[asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1

[Olle E Johansson]

> On None, Olle E Johansson wrote:
> > First, submit the patch through our bug tracker to make sure we have a proper license agreement. Also, please fill in the form of the first page here in review board.
> > 
> > Secondly, I can't make sense of having a "sip connect" setting. I would like to propose that you separate the path work, which we already have a patch for, from the gin work. Some of your changes of id's should be handled by the dial plan (adding a + can be done there for e.164 phone numbers). Some of them may not be doable and we need to discuss it.
> > 
> > So instead of a big patch, please break it up and let us review smaller patches. 
> > 
> > I can not accept this as part of asterisk as it stands, but really want to see the functionality in asterisk. Let's continue working on this. Thank you very much for contributing this code!
> > 
> > /O
> 
> Andrew Olmsted wrote:
>     Hi Olle,
>     
>     I worked on this patch with Neeharika and appreciate your comments.  We are currently testing e.164 in the dial plan and if this is successful I will take it out of the patch.
>     
>     For the RFC 6140 changes for GIN support do you have a recommendation on being able to turn this on? We are considering a RCF6140 flag on a per-peer basis, but you said you didn't see the point to a sip connect setting.
>     
>     Taking e.164 out of the patch will already make it much smaller, and doing the path work with the previously submitted patch will make it smaller still. Hopefully reducing the patch this much will make it easier to review. Should we resubmit the patch here after implementing your recommendations?
>     
>     Thanks,
>     Andrew

Andrew - can you please verify that the Path: header support is written by you and not the patch we already have in the issue tracker? licensing is important for us. Thanks.

I would propose we do take the code in bit by bit, starting with the Path code.

/O


- Olle E


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On Oct. 18, 2011, 3:11 p.m., Neeharika Allanki wrote:
> 
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> 
> (Updated Oct. 18, 2011, 3:11 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> This Chan-SIP patch brings Asterisk into compliance with the SIPconnect1.1.  SIPconnect1.1 is a newly released SIP Forum specification that describes a common set of signaling and media interworking procedures for the SIP Trunk interface between a SIP-based IP-PBX and a SIP-enabled Service Provider network. This patch, coupled with specific Asterisk configuration settings, will enable Asterisk to comply with the normative SIP-PBX requirements specified in SIPconnect1.1.
> 
> The patch diff listings being submitted are against Asterisk version 1.8.11.The patch itself has been tested against the 1.8.0 version of Asterisk for the following SIPconnect1.1 functions/capabilities:
> 
> Security
> -TLS
> -SIP Digest
> 
> Registration (RFC 6140)
> -Basic GIN registration
> -did not test the GIN interactions with the GRUU and reg-event package extensions)
> 
> Calling features
> -Basic DID/DOD calls
> -Calling name/number delivery with and without privacy
> -Early media
> -Call Forwarding
> -Call Transfer (attended and blind)
> -Emergency calls
> -DTMF relay
> 
> 
> This addresses bug ASTERISK-18705.
>     https://issues.asterisk.org/jira/browse/ASTERISK-18705
> 
> 
> Diffs
> -----
> 
>   /trunk/channels/chan_sip.c 333472 
>   /trunk/channels/sip/include/sip.h 333472 
>   /trunk/configs/sip.conf.sample 333472 
>   /trunk/include/asterisk/strings.h 333472 
> 
> Diff: https://reviewboard.asterisk.org/r/1515/diff
> 
> 
> Testing
> -------
> 
> 
> Thanks,
> 
> Neeharika
> 
>

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