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

[Terry Wilson]

> 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.

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.


- Terry


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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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