[asterisk-dev] [Code Review]: Asterisk Support of SIP Connect 1.1
Terry Wilson
reviewboard at asterisk.org
Tue Oct 18 16:49:14 CDT 2011
> 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.
is_int() is the name, btw.
- Terry
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On Oct. 18, 2011, 3:11 p.m., Neeharika Allanki wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1515/
> -----------------------------------------------------------
>
> (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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