________
Catch up on fall's hot new shows on Yahoo! TV. Watch
previews, get listings, and more!
http://tv.yahoo.
com/collections/3658 

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--

asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-video

Re: Disconnect Request Problem
country flaguser name
Spain
2007-10-17 15:15:00
It would be good if you could send me the h245.log, also
it's the behaviour
simillar with different brands of mobile phones?

BR
Sergio

----- Original Message ----- 
From: "Reza Fatahillah" <ezhot_95yahoo.com>
To: <asterisk-videolists.digium.com>
Sent: Wednesday, October 17, 2007 5:26 AM
Subject: [Asterisk-video] Disconnect Request Problem


> Hi,
>
> I still have problem with Disconnect Request, when
> doing call setup.
> Here are my extension.conf snippet
>
> [from-zaptel]
> exten => s,1,h324m_gw(callsipphone)
>
> [sipphone]
> exten => call,1,h324m_gw_answer()
> exten => call,n,Dial(SIP/1005openser)
>
> [sipout]
> exten => _X.,1,Set(GLOBAL(number)=$)
> exten => _X.,2,h324m_call(_X.threegvideo)
>
> [threegvideo]
> exten =>
_X.,1,Set(CHANNEL(transfercapability)=VIDEO)
> exten => _X.,n,Dial(Zap/g0/$)
>
> I have tested against asterisk 1.4.11 and 1.4.13.
> The calls sometime disconnected in the middle of call
> setup, for call-in after invoked
> Dial(SIP/1005openser).
> And as for call-out, the call disconnected after
> CONNECTION ACKNOWLEDGED state.
>
> Also for the call-in against asterisk 1.4.13 i still
> can't make it. The calls always disconnected after
> Dial(SIP/1005openser), only call-out works.
> Below some logs from 1.4.13:
> ----------------------------
> Received out of order SRP_NSRP_RESPONSE [5]
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate
> Active, peerstate Active
> q931.c:2798 q931_disconnect: call 101 on channel 23
> enters state 11 (Disconnect Request)
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 101/0x65)
(Terminator)
> > Message type: DISCONNECT (69)
> > [08 02 81 90]
> > Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU)
> standard (0)  Spare: 0  Location: Private network
> serving the local user (1)
> >                  Ext: 1  Cause: Normal Clearing
> (16), class = Normal Event (1) ]
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate
> Active, peerstate Active
> q931.c:2798 q931_disconnect: call 101 on channel 23
> enters state 11 (Disconnect Request)
> > Protocol Discriminator: Q.931 (8)  len=9
> > Call Ref: len= 2 (reference 101/0x65)
(Terminator)
>
>
> Let me know if you need more info.
>
> Regards,
> Reza
>
>
>
____________________________________________________________
________________
________
> Catch up on fall's hot new shows on Yahoo! TV. Watch
previews, get
listings, and more!
> http://tv.yahoo.
com/collections/3658
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-video
>


_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--

asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-video

Re: Disconnect Request Problem
country flaguser name
Austria
2007-10-18 04:42:43
http:
//sip.fontventa.com/trac/asterisk/ticket/6

Sergio Garcia Murillo schrieb:
> It would be good if you could send me the h245.log,
also it's the behaviour
> simillar with different brands of mobile phones?
> 
> BR
> Sergio
> 
> ----- Original Message ----- 
> From: "Reza Fatahillah" <ezhot_95yahoo.com>
> To: <asterisk-videolists.digium.com>
> Sent: Wednesday, October 17, 2007 5:26 AM
> Subject: [Asterisk-video] Disconnect Request Problem
> 
> 
>> Hi,
>>
>> I still have problem with Disconnect Request, when
>> doing call setup.
>> Here are my extension.conf snippet
>>
>> [from-zaptel]
>> exten => s,1,h324m_gw(callsipphone)
>>
>> [sipphone]
>> exten => call,1,h324m_gw_answer()
>> exten => call,n,Dial(SIP/1005openser)
>>
>> [sipout]
>> exten => _X.,1,Set(GLOBAL(number)=$)
>> exten => _X.,2,h324m_call(_X.threegvideo)
>>
>> [threegvideo]
>> exten =>
_X.,1,Set(CHANNEL(transfercapability)=VIDEO)
>> exten => _X.,n,Dial(Zap/g0/$)
>>
>> I have tested against asterisk 1.4.11 and 1.4.13.
>> The calls sometime disconnected in the middle of
call
>> setup, for call-in after invoked
>> Dial(SIP/1005openser).
>> And as for call-out, the call disconnected after
>> CONNECTION ACKNOWLEDGED state.
>>
>> Also for the call-in against asterisk 1.4.13 i
still
>> can't make it. The calls always disconnected after
>> Dial(SIP/1005openser), only call-out works.
>> Below some logs from 1.4.13:
>> ----------------------------
>> Received out of order SRP_NSRP_RESPONSE [5]
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate
>> Active, peerstate Active
>> q931.c:2798 q931_disconnect: call 101 on channel
23
>> enters state 11 (Disconnect Request)
>>> Protocol Discriminator: Q.931 (8)  len=9
>>> Call Ref: len= 2 (reference 101/0x65)
(Terminator)
>>> Message type: DISCONNECT (69)
>>> [08 02 81 90]
>>> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU)
>> standard (0)  Spare: 0  Location: Private network
>> serving the local user (1)
>>>                  Ext: 1  Cause: Normal
Clearing
>> (16), class = Normal Event (1) ]
>> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate
>> Active, peerstate Active
>> q931.c:2798 q931_disconnect: call 101 on channel
23
>> enters state 11 (Disconnect Request)
>>> Protocol Discriminator: Q.931 (8)  len=9
>>> Call Ref: len= 2 (reference 101/0x65)
(Terminator)
>>
>> Let me know if you need more info.
>>
>> Regards,
>> Reza
>>
>>
>>
>
____________________________________________________________
________________
> ________
>> Catch up on fall's hot new shows on Yahoo! TV.
Watch previews, get
> listings, and more!
>> http://tv.yahoo.
com/collections/3658
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
>>
>> asterisk-video mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-video
>>
> 
> 
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
> 
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-video

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--

asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-video

[1-3]

about | contact  Other archives ( Real Estate discussion Medical topics )