Summary:ASTERISK-18705: Asterisk Support of SipConnect 1.1
Reporter:Neeharika Allanki (neeharika)Labels:
Date Opened:2011-10-11 09:59:20Date Closed:2017-12-13 07:42:42.000-0600
Versions:Frequency of
Description: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:

-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
Comments:By: Paul Belanger (pabelanger) 2011-10-12 11:50:13.692-0500

New features needed to be applied to trunk.   Also, it might be worth discussion this patch on IRC or asterisk-dev mailing list.

By: Leif Madsen (lmadsen) 2011-11-01 08:52:31.416-0500

Agreed. This will definitely need to be brought up on the asterisk-dev mailing list for discussion. Thanks for the patch!

By: Leif Madsen (lmadsen) 2011-11-01 14:50:15.757-0500

We'll need the initial patch submitted against trunk, and attached to this issue initially.

By: Olle Johansson (oej) 2014-10-13 07:46:00.260-0500

What's the status after all this time? Do we have a disclaimer for this code?