> On March 22, 2014, 4:39 p.m., Olle E Johansson wrote: > > I don't see what happens with the phone-context argument. Shouldn't we pass > > that on as a channel variable? > > Geert Van Pamel wrote: > We return this into the hostport. > > Geert Van Pamel wrote: > According to RFC 3966 phone-context is either a domain-name, or (part of) > an international telephone number (indicated with +prefix). > It is used by a gateway to know how to dial the "local" number... the > local number must be unique within its context... > > Olle E Johansson wrote: > So it ends up in the SIPDOMAIN variable in the dial plan? It has to be > reachable in the dial plan somehow. > > Geert Van Pamel wrote: > The variable ${SIPDOMAIN} contains the local IP address of the Asterisk > server. > The userinfo arrives in ${CALLERID} and is displayed on the display of > the called device, and arrives in the CDR file. > Actually I do not know into which variable the incoming hostport info is > copied to? > Could somebody else answer this question?
If I place a normal call to sip:ge...@example.com to my Asterisk server. "geert" will be the extension I'm looking for, "example.com" ends up in SIPDOMAIN. It's not the local IP address, it's the domain/host part of the request URI in the INVITE. I would prefer if phone context ended up in TELPHONECONTEXT so I could use it the same way as SIPDOMAIN in the dial plan. It should not end up in SIPDOMAIN as it is not a SIP uri. That way an extension in a local context can be routed differently than an extension in a global context. - Olle E ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/#review11323 ----------------------------------------------------------- On March 22, 2014, 2:08 p.m., Geert Van Pamel wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3349/ > ----------------------------------------------------------- > > (Updated March 22, 2014, 2:08 p.m.) > > > Review request for Asterisk Developers, Corey Farrell, lmadensen, Matt > Jordan, and wdoekes. > > > Bugs: ASTERISK-17179 > https://issues.asterisk.org/jira/browse/ASTERISK-17179 > > > Repository: Asterisk > > > Description > ------- > > Implements RFC-3966 TEL URI incoming INVITE. > > See https://issues.asterisk.org/jira/browse/ASTERISK-17179 for a description > of the original isssue. > > I have been patching all versions since Asterisk 1.6. I would like to include > the code into the main trunk for version 13. > > Previously Asterisk was failing with error on incoming IMS call: > > Nov 13 17:52:05 NOTICE[27459]: chan_sip.c:6973 check_user_full: From address > missing 'sip:', using it anyway > > Nov 13 17:52:05 WARNING[27459]: chan_sip.c:6525 get_destination: Huh? Not a > SIP header (tel:0987654321;phone-context=+32987654321)? > > Reason: tel: protocol was not recognized. > > > Diffs > ----- > > /trunk/channels/sip/reqresp_parser.c 410429 > /trunk/channels/chan_sip.c 410429 > > Diff: https://reviewboard.asterisk.org/r/3349/diff/ > > > Testing > ------- > > Executed an incoming TEL URI INVITE connection. > CLI was present on the display and in the CDR file. > No errors on SIP debug output. > > > File Attachments > ---------------- > > RFC-3966 tel URI patch > > https://reviewboard.asterisk.org/media/uploaded/files/2014/03/13/cad7a996-88c1-47fe-a2a9-cc6987af3b75__rfc-3966-tel-uri-patch-diff.txt > > > Thanks, > > Geert Van Pamel > >
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