[asterisk-users] Send SIP 100 Trying instead of 183 Session Progress

[Richard Revels]

And in case that link doesn't work so well in text email clients here  
is the real address.

lists.digium.com/pipermail/asterisk-dev/2006-May.txt.gz

Richard

On Dec 21, 2007, at 4:24 PM, Richard Revels wrote:

> You are probably running into the problem described below.  Below  
> that is a link to the original document with the code patch.  I put  
> it on a PRI box we use inhouse and it took care of the 183 before a  
> busy for me.  However, this is a box we use inhouse.  I've never put  
> it on anything in production.  Your mileage may vary
>
> >>>>>>>>>>>>>>>>>>>>>>
> gday guys (n'gals).
>
> I have a third party SIP platform which generates outbound calls via
> asterisk to ISDN (Australia - so thats ETSI ISDN).   This platform  
> doesn't
> really like inband signalling on outbound calls (ie getting 183's  
> with SDP
> -- its fine with 180 Ringing etc...)
>
> Having had a bit of a silly time with the sip.conf variable
> progressinband=never,no,yes (arg!) I dug a little deeper into the  
> chan_sip
> code.
>
> It appears on a SIP->Zap call the ISDN channel is opened, and before  
> you can
> say 'boo' sip_write() in chan_sip is called.... this appears to  
> occurs prior
> to any ISDN signalling (such as PRI_EVENT_PROCEEDING etc..)
>
> sip_write doesn't seem to care at all what progressinband is set to,  
> and if
> it gets a frame when the SIP channel is not in AST_STATE_UP it  
> generates a
> 183 with SDP (then sets SIP_PROGRESS_SENT)
>
> Does this behaviour seem strange?   I'm not really sure if this is a  
> bug, a
> 'its just like that' thing, or something strange with our ISDN that is
> unusual?
>
> In an ideal world (for me anyway... *grin*) I would think that
> progressinband=never (or even progressinband=no) would mean that 180
> Ringing, 486 Busy etc would be used and 183 Session Progress with  
> SDP would
> not...
>
> I have done some basic testing and if I patch as follows...
> >>>>>>>>>>>>>>>>>>>>>
>
> url to patch document:
> From ds at seiros.ru Mon May 1 04:41:40 2006 From: ds at seiros.ru ...
>
> Richard
>
>
> On Dec 21, 2007, at 9:57 AM, Remi Quezada wrote:
>
>> Hi,
>>
>> I have a Asterisk that connects to the PSTN via a PRI.   After  
>> Asterisk
>> sends the setup message it immediately sends a 183 Session  
>> Progress.  Is
>> there a way I can change it so that it sends a 100 Trying instead?
>> Because I am having some issues with a equipment where it does not  
>> play
>> a busy tone as a result of sending a 183 Session Progress then the  
>> 486 Busy.
>>
>> Thanks
>>
>> Remi
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071221/8ccfa0fe/attachment-0001.htm