|Summary:||ASTERISK-18705: Asterisk Support of SipConnect 1.1|
|Reporter:||Neeharika Allanki (neeharika)||Labels:|
|Date Opened:||2011-10-11 09:59:20||Date Closed:||2017-12-13 07:42:42.000-0600|
|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:
Registration (RFC 6140)
-Basic GIN registration
-did not test the GIN interactions with the GRUU and reg-event package extensions)
-Basic DID/DOD calls
-Calling name/number delivery with and without privacy
-Call Transfer (attended and blind)
|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?