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

[Andrew Olmsted]

> On Oct. 18, 2011, 4:47 p.m., Terry Wilson wrote:
> > /trunk/include/asterisk/strings.h, lines 86-101
> > <https://reviewboard.asterisk.org/r/1515/diff/1/?file=21056#file21056line86>
> >
> >     This function already exists in a weird place: pval.h and implemented in res/ael/pval.c.
> 
> Terry Wilson wrote:
>     is_int() is the name, btw.
> 
> Terry Wilson wrote:
>     Actually, in some cases the pval.c version doesn't show up (like on Solaris, apparently). I'm probably going to add the ast_check_digits function to strings.h for another issue, but using the code from pval.c. It will break this patch, but be easy enough for you to fix.

Thanks for this information, I did indeed miss it in that location. Hopefully we will be moving e.164 out of the patch and into the dial plan which should eliminate the need for this function in our code, but I'm glad it is useful for another purpose.


- Andrew


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On Oct. 18, 2011, 3:11 p.m., Neeharika Allanki wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1515/
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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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